| 00:00.18 | tzafrir_laptop | blop, if it is a simple patch and it either fixes a problem or you have a very good reason to believe it doesn't harm eisting users, it is a good candidate to be merged in stable |
| 00:00.47 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 00:01.08 | tzafrir_laptop | I'll have to do the work for 1.0.8 pretty soon anyway |
| 00:01.31 | |Vulture| | okay back to home later guys |
| 00:01.45 | blop | i think its already merged actually |
| 00:01.48 | blop | w8 :) |
| 00:02.18 | blop | that one http://bugs.digium.com/bug_view_page.php?bug_id=0002973 :) |
| 00:03.41 | criptos | you can make a callpickup at a iaxy without app_pickup rigth? |
| 00:05.09 | tzafrir_laptop | blop, there's a comment there about "not in 1.0". Is app_sms in 1.0? |
| 00:06.01 | tzafrir_laptop | yes, it is. So it seems that it has already drifted enough |
| 00:06.20 | tzafrir_laptop | Or that those changes depend on other HEAD code |
| 00:06.51 | blop | hum :) |
| 00:08.05 | blop | dunno :) i still must fix my sms confs ...they're still not running flawlessly |
| 00:08.09 | tzafrir_laptop | Any idea if what I wrote is correct or nonsense? |
| 00:08.38 | blop | i'm not use to mantis/cvs and such things |
| 00:08.45 | blop | +d |
| 00:09.17 | blop | so i presume those changes will be merged one day :) |
| 00:09.25 | blop | anyway |
| 00:09.30 | blop | i got a small problem here |
| 00:09.34 | blop | with a pots phone |
| 00:09.49 | blop | which is working fine on an iaxy, but not on a fxs @ tdm |
| 00:10.04 | JunK-Y | tzafrir_laptop: are u talking about ur he patch? |
| 00:10.18 | blop | it seems the fxs dont detect the dtmf tones when pressing digits |
| 00:11.45 | Chuji | you sure it's not detecting? |
| 00:11.52 | Chuji | put the logger in debug |
| 00:12.12 | Chuji | ~debug |
| 00:12.14 | jbot | ACTION DeBuggers $1 |
| 00:12.24 | Chuji | ~logger |
| 00:12.53 | blop | the fact is |
| 00:13.16 | blop | i get the dialtone, and it stays even if i press digits |
| 00:13.26 | blop | and when using another pots phone in // |
| 00:13.36 | blop | i can dial using the other phone's dialpad |
| 00:14.16 | blop | but the phone is working correctly when plugged into an iaxy |
| 00:14.26 | tzafrir_laptop | JunK-Y, I'm working right now on my e patch |
| 00:14.36 | blop | 0-8 seems to have no effect, 9 trig a sort of congestion |
| 00:14.37 | tzafrir_laptop | the stuff in the mantis is total crap |
| 00:14.50 | Chuji | whatch the debug |
| 00:14.58 | JunK-Y | tzafrir_laptop: u should have any call to _fr in ur function |
| 00:15.00 | JunK-Y | just to _he |
| 00:15.02 | Chuji | /etc/asterisk/logger.conf |
| 00:15.04 | Chuji | console => notice,warning,error,debug |
| 00:15.13 | JunK-Y | s/should/shouldnt/ :) |
| 00:15.14 | Chuji | set that, and do a logger rotate |
| 00:15.15 | blop | i'll check that |
| 00:15.31 | tzafrir_laptop | JunK-Y, just a bad case of cut&paste programming |
| 00:15.53 | JunK-Y | that what ive realized a long time ago, see my note :) |
| 00:15.59 | blop | alright, debug activated, i'll retest |
| 00:17.20 | tzafrir_laptop | I have a much improved code right now, but it still seems to have bad problems. I'll probably post it even if I don't have anything working, because it is something others can comment on |
| 00:18.31 | JunK-Y | to be honest, i dont have so much time, to spend on, i suggest, work on it, and when it works, just submit new patch. |
| 00:19.45 | blop | hum |
| 00:19.45 | blop | Verbosity is at least 10 |
| 00:19.45 | blop | -- Starting simple switch on 'Zap/1-1' |
| 00:19.45 | blop | -- Hungup 'Zap/1-1' |
| 00:19.56 | blop | nothing happens when pressing digits :D |
| 00:20.08 | fugitivo | blop: damn, i thought this was my asterisk console |
| 00:20.21 | blop | lol |
| 00:20.28 | fugitivo | :) |
| 00:21.31 | Chuji | You sure it's in debug? Verbosity doesn't really have anyhthing to do with it |
| 00:22.29 | blop | yes |
| 00:22.32 | blop | sure :) |
| 00:22.37 | blop | oh :) |
| 00:22.47 | blop | now that i log rotated :D |
| 00:22.52 | blop | it says sth :) |
| 00:22.59 | blop | soz :) |
| 00:23.27 | blop | maybe i should use pastbin or so? |
| 00:23.32 | Chuji | It should echo dtmf activity now |
| 00:24.46 | blop | i get that http://pastebin.com/296824 when : pick up, dialing 21 .. waiting .. nothing , hangup |
| 00:25.20 | *** join/#asterisk Eminence (achin@cpe-24-198-66-186.maine.res.rr.com) |
| 00:26.01 | blop | its weird it said disabling echo cancellation, i sure have echocancel=yes in zapata.conf, but anyway i tested disabling it already |
| 00:30.03 | *** join/#asterisk ryguillian (~ryguillia@c-67-176-193-252.hsd1.il.comcast.net) |
| 00:33.13 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 00:34.03 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 00:35.49 | Eminence | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 00:35.56 | Eminence | is that document still accurate? |
| 00:37.01 | Sedorox | looks to be |
| 00:38.43 | Eminence | ok. i followed all those directions and things arnt working yet for me. i cant tell if asterisk isn't configured properly, or my client softphone isnt |
| 00:39.14 | puppet | eminence: yeah looks to be |
| 00:39.43 | Sedorox | Eminence: could be either |
| 00:39.47 | Sedorox | whats your problem? |
| 00:39.59 | *** join/#asterisk SwK (~ken@12-219-156-206.client.mchsi.com) |
| 00:40.02 | puppet | pastebin your logoutput on asterisk |
| 00:40.19 | Eminence | asterisk starts fine, with no errors that i can see |
| 00:40.31 | puppet | start asterisk with |
| 00:40.35 | puppet | asterisk -vvvvvvvvvvvvvvvvvvvcr |
| 00:40.54 | puppet | and tell us what it says when u try to connect with softhpone |
| 00:41.23 | SwK | set verbose 99 |
| 00:41.27 | SwK | that works good too |
| 00:41.39 | puppet | swk: but vvvvvvvvvv is easier to findin ctrl+r ;P |
| 00:41.49 | Eminence | my softphone is X-lite (for windows) |
| 00:42.10 | Eminence | i'm checking Xlite's configuration; one sec |
| 00:43.30 | meppl | gute nacht - good night |
| 00:43.59 | SwK | any cisco as5300 experts around? |
| 00:46.39 | *** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188) |
| 00:47.23 | axscode | hi guyz.. can i connect my asterisk server to another SIP server? so that my user from my server can dial a phone from another SIP server? |
| 00:47.29 | Eminence | ok, xlite starts up and says "awating proxy login info" while my astrisk console says nothing |
| 00:47.46 | file | axscode: yes, see sip.conf |
| 00:48.07 | file | Eminence: configure it, restart it |
| 00:48.41 | *** join/#asterisk desipher (~kvirc@68-232-94-166.chvlva.adelphia.net) |
| 00:48.52 | Eminence | xlite is configured with 'username' and 'password' from sip.conf. 'sip proxy' is the ip address of the box running astrisk. 'domain/relam' is blank |
| 00:49.21 | file | restart Xlite, it seems to be ... evil about that |
| 00:49.32 | Eminence | i already have |
| 00:49.40 | file | then put in mor einfo |
| 00:49.42 | file | more info |
| 00:49.59 | file | I usually put in all that stuff, plus domain, plus outbound sip proxy |
| 00:49.59 | Eminence | more info into xlite? such as what? |
| 00:50.03 | Sedorox | let me grab it real quick |
| 00:50.09 | Sedorox | there is something more you have to add |
| 00:50.29 | SwK | usually puttying asterisk in the domain/realm makes it start working |
| 00:50.40 | Eminence | what should domain be? (and/or where is that defined in the asterick conf files? |
| 00:50.42 | file | such a picky picky piece of software sometimes |
| 00:50.42 | SwK | unless you have your realm in sip.conf set to something |
| 00:50.50 | SwK | in which case you should put that in there |
| 00:51.06 | Eminence | i havent defined any realms |
| 00:51.19 | Sedorox | as the realm.. put in the IP of the * box... |
| 00:51.25 | SwK | then the deafult realm of "asterisk" is in effect |
| 00:51.36 | Sedorox | lol |
| 00:51.37 | axscode | file: would u like to help me of the scenario.. just a glimps.. is the sip proxy something.. that makes my asterisk server register to another sip-proxy? then if thats the case.. how would i put to extension to dial to a phone from another sip server? |
| 00:52.13 | SwK | if you real sip.conf it says in the realm comment something along the lines of "the default is asterisk" |
| 00:52.15 | Eminence | 'login failed' well that's good. at least now my softphone is contacting my asterisk server |
| 00:52.17 | file | axscode: you need a peer entry, and a register line (if you need to register to them), plus a line for dialing in extensions.conf |
| 00:52.40 | dav__ | asterisk+fwd problem: could someone take a look at this? http://cpp.enisoc.com/pastebin/?5837 |
| 00:52.57 | *** join/#asterisk ApEtc (apetc@ip-66-218-241-107.cableaz.com) |
| 00:53.07 | file | dav__: did you follow the example on the FWD site? cause it works great |
| 00:53.39 | Sedorox | 0_o |
| 00:53.42 | Sedorox | xlite for linux.... wow... |
| 00:54.11 | SwK | yeah Sedorox |
| 00:54.15 | SwK | there's even an OSX version |
| 00:54.20 | Sedorox | yea.. I saw.. |
| 00:54.34 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 00:54.36 | Eminence | setting domain to 'asterisk' fixed all my problems. thanks for the help folks! |
| 00:54.38 | axscode | file: yes.. thanks.. and what is the line for dialing to anothe SIP server? through realm? or how to do that? |
| 00:54.40 | FuriousGeorge | hi all |
| 00:54.52 | FuriousGeorge | shido6: are you available |
| 00:55.09 | file | axscode: look at sip.conf, there's examples GALORE |
| 00:55.20 | dav__ | file, i can't find that |
| 00:55.40 | file | look at sip.conf.sample in the configs directory, in the asterisk directory |
| 00:55.44 | Eminence | now that i got it working, time to play around with the conf files |
| 00:56.03 | file | dav__: Google. |
| 00:56.12 | file | type in "iax fwd" |
| 00:56.16 | file | it's the first page that comes up :P |
| 00:58.46 | *** part/#asterisk ryguillian (~ryguillia@c-67-176-193-252.hsd1.il.comcast.net) |
| 00:59.07 | *** join/#asterisk irv999 (~irv999@ool-44c241c7.dyn.optonline.net) |
| 00:59.09 | irv999 | lo all |
| 00:59.23 | irv999 | I have a stupid polycom / asterisk question |
| 00:59.36 | FuriousGeorge | anyone know of any way to tell if i have sip or aix minutes from nufone? i cant remember what i bought |
| 00:59.54 | irv999 | fur: maybe their website |
| 01:00.39 | FuriousGeorge | irv999: thats what i was trying, ill look again. maybe shido comes back soon |
| 01:00.47 | irv999 | fur: could bd |
| 01:01.23 | irv999 | I have added an extension in phonexxx.cfg and sip.conf. what else do I have to do to make the extension on the phone active? |
| 01:03.04 | *** join/#asterisk nine76 (~t00r@cpe-24-33-147-236.woh.res.rr.com) |
| 01:03.11 | nine76 | hello all |
| 01:03.50 | irv999 | err nto phone.. extension active.. |
| 01:04.56 | axscode | extension.conf and sip.conf .. thats all i touch to register all my VoIP SoftPhone and HardPhone. |
| 01:05.36 | axscode | and it works well... if u want to add some feature like conferencing and voicemaill thats when u touch other confs |
| 01:05.45 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 01:06.12 | irv999 | axs: extension.conf?... hrmmm |
| 01:06.34 | axscode | extensions.conf with s. |
| 01:07.58 | irv999 | axs: I am looking at that.. I have other phones working without them ebing in extnesions.conf.. |
| 01:08.19 | *** join/#asterisk gpearson (~Graham@bluecow.niesc.k12.in.us) |
| 01:08.33 | *** part/#asterisk zwi (~zwi@216.88.131.43) |
| 01:08.51 | irv999 | axs: the icon is listed on the phone.. just it is hollow.. not solid |
| 01:09.10 | irv999 | axs: I am missing something else.. a reload or something like that |
| 01:09.44 | axscode | have u reload it? go to the CLI console then.. type reload<enter> |
| 01:12.01 | *** join/#asterisk NormAst (NormAst@toronto-HSE-ppp3975457.sympatico.ca) |
| 01:12.08 | NormAst | Hi All! |
| 01:12.56 | irv999 | axs: trying 1 sec |
| 01:16.55 | axscode | asterisk -x reload also |
| 01:17.04 | axscode | or asterisk -rvvvc |
| 01:17.06 | FuriousGeorge | while im here and its slow, i may as well ask: i use gentoo, and for some reason every system i install asterisk on wont work until i run a ztcfg -vv after a reboot. rather than try to figure out whats wrong with the gentoo bootscript, im thinking of sticking it in rc.conf. does that sound like a good idea? |
| 01:17.24 | FuriousGeorge | or should i look in a different script |
| 01:18.16 | FuriousGeorge | sadly im almost as new to linux as i am to asterisk :) |
| 01:19.35 | irv999 | axs: worked.. thanks!~ |
| 01:20.05 | axscode | no. thanks to reload.. not ot me |
| 01:20.07 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 01:20.13 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 01:20.26 | redder86 | bkw_: there? |
| 01:21.58 | Nugget | <bumper> http://lowdrag.org/gallery/screenies/IMG_0801?full=1 <-- hah |
| 01:26.55 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 01:26.56 | *** join/#asterisk AgiNamu (~Michael@200.6.212.133) |
| 01:27.09 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 01:27.28 | |Vulture| | if I kill myself... you will all know its because * + faxing drove me insane |
| 01:27.52 | axscode | haha |
| 01:27.53 | AgiNamu | Thanks for the infol |
| 01:28.03 | axscode | thats quite and intro vulture |
| 01:30.42 | |Vulture| | thank you |
| 01:31.49 | |Vulture| | yea... PRI + * + Faxmachine == insanity |
| 01:32.19 | bkw_ | i'm here |
| 01:32.21 | bkw_ | whats up |
| 01:33.07 | axscode | dont know about fax |
| 01:33.31 | |Vulture| | bkw_: ever setup PRI --> * --> TDM FXS --> Fax Machine? |
| 01:33.56 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
| 01:34.14 | PatrickDK | vulture, what is the pri connecting using? |
| 01:34.18 | PatrickDK | digium t1 card? |
| 01:34.40 | |Vulture| | te110xp |
| 01:34.53 | |Vulture| | v1.2 |
| 01:35.10 | PatrickDK | did you tell asterisk not to detect fax signals on the fxs port? |
| 01:35.10 | |Vulture| | tried v1-0;HEAD;1.0.7 |
| 01:35.22 | |Vulture| | PatrickDK: not to? |
| 01:35.28 | axscode | a 1MBPS of Internet Connection.. how many user can call simultanously? |
| 01:35.32 | |Vulture| | I have it faxdetect=both |
| 01:35.45 | PatrickDK | vulture, you testing fax machine using sending or recieving? |
| 01:35.45 | |Vulture| | axscode: depends on the codec |
| 01:36.00 | axscode | hmm.. using ulaw and alaw i guess |
| 01:36.07 | |Vulture| | PatrickDK: both, and I am using spandsp now to test because I had to leave the office |
| 01:36.38 | PatrickDK | vulture, I don't get it, pri -> * -> tdm fxs-> faxmachine doesn't involve spandsp |
| 01:38.02 | axscode | using ulaw and alaw codec, using rackmount 3Ghz Dual processor, Asterisk on FreeBSD 5.4 with Gigabit Ethernet Port. with 1Gig of RAM... and 1MBPS of Internet... so how many user do you think more or less going outside the net with no diether? |
| 01:38.03 | |Vulture| | PatrickDK: well I was using the fax machine but now I had to come home and spandsp is getting choppy recieves too |
| 01:38.11 | |Vulture| | so it has to be between the PRI --> * |
| 01:39.33 | PatrickDK | axscode, using ulaw or alaw, the problem won't be cpu power |
| 01:39.46 | PatrickDK | the problem will be network connection, then ram |
| 01:40.12 | |Vulture| | I prefer ilibc between * and g729 on clients |
| 01:40.25 | PatrickDK | 1mbps of connection with ulaw/alaw will give you 11 calls |
| 01:40.33 | PatrickDK | I descent 500mhz computer can do that |
| 01:40.43 | PatrickDK | with around 256-512m of ram |
| 01:41.03 | PatrickDK | heh, I perfer ulaw between phones |
| 01:41.11 | PatrickDK | and ulaw if I can between *, otherwise gsm |
| 01:41.57 | |Vulture| | well ulaw if they are internal |
| 01:42.03 | |Vulture| | I use g729 if they are remote |
| 01:42.31 | axscode | 11 calls? with 1 MBPS? omg.. thats bad.. |
| 01:42.41 | axscode | so locally with 100MBPS thats make 1100 calls? |
| 01:42.51 | PatrickDK | axscode, I assume that is 1mbps both ways |
| 01:43.04 | axscode | not bad though.. |
| 01:43.20 | axscode | ~GSM |
| 01:43.21 | jbot | from memory, gsm is Groupe Spécial Mobile |
| 01:43.43 | PatrickDK | gsm will give you around 28 calls |
| 01:44.04 | PatrickDK | I think, that might be alittle low |
| 01:44.32 | axscode | but that depends on the phone right? if it supports GSM? |
| 01:44.51 | PatrickDK | asxcode, it depends if both sides of the connection supports it |
| 01:44.54 | robl^ | if you use IAX2 trunking with ulaw/alaw you will save a bit of bandwidth and be able to go like 14-15 calls on 1mbps |
| 01:44.55 | axscode | and one more thing.. why i have echo on the phone.. i really hate that echo.. |
| 01:45.19 | PatrickDK | echo is when you convert telephone lines to digital |
| 01:45.23 | PatrickDK | it will always be there |
| 01:45.33 | PatrickDK | unless every phone line and cnnection is perfect |
| 01:45.45 | PatrickDK | the best you can do it try to cancel itself out |
| 01:45.59 | axscode | ahh ok. ic ic.. thats where echo starts.. hmm.. ok nevermind.. thats fine though.. just make me feel that theres a delay |
| 01:46.12 | axscode | cancel it self out? wwhat u mean? |
| 01:46.20 | PatrickDK | there is a delay, in the analog to digital conversions |
| 01:46.35 | PatrickDK | that is what makes the phone lines problem noticable in the form of echo |
| 01:46.49 | PatrickDK | normal phone lines echo is so fast, that you don't notice it |
| 01:47.16 | PatrickDK | you have to setup a dsp to record the echo and attempt to kill it |
| 01:47.41 | axscode | hey pats.. i have a local phone from a telco with alive connection from it.. how will i connect my ASTERISK to it. so that my VoIP phones can call outside using that TELCO. its like when ur in a hospital u dial 9 then the number.. and u can call outside the hospital |
| 01:48.20 | PatrickDK | that all depends |
| 01:48.27 | PatrickDK | I can say unless you give alot of detail |
| 01:48.27 | axscode | do i need a device? |
| 01:48.35 | PatrickDK | and I'm not going stay around long enouh for that |
| 01:48.41 | PatrickDK | asxcode,normal pots line? |
| 01:48.49 | axscode | ~pots |
| 01:48.50 | jbot | hmm... pots is Plain Old Telephone Service as in "Old Analogue Crap" |
| 01:48.57 | PatrickDK | what most people have at home |
| 01:48.59 | axscode | hmm yeah.. pots.. |
| 01:49.10 | PatrickDK | ya,you need a fxo module/adaptor |
| 01:49.16 | axscode | ~FXO |
| 01:49.17 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 01:49.33 | axscode | ~PBX |
| 01:49.34 | jbot | extra, extra, read all about it, pbx is a Private Branch eXchange |
| 01:49.55 | axscode | hehe.. is PBX a device or it can be a software? |
| 01:50.07 | PatrickDK | pbx is normally software |
| 01:50.15 | PatrickDK | but it nomally runs on special hardware |
| 01:50.24 | PatrickDK | so people think of the pbx as hardware |
| 01:50.41 | PatrickDK | asterisk is the only generic software version, that is free, ccm is the other |
| 01:50.52 | axscode | ok.. so is there a PBX and FXO software for i386~ |
| 01:51.00 | PatrickDK | fxo isn't software |
| 01:51.10 | axscode | FXO is a modem type? |
| 01:51.38 | PatrickDK | fxo is basically a souncard that emulates a phone |
| 01:51.54 | PatrickDK | look at voip-info.org |
| 01:52.07 | axscode | ahh... soundcard.. |
| 01:52.15 | tzanger | PatrickDK: interesting way of putting it |
| 01:52.16 | axscode | how about fax voice modem? |
| 01:52.19 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
| 01:52.32 | PatrickDK | modern modems do the soundcard stuff, for doing answering machine stuff |
| 01:52.49 | PatrickDK | tzanger, I think it is easy to undersand that way |
| 01:53.00 | PatrickDK | if you say modem, people think digital, and well, it isn't digital :) |
| 01:53.06 | axscode | thing is i dont even have a modem. haha... all DSL modem.. and its an external one.. |
| 01:53.07 | tzanger | PatrickDK: how about "it emulates a telephone" |
| 01:53.25 | PatrickDK | tzanger, they still normally have questions after that :) |
| 01:53.32 | tzanger | :-) |
| 01:54.07 | axscode | ahehehe... ur smart guyz. im new to this VoIP thing.. and im planning to integrate it with wireless connectivity.. |
| 01:54.16 | axscode | that makes my IPAQ a phone i guess. |
| 01:54.20 | |Vulture| | wow... have fun |
| 01:54.35 | PatrickDK | wireless adds alot of new problems to work around :) |
| 01:54.37 | |Vulture| | wireless + voip.... |
| 01:54.42 | PatrickDK | as 802.11 sucks for voip |
| 01:54.53 | |Vulture| | yea I have 1 phone on wireless bridge |
| 01:55.02 | PatrickDK | 802.11 was made more for power savings, and that makes it suck for voip |
| 01:55.09 | axscode | hehe 802.11G = 56MBPs... morelikely 108 soon.. |
| 01:55.20 | PatrickDK | axscode, speed doesn't matter |
| 01:55.21 | |Vulture| | thats not your problem |
| 01:55.25 | |Vulture| | its latency |
| 01:55.26 | axscode | o bad |
| 01:55.38 | axscode | hmm.. speed like processing u mean? |
| 01:55.43 | |Vulture| | no |
| 01:55.49 | PatrickDK | no, wireless speed has no effect |
| 01:56.02 | |Vulture| | as long as it can transmit the codec |
| 01:56.06 | axscode | coz i have tried it.. it works fine.. |
| 01:56.38 | *** join/#asterisk stkn (nobody@stkn.developer.gentoo) |
| 01:57.18 | axscode | i tried this.... VoIP PHONE-> PowerLine Communication -> WIRELESS -> ASTERISK <--- WIRELESS <--- POWERLINE COMMUNICATION <-- VoIP PHONE |
| 01:57.37 | PatrickDK | that is evil |
| 01:57.40 | |Vulture| | yea |
| 01:57.42 | |Vulture| | thats a cluster |
| 01:57.54 | PatrickDK | powerline is so bad in and of itself |
| 01:58.01 | |Vulture| | you should just take a pic thats funny |
| 01:58.06 | PatrickDK | that must be the 1mb connection :) |
| 01:58.31 | axscode | PowerLine in here works great for me... im transferring 14MBPS with that powerline connectivity. |
| 01:59.00 | PatrickDK | well, powerline is bad, cause it makes it a very large antenna, and fucks with all radio equipment horribly |
| 01:59.06 | PatrickDK | I'm glad your not my neighbor |
| 01:59.14 | |Vulture| | OMG I AM GLOWING! |
| 01:59.15 | |Vulture| | :P |
| 01:59.30 | axscode | nope... actually that was before pat.. not now.... |
| 02:00.03 | PatrickDK | hmm, 19g tar file |
| 02:00.14 | |Vulture| | I love all these deprecated messages lol |
| 02:00.26 | |Vulture| | PatrickDK: backing up your porn again? |
| 02:00.36 | PatrickDK | na, accountants computer |
| 02:00.39 | axscode | but im not saying that its already a mature technology.... and thus with wireless.. but all works fine and.. im expanding them beyond their limits.. |
| 02:01.08 | axscode | im even throwing 52 Miles with my wireless.. |
| 02:01.20 | axscode | sounds great dont u think.? |
| 02:01.33 | PatrickDK | I hope you turn off all the 1mbps wireless connection stuff |
| 02:02.04 | axscode | im aware of security though.. dont worry... nothing to be compromise on my box iether. |
| 02:02.50 | axscode | its 54MBPS by the way... though having 11MBPS beyond 30miles |
| 02:03.09 | PatrickDK | well, every second, it goes at 1mbps to transmit it's id |
| 02:03.13 | PatrickDK | unless you change it |
| 02:03.36 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 02:03.45 | axscode | nope.. i quite familiar with synchronisation.. |
| 02:03.48 | AgiNamu | I use IAX2 + G729 over EVDO network |
| 02:03.50 | AgiNamu | works gret |
| 02:03.51 | axscode | im* |
| 02:03.58 | axscode | ~IAX |
| 02:03.59 | jbot | [iax] port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". |
| 02:04.07 | AgiNamu | but it sucks majorly on CDMA |
| 02:04.15 | axscode | whats EEKS for? |
| 02:04.29 | AgiNamu | EEKS? |
| 02:04.30 | AgiNamu | or IAX |
| 02:04.35 | file | I'm answering asterisk-users posts |
| 02:04.38 | file | someone help me |
| 02:04.40 | AgiNamu | IAX2 is a voip protocol, like SIP or H323 |
| 02:04.46 | file | like, knock me unconcious |
| 02:04.47 | AgiNamu | except not as sucky |
| 02:04.48 | axscode | IAX... promounced as eeks.. sounds cool. haha. |
| 02:04.56 | axscode | ahh.. ice. its a proto... |
| 02:05.09 | axscode | what proto supports video phones? |
| 02:05.36 | shmaltz | anybody here using bayoone? |
| 02:07.25 | shmaltz | I mean bayonne |
| 02:08.10 | axscode | ~bayonne |
| 02:09.33 | bjohnson | [bayonne] bugger off and find someone else to bother |
| 02:09.34 | bjohnson | \ |
| 02:10.04 | robl^ | bayonne is another IP phone switch software. its 100x more complext to configure, 100x buggier, 100x pickier about hardware.. other than that it compares to Asterisk.. oh.. and has 100x less features |
| 02:10.07 | shmaltz | Corydon76-home, we know about that, I just found out abound bayonne, so I'm trying to figure out if anybody here has tested it |
| 02:10.44 | shmaltz | robl^, I don't think its IP pbx, I think its rather hardware based (dialogic etc) |
| 02:10.57 | shmaltz | its just a PC based system |
| 02:11.13 | bjohnson | bayonne is software |
| 02:11.16 | robl^ | shmaltz, I thought it had a sip stack too |
| 02:11.27 | axscode | ahhehehe.. |
| 02:11.31 | shmaltz | robl^, at best it has h323 |
| 02:11.38 | Micc | Anyone know why I've got some big latency with meetme? |
| 02:11.49 | shmaltz | anyhow I'm trying to find out if it is more stable |
| 02:12.21 | shmaltz | like what he claims: |
| 02:12.23 | shmaltz | http://www.voip-info.org/wiki-Bayonne |
| 02:12.49 | robl^ | shmaltz, this is like going to #windows and asking if Mac OS X is better than XP. :) |
| 02:13.12 | shmaltz | robl^, i know, but still I though I might find someone here that has tested it |
| 02:13.28 | PatrickDK | duh, winxp is better :) |
| 02:13.38 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 02:13.56 | shmaltz | yeah they do sip also: |
| 02:13.58 | shmaltz | http://www.gnu.org/software/bayonne/news.html |
| 02:20.12 | Sedorox | Eeekkk... Cisco CCIE VOIP |
| 02:20.14 | Sedorox | Lab Exam Overview |
| 02:20.14 | Sedorox | The lab exam is an eight-hour, hands-on exam that tests the ability to get the VoIP solution operating in a timed test situation. |
| 02:20.34 | Micc | Can someone tell me if this is normal for meetme. (888) 206-4443 room #1234 |
| 02:21.58 | axscode | hhehehe.. WinXP rules.. thats why im here at #maxOS to convert all of you to use winXP |
| 02:22.51 | Micc | latency in a conference room is no good. |
| 02:23.55 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 02:30.01 | Nugget | what is "Max OS"? :) |
| 02:30.10 | riksta | Sedorox: set up asterisk in the session ;) |
| 02:31.04 | iq | Hi, a question: Does TE110P supports both Rob Bit T1 and ISDN T1? |
| 02:31.12 | drumkilla | iq: yes |
| 02:31.16 | shido6 | ok that was weird |
| 02:31.19 | shido6 | oh |
| 02:31.27 | shido6 | conferencing on the phone, sorry micc |
| 02:31.32 | iq | drumkilla, thanks |
| 02:31.44 | Micc | no problem. |
| 02:31.52 | Sedorox | riksta: ahahaha |
| 02:32.01 | Sedorox | "how did you solve XXXX problem " " I installed asterisk" |
| 02:32.12 | riksta | use skinny, youll get some points :) |
| 02:32.19 | Sedorox | lol |
| 02:32.32 | riksta | and bonus points for upgrading a 7960 to sip firmware |
| 02:32.38 | riksta | woooo |
| 02:32.47 | Sedorox | I'm debating if I should set some CCIE certs in my goals of life |
| 02:33.15 | file | I'm actually going through my backlog of messages and answering them |
| 02:33.19 | file | I'm crazy |
| 02:33.29 | iq | Would you know any good place to purchase TE110P... Like someone with good support |
| 02:33.38 | file | Digium! |
| 02:33.47 | shido6 | Digium, iq |
| 02:33.57 | Sedorox | Me! |
| 02:33.57 | Sedorox | j/k |
| 02:33.58 | Sedorox | :p |
| 02:34.00 | iq | file, I couldn't find Digium selling TE110P... they do have links to resellers |
| 02:35.03 | file | look at 'da bottom of the page |
| 02:35.14 | file | or look at the top |
| 02:35.20 | file | "or through the Asterisk Store." |
| 02:35.26 | file | http://store.yahoo.com/asteriskpbx/index.html |
| 02:35.28 | file | which is there |
| 02:35.42 | file | Telephony Hardware -> Wildcard TE110P |
| 02:35.55 | iq | thanks file |
| 02:36.52 | file | You're welcome, have a nice night. |
| 02:37.32 | file | I bet people will raise an eyebrow when they see 7 e-mails just from me |
| 02:38.13 | drumkilla | file: 7 emails for what? |
| 02:38.23 | file | I'm replying to posts on asterisk-users |
| 02:38.36 | drumkilla | cool |
| 02:38.43 | file | I know, it's crazy |
| 02:38.59 | file | I reached 3000 emails so I thought I should clean up |
| 02:39.54 | citats | file: i go through that every once in a while. but my answer is usually rm mail/asterisk |
| 02:43.45 | *** join/#asterisk mutilator (WebChat@i.think.napoleon.dynamiteblows.com) |
| 02:44.58 | mutilator | off topic but.. |
| 02:44.59 | mutilator | anyone here know their radius? i have the weirdest problem in existance.. |
| 02:45.24 | mutilator | kinda long to explain so i don't wanna spam it out if no one knows.. |
| 02:53.11 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 03:00.56 | Nuxi | Anybody ever compiled * on a yellowdog linux imac? |
| 03:01.00 | *** join/#asterisk brettnem (~brettnem@user-0ccsrb6.cable.mindspring.com) |
| 03:01.14 | Nugget | I run it on my powerbook, in osx. :) |
| 03:01.21 | Nugget | I don't see much point to running linux on a mac |
| 03:01.42 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
| 03:01.54 | |Vulture| | YAY I don't have to kill myself * + FAX == WINZ! |
| 03:02.00 | Nuxi | I got it at a yard sale and os 9 was messed up on it. I have no os cds soooo... |
| 03:02.20 | file | try OSX on it D00D |
| 03:02.22 | Nugget | if it's old enough to have 9 on it, yeah, linux seems like a pragmatic solution. |
| 03:02.50 | Nuxi | It's a whopping 350Mhz with 64MB ram. |
| 03:02.59 | Nugget | it's going to be worthless either way, so it might as well be worthless FAST. :) |
| 03:03.10 | file | Nuxi: FAST!!! |
| 03:03.21 | Corydon76-home | Actually, it won't be worthless if you up the memory on it... 64 is scant |
| 03:03.54 | Nuxi | I actually bought it for $10 thinking it would be good for testing endian issues. |
| 03:04.37 | Nuxi | plus, I've never had a mac before |
| 03:05.00 | robl^ | My first mac had 128K RAM :) |
| 03:05.05 | robl^ | not M |
| 03:05.28 | file | my first Mac is on my desk, my Mac Mini :) |
| 03:05.36 | Nuxi | My first computer had 8k. |
| 03:05.55 | robl^ | nice! I want a mini.. |
| 03:05.57 | Qwell | my first computer has vacuum tubes |
| 03:06.00 | Qwell | had* |
| 03:06.17 | Nuxi | I had to modify the makefile for mpg123, but so far it's building well. We'll see in a couple of hours. lol. |
| 03:06.28 | robl^ | My first computer had little beads on metal rods. |
| 03:07.16 | Nuxi | robl^ had to mine the metal and carve the bones to make it. |
| 03:08.05 | libpcp | what does it mean by AHH ? |
| 03:08.33 | robl^ | that's right.. and I had to walk 10 miles with out shoes in the snow up hill both ways to get to the mine |
| 03:09.49 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
| 03:10.23 | |Vulture| | roamer323: uphill... don't forget |
| 03:10.34 | |Vulture| | oh damn |
| 03:10.36 | |Vulture| | that cut off |
| 03:10.38 | |Vulture| | lol |
| 03:11.57 | Nugget | when I was your age we had to bang rocks together to make 1s |
| 03:13.18 | niZon | has anyone got * working on a mac mini? |
| 03:13.36 | file | niZon: yes it works fine and dandy, mmmkthx |
| 03:14.25 | santiago | hi, does anyone is using ubuntu here? |
| 03:19.25 | *** join/#asterisk bprice20 (~brandon@216.120.224.160) |
| 03:20.56 | TESTER2 | How can I change the festival voice using festival-script.pl script? |
| 03:22.11 | santiago | I need a good iax client that runs in my ubuntu |
| 03:24.50 | Qwell | santiago: try iaxcomm |
| 03:25.59 | santiago | Qwell, I get this error: "Error wxWindows Fatal Error : Couldn't Initialize IAX Client ." |
| 03:26.03 | ptiggerdine | chan_zap in HEAD is broken |
| 03:26.18 | Qwell | santiago: yeah, the errors are pretty generic |
| 03:26.33 | santiago | it works well on windows :S |
| 03:26.35 | Qwell | santiago: make sure the user you're running it as has working sound |
| 03:26.48 | Qwell | or, check if there is a verbose mode |
| 03:28.46 | santiago | Qwell, thanks, I'll try |
| 03:29.06 | mutilator | ugh |
| 03:29.11 | mutilator | i think i'm about to crack |
| 03:29.16 | mutilator | i'm kinda tired right now but don't wannt go to sleep, i don't feel like waking up to go to work anymore |
| 03:34.32 | |Vulture| | ptiggerdine: how is it broken? |
| 03:34.45 | |Vulture| | I find it hard to believe that chan_zap doesn't work at all |
| 03:38.11 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 03:40.06 | *** join/#asterisk Nukemizer (~Nuke@160.7.249.15) |
| 03:40.41 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 03:41.38 | ptiggerdine | broken == dosen't compile |
| 03:42.06 | ptiggerdine | from HEAD as of 10mins ago |
| 03:42.50 | Qwell | ptiggerdine: What errors do you get? |
| 03:44.35 | *** join/#asterisk brookshire (~matt@esbrooks3.traveller.com) |
| 03:46.55 | ptiggerdine | Qwell, chan_zap.c:35:2: #error "You need newer libpri" |
| 03:47.01 | |Vulture| | lol |
| 03:47.05 | ptiggerdine | I shall use pastebin.com for the rest |
| 03:47.11 | |Vulture| | don't bother |
| 03:47.26 | |Vulture| | run make clean;make;make install for updated libpri |
| 03:47.37 | |Vulture| | then run zaptel compile |
| 03:47.59 | |Vulture| | unless you did and they just botched the version check |
| 03:49.35 | ptiggerdine | no i haven't |
| 03:49.58 | |Vulture| | do that, then compile zaptel |
| 03:50.24 | |Vulture| | remove modules dir;compile libpri;zaptel;asterisk;asterisk-addons |
| 03:50.26 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
| 03:52.26 | ptiggerdine | thanks |Vulture| that worked well |
| 03:52.44 | dca | anyone know how to can astdb in favor of using the RT db? |
| 03:54.20 | PyroSteve | may I ask a question about nortel ? |
| 03:55.32 | Nukemizer | Does anyone know a way to make a Zap channel go Offhook, send a few DTMF digits, then "Playback" a prompt or wav file ? while staying offhook ? |
| 03:55.47 | |Vulture| | PyroSteve: like norstar? |
| 03:56.13 | |Vulture| | Nukemizer: this for a door entry device? |
| 03:56.36 | Nukemizer | Vulture, Legacy PBX Paging |
| 03:57.09 | |Vulture| | Nukemizer: you can prolly create a script to generate a file into asterisk/outgoing that runs a context |
| 03:57.45 | |Vulture| | and Zap/Channel |
| 03:58.03 | |Vulture| | that will open the channel then do the context |
| 03:58.39 | PyroSteve | |Vulture|: well sort of ... can thier phones be used with Nortel newer IP phone server ? |
| 03:59.38 | |Vulture| | PyroSteve: sorry I have never worked with Nortel VoIP technology |
| 04:00.54 | Nukemizer | Vulture, not sure I follow you, but I will experiment some more. tnx |
| 04:02.28 | |Vulture| | Nukemizer: http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 04:04.18 | Nukemizer | vulture, ahhh yes that looks promising ! thank you for your help I appriciate it :) |
| 04:04.24 | |Vulture| | np |
| 04:05.06 | zobia | hello morning |
| 04:05.37 | zobia | any body know how to registrate the software with asterisk? |
| 04:06.03 | essobi | what what? |
| 04:06.05 | Qwell | registrate which software? |
| 04:06.11 | zobia | whatever i did is chan_sip.c:7720 handle_request: Registration from |
| 04:06.16 | zobia | registrate with xlite |
| 04:06.20 | Qwell | register? |
| 04:06.33 | zobia | registration |
| 04:06.41 | zobia | registration the softphone client |
| 04:07.23 | |Vulture| | thats like basic * stuff |
| 04:07.47 | zobia | yes. for u is basic, for me is not. i am newbie to do this |
| 04:08.16 | |Vulture| | zobia: have you reviewed the wiki and some of the startup tutorials? |
| 04:08.42 | zobia | yes. i fowllow one from begin to end. i will give u the url |
| 04:08.54 | zobia | http://www.automated.it/asterisk/lah-3-6-05_3.html |
| 04:09.14 | zobia | all the same like what he said in the document. but still not working |
| 04:09.30 | zobia | i just hope i follow the right sample. |
| 04:09.50 | zobia | i spent on this for 2 nights. still result. really no idea. |
| 04:09.55 | |Vulture| | http://www.automated.it/guidetoasterisk.htm |
| 04:10.20 | |Vulture| | thats the main one for starters... you can skip down past the OS install |
| 04:10.35 | ptiggerdine | <PROTECTED> |
| 04:10.38 | ptiggerdine | yay |
| 04:11.52 | *** part/#asterisk zobia (~laura_shr@218.6.242.212) |
| 04:12.17 | *** join/#asterisk zobia (~laura_shr@218.6.242.212) |
| 04:12.43 | zobia | hello http://www.automated.it/guidetoasterisk.htm is what i already read and follow |
| 04:13.04 | zobia | and i am using the sample inside from tdm400 and for xlite |
| 04:14.04 | |Vulture| | pastebin your sip.conf |
| 04:14.08 | |Vulture| | www.pastebin.ca |
| 04:14.13 | zobia | okay. |
| 04:14.59 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 04:14.59 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 04:17.08 | |Vulture| | sup bkw_ |
| 04:19.00 | zobia | http://pastebin.ca/13699 |
| 04:19.06 | zobia | i post it there |
| 04:20.09 | |Vulture| | zobia: so in xlite what are your settings? |
| 04:20.14 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.hsd1.co.comcast.net) |
| 04:20.36 | |Vulture| | wow... thats one crazy user config |
| 04:21.11 | |Vulture| | I am guessing you want username: rkeele password: 5463 |
| 04:21.29 | zobia | i set sip proxy :64.46.165.235 ;username=1000;identify user = 1000 ; password=abc123 |
| 04:21.47 | zobia | no that one i am not using |
| 04:21.54 | zobia | i am using 1000/1000 |
| 04:21.57 | |Vulture| | okay |
| 04:22.26 | zobia | why crazy? too bad? |
| 04:22.36 | |Vulture| | 1s Ill pastebin a replacement for 1000 |
| 04:23.45 | |Vulture| | http://pastebin.ca/13700 |
| 04:24.03 | zobia | no replacement |
| 04:24.04 | dca | bkw_ you around? |
| 04:24.07 | zobia | ok i will look at it |
| 04:24.23 | |Vulture| | change the context to mytest |
| 04:24.51 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 04:25.18 | zobia | oh, i see it. u mean i missed so lots of thing? |
| 04:25.33 | zobia | i copy this from that sample. |
| 04:25.48 | |Vulture| | zobia: plus you will fix your sip.conf later right now it has a lot of trash in it |
| 04:26.12 | FuriousGeorge | why is it that in this example file both the IAX server and my local IAX client have the same name |
| 04:26.13 | FuriousGeorge | http://voip-info.org/wiki-Asterisk+settings+nufone |
| 04:26.17 | FuriousGeorge | wont that break things? |
| 04:26.38 | zobia | okay thnx |Vulture|, i will try the one u gave me. |
| 04:26.46 | FuriousGeorge | im comparing in my head to sip.conf |
| 04:27.55 | FuriousGeorge | and there the pormer is [sipproxy] and the latter is [localclientname] |
| 04:30.10 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 04:31.53 | bkw_ | dca yes |
| 04:32.08 | dca | can you enlighten me as to the intricacies of astdb vs. RT cache? |
| 04:32.34 | dca | seems the REAL info taht asterisk uses in in astdb, so, if i may ask, what is the purpose of RT? |
| 04:33.12 | dca | i'm it works weel for extensions, which is great |
| 04:33.35 | dca | but sip and iax peer/users ip and nat info is all stored in astb |
| 04:33.52 | dca | stored in RT db too, but if they differ, it seems to rely more on astdb |
| 04:34.17 | dca | in theory the shouold not differ, but oh, they do |
| 04:35.16 | zobia | hello |vulture| , u know when i input the username for xlite, actually the username is the one inside the [1000] or the context name '1000' |
| 04:35.56 | zobia | i am just wondering the context name is the username , or inside the username is the actual username i used for xlite |
| 04:36.34 | zobia | because this sample makes me so confused. |
| 04:39.05 | zobia | and callerid="" <1000> the following "<1000>" is the exten name? if it so , the sample gave it in extensions.conf |
| 04:41.13 | zobia | i post my extensions.conf in http://pastebin.ca/13702 |
| 04:43.50 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 04:48.01 | citats | nite kram |
| 04:49.13 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 04:49.57 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:51.22 | jayk_ | i'm trying to get asterisk to forward directory to a voicemail box. i have exten => _*1XX,1,Voicemail(u${EXTEN:1}) and exten => _*1XX,2,Hangup, but when i dial *101, it doesn't go through. |
| 04:51.26 | jayk_ | does anybody have any ideas? |
| 04:51.39 | jayk_ | oops, directory=directly |
| 04:52.45 | |Vulture| | jayk_: thats not a bad idea |
| 04:52.53 | |Vulture| | that should work |
| 04:53.01 | |Vulture| | what is the error? |
| 04:53.13 | jayk_ | on my CIsco 7960 phone, it says Reorder |
| 04:53.17 | jayk_ | which means the call didn't go through |
| 04:53.34 | |Vulture| | what does the * output say? |
| 04:53.36 | jayk_ | i have a feeling the "*" may not have gotten passed properly |
| 04:53.40 | jayk_ | it didn't say anything at all! |
| 04:53.44 | |Vulture| | oh |
| 04:53.46 | |Vulture| | no I know |
| 04:53.49 | |Vulture| | fix your dialplan |
| 04:53.50 | jayk_ | i have astersk -vvvvc |
| 04:53.55 | |Vulture| | dialplan.xml |
| 04:53.59 | Nukemizer | Vulture, it took me a while to get it into my grey matter but outdial works AWESOME ! thanks again |
| 04:54.04 | |Vulture| | np |
| 04:54.08 | jayk_ | i have an entry in my dialplan for *1.. |
| 04:54.26 | |Vulture| | jayk_: hmm something isn't passing it |
| 04:54.30 | jayk_ | yeah hrm |
| 04:54.35 | |Vulture| | cause that should work |
| 04:55.23 | jayk_ | http://pastebin.ca/13703 |
| 04:55.27 | jayk_ | does that dialplan.xml look okay to you? |
| 04:55.50 | |Vulture| | no that will error |
| 04:55.56 | |Vulture| | oh wait |
| 04:56.07 | |Vulture| | hmm that should work the order is right |
| 04:56.30 | |Vulture| | have you tried dialing *101 without getting a line? then sending it? |
| 04:56.32 | jayk_ | hmm |
| 04:56.34 | |Vulture| | see if it reorders |
| 04:56.47 | jayk_ | without getting a line? |
| 04:57.01 | |Vulture| | yea just hit "*101" then hit Dial or the line |
| 04:57.10 | jayk_ | yeah |
| 04:57.16 | |Vulture| | I use mostly IP500s but I have a few 7960s just been awhile |
| 04:57.17 | jayk_ | it doesn't catch it, i have to hit dial |
| 04:57.36 | |Vulture| | does *200 work? |
| 04:57.52 | jayk_ | well hrmm |
| 04:57.58 | jayk_ | something is screwed up |
| 04:58.13 | jayk_ | i have two phones sitting here..on the other one, when i hit *, it immediately catches and goes to Reorder |
| 04:58.14 | jayk_ | heh |
| 04:58.20 | jayk_ | and i've rebooted them both with the new dialplan |
| 04:59.17 | |Vulture| | change the timeout of * to 10 |
| 05:00.00 | jayk_ | im going to try with a shortened dialplan.xml |
| 05:00.09 | jayk_ | something must have got broke in it |
| 05:00.12 | |Vulture| | yea |
| 05:00.23 | |Vulture| | cause its not sending to * |
| 05:00.56 | jayk_ | yeah. |
| 05:03.15 | Juggie | i can show u a working dialplan |
| 05:06.19 | |Vulture| | Juggie: did you see his pastebin? it looks like it should work, he just cant get *1XX working |
| 05:07.19 | DaLion | anyone could tell me why i get no audio on a firewalled sessions ? |
| 05:07.30 | Qwell | grr |
| 05:07.37 | DaLion | i passed 4569 usp and 9999<>20000 |
| 05:07.39 | Qwell | slepp needs to add an ipv4. host to pastebin.ca |
| 05:08.07 | Qwell | same goes to all of you, who have an AAAA record on your www. and @ |
| 05:08.19 | Qwell | always have an ipv4 A record without an AAAA record |
| 05:09.05 | |Vulture| | DaLion: NAT=YES? |
| 05:09.27 | DaLion | ? |
| 05:09.29 | |Vulture| | DaLion: wait what is this for SIP/IAX? |
| 05:09.37 | DaLion | nah from xten to a asterisisk server.. |
| 05:09.44 | DaLion | i get no sound |
| 05:10.27 | |Vulture| | xten? |
| 05:12.34 | DaLion | eyebeam |
| 05:12.38 | DaLion | i call 800444-4444 |
| 05:12.45 | DaLion | and connects and all but no audio |
| 05:12.51 | DaLion | im wathinc tcpdump |
| 05:12.58 | DaLion | i see a few weird icmp 36 erros |
| 05:13.49 | |Vulture| | no clue thats strange |
| 05:15.38 | DaLion | icmp mobregrep ? |
| 05:15.40 | DaLion | whats that |
| 05:16.23 | DaLion | <PROTECTED> |
| 05:16.25 | DaLion | hmm |
| 05:16.35 | DaLion | could it be in need of rewriting |
| 05:17.06 | |Vulture| | check sip.conf for your externip= |
| 05:17.16 | DaLion | ahah |
| 05:17.19 | DaLion | nah |
| 05:17.25 | DaLion | i odnt have sip.conf |
| 05:17.57 | |Vulture| | kk |
| 05:25.04 | SwK | anyone around with a real faxmachine? |
| 05:25.45 | ptiggerdine | yep |
| 05:26.21 | ptiggerdine | though I don't think you'll want to send a fax here |
| 05:26.24 | SwK | i doubt you really want to test a fax to a beverly hills did tho hah |
| 05:26.24 | ptiggerdine | :P |
| 05:26.30 | SwK | :P |
| 05:26.31 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 05:26.32 | SwK | heh |
| 05:26.36 | ptiggerdine | no :D |
| 05:26.37 | lehel | hello |
| 05:26.53 | SwK | ptiggerdine: i'll send a fax there I dont care |
| 05:26.58 | ptiggerdine | lol |
| 05:27.05 | SwK | hell i call there |
| 05:27.24 | SwK | i call .nz all the time (.nz cells even) |
| 05:28.26 | Smi|k | anyone here used the new astrisk@home with sugarcrm? |
| 05:29.05 | Nukemizer | Vulture, I can make outdial work by moving my file to /outgoing folder but I have yet to understand how to call this proccess from my dialplan. Have you created a plan as this before ( Iam sure it is easier than i am making this out to be) |
| 05:32.50 | Nukemizer | Perhaps I just missed it, but it looks like dial-out can only be used by cron or other external event and not called using an extension in my Dial plan |
| 05:33.14 | Juggie | you can dialout from dialplan |
| 05:34.23 | Nukemizer | Waht I want to do i connect to a menu I have made and press "4" to activate my dialout |
| 05:35.02 | Juggie | you want to connect the active call to someone else |
| 05:35.06 | Juggie | or start a new call |
| 05:37.05 | Nukemizer | I want to connect to a Zap Channel like , hear a menu and select "4" which will pickup another zap channel and play a message. |
| 05:37.24 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 05:37.33 | Cresl1n | hey y'all :-) |
| 05:37.57 | *** join/#asterisk jbAU (~johnblade@61.8.110.41) |
| 05:38.04 | Qwell | Cresl1n: afternoon |
| 05:38.08 | Nukemizer | proccess works if i manually move the file to outgoing as "outcall" requires - perhaps I want only to issue a "cp" file when i press "4" |
| 05:38.13 | Cresl1n | ooh, it's night for me :-) |
| 05:38.20 | Qwell | Cresl1n: yeah, me too :p |
| 05:38.50 | Qwell | Cresl1n: "good night", is stupid in English |
| 05:38.53 | Qwell | so I avoid it |
| 05:39.07 | Cresl1n | lol, yep |
| 05:39.23 | Qwell | I figure, one in 24 times, I'll be right |
| 05:39.28 | Cresl1n | heh |
| 05:39.38 | Cresl1n | that's counting on an even distribution of times :-) |
| 05:39.59 | Qwell | no, because its "afternoon" in quite a few timezones |
| 05:40.14 | Cresl1n | lol, I guess that's true as well |
| 05:40.15 | Cresl1n | :-) |
| 05:40.23 | Qwell | I've put alot of thought into it. |
| 05:40.32 | lehel | Smi|k: what's that? |
| 05:40.45 | Qwell | lehel: a crm, I'd imagine |
| 05:41.06 | twisted | Qwell, so say good evening |
| 05:41.14 | Qwell | twisted: its past evening |
| 05:41.39 | twisted | Qwell, good night is usually reserved for when someone is going to bed |
| 05:41.39 | Qwell | there should be a good aftermidnight |
| 05:41.39 | Qwell | twisted: indeed, but not in many languages |
| 05:41.39 | twisted | yes, it's called good morning ;) |
| 05:41.42 | twisted | morning |
| 05:41.53 | Qwell | I'd consider morning to be after 5ish |
| 05:42.07 | Qwell | evening is until about 10 |
| 05:43.09 | lehel | here is 8:42 am, and i'd sleep 4 hours.. my eyes just want to close |
| 05:43.16 | jbAU | I have an interesting problem, I want to be able to have Asterisk tell me who is calling before I pick up the phone. I've been playing with the IVR and I'd like to throw up an incoming caller's answers on a web interface or something so that the person taking the call can see what they've pressed so far.... does anyone have any idea's which way I should go? |
| 05:43.37 | Qwell | jbAU: dialplan logic, and an agi perhaps |
| 05:43.47 | shido6 | yep |
| 05:43.55 | shido6 | maybe even alsa/oss |
| 05:44.02 | shido6 | to announce through the soundcard in the device |
| 05:44.06 | jbAU | Qwell: indeed - the logic is fair enough, it's just getting those details on a PC is the interesting one |
| 05:44.15 | shido6 | "incoming call from $CALLERID" |
| 05:44.29 | Qwell | or, you know what would be really neat? |
| 05:44.32 | shido6 | "automagically tellin them to muck off" |
| 05:44.44 | Qwell | a dedicated device that can display the information in front of you |
| 05:44.46 | jbAU | shido6: mmm that's true, but I also thought it would be a good idea to see what they've put in the IVR so far |
| 05:44.50 | Qwell | I'd call it a...caller id display |
| 05:44.57 | Qwell | I bet I'd make millions |
| 05:45.09 | jbAU | Qwell: lol - yeah but I want to also show what they've answered in the IVR so far |
| 05:45.14 | Qwell | :p |
| 05:45.39 | jbAU | I wasn't quite clear with that first part - it's jot just caller id display |
| 05:45.45 | jbAU | jot = not |
| 05:46.34 | Qwell | certainly you could write some sort of "dashboard", to show all of that |
| 05:47.02 | mutilator | anyone an md? write me a prescription for allegra? |
| 05:47.06 | jbAU | hrm, like have the asterisk system write to a DB and have the dashboard just sit there and poll it |
| 05:47.23 | mutilator | .. or get me like 200 samples of it |
| 05:47.24 | Qwell | jbAU: the agi could send a notification the users system |
| 05:47.31 | Qwell | to the |
| 05:48.08 | *** join/#asterisk outtolunc (~me@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
| 05:51.48 | Cresl1n | woohoo |
| 05:52.06 | shido6 | hey qwell |
| 05:52.12 | Qwell | yeah? |
| 05:57.30 | shido6 | what is a qwell? |
| 05:57.37 | shido6 | whered the name come from? |
| 05:57.43 | Qwell | dunno |
| 05:58.31 | Qwell | Kwell is a hair lice treatment, I've found out |
| 06:03.18 | jbAU | that sounds swell |
| 06:04.31 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 06:05.22 | *** join/#asterisk Maksim (~max@213.142.207.20) |
| 06:08.45 | *** join/#asterisk xbmodder (~xbmodder@dsl092-015-136.sfo1.dsl.speakeasy.net) |
| 06:08.48 | xbmodder | anyone home? |
| 06:09.26 | Cresl1n | Ack |
| 06:10.25 | *** join/#asterisk SkyNetishere (sdsdsd@66.195.185.34) |
| 06:10.36 | SkyNetishere | hi |
| 06:10.50 | xbmodder | ok |
| 06:10.56 | xbmodder | Outgoing calls are not working |
| 06:11.04 | xbmodder | How do I make asterisk work? |
| 06:11.23 | Qwell | is that a question? |
| 06:11.27 | Smi|k | reboot |
| 06:11.31 | Smi|k | I heard if you reboot it works |
| 06:11.35 | SkyNetishere | I want to catch all invalid extensions and send the to Say(${EXTEN}) is not a valid number for assistance press ..... |
| 06:11.37 | xbmodder | how do I make outgoing calls work. |
| 06:11.48 | Qwell | xbmodder: by rtfming, and asking a better question |
| 06:11.49 | SkyNetishere | the 'i' doesnt seem to catch 404s |
| 06:12.00 | Cresl1n | xbmodder: that doesn't help a whole lot |
| 06:12.01 | Qwell | xbmodder: That means absolutely nothing, unless you qualify it...alot |
| 06:12.16 | xbmodder | I read the manual |
| 06:12.22 | Smi|k | is there anything more stable then astrisk with an API for handling of calls? |
| 06:12.37 | xbmodder | I set everything up to the specs given in the asterisk startup |
| 06:12.38 | cochi | *meep* wrong channel Smi|k |
| 06:12.42 | cochi | ;) |
| 06:12.45 | Cresl1n | Smi|k: what are you talking about? |
| 06:12.52 | xbmodder | maybe its outgoing routing |
| 06:12.58 | xbmodder | how do I do outgoing routing? |
| 06:13.04 | Smi|k | really though, I run a company, and need solid phone service, but want to have my own custom interface on a web site for each user |
| 06:13.06 | SkyNetishere | Smi|k Realtime gives you a good api... |
| 06:13.11 | Smi|k | create my own software frontend |
| 06:13.17 | Smi|k | so I look at asterisk |
| 06:13.29 | SkyNetishere | Smi|k Realtime + apache + php (w/mysql) poof |
| 06:13.32 | Smi|k | but there are just so many problems posted all the time I worry one day down and I'll loose so much business |
| 06:13.48 | Smi|k | is Realtime asterisk or something similar? I'm getting a full pri t1 |
| 06:14.08 | Cresl1n | Smi|k: Asterisk is probably going to be the closest solution to that problem that you'll be able to find |
| 06:14.10 | SkyNetishere | Smi|k Realtime is part of the HEAD d/l asterisk via cvs |
| 06:14.17 | xbmodder | I am losing my mind. |
| 06:14.28 | Cresl1n | eh? |
| 06:14.30 | Qwell | xbmodder: |
| 06:14.31 | cochi | xbmodder you already did |
| 06:14.32 | Qwell | ~docs |
| 06:14.51 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 06:14.51 | Qwell | read that |
| 06:14.51 | xbmodder | ~docs |
| 06:15.06 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 06:15.07 | xbmodder | I read them... |
| 06:15.07 | xbmodder | I think I did. |
| 06:15.07 | Qwell | there |
| 06:15.07 | Qwell | 4th one |
| 06:15.07 | Qwell | actually, read all of them |
| 06:15.15 | Smi|k | I have a guy whose done programming for me before who hacks and hooks into C on a low level to integrate programs |
| 06:15.22 | SkyNetishere | Smi|k try http://www.voip-info.org/wiki-Asterisk+RealTime |
| 06:15.39 | Smi|k | is the asterisk code good enough and solid enough that if he creates a program next asterisk version will still be compatible |
| 06:15.46 | SkyNetishere | Smi|k i dont see a reason to hack it to much... |
| 06:15.50 | Smi|k | I need each desk to be able to create, route, transfer, hold etc..etc.. calls |
| 06:16.01 | Smi|k | SkyNetishere, not hack, I just mean hook right into low level C |
| 06:16.08 | SkyNetishere | Smi|k if you hack software... it probably wont be compatible with the next version anyway |
| 06:16.10 | Smi|k | to perform each function needed so it is done right |
| 06:16.14 | cochi | (i know some -very- big providers use asterisk. so i'd guess * is "stable" enough) |
| 06:16.30 | SkyNetishere | very very big..... |
| 06:16.31 | Smi|k | I know some very big providers who had phone service DOWN for like 5 days straight |
| 06:16.45 | Smi|k | that would kill me |
| 06:16.48 | SkyNetishere | i think pennyphone uses asterisk |
| 06:17.10 | cochi | i think web.de does so, too |
| 06:17.29 | cochi | my fiancee is regularly chatting with the former it director there, mentioned sth like that |
| 06:17.42 | SkyNetishere | I think i am using the 'i' extension wrong... is it just for after answer or can it be used to catch 404s (or how do you catch 404s) |
| 06:18.04 | SkyNetishere | (without _XXX etc... i want it to evaluate after looking for a real ext) |
| 06:18.16 | Cresl1n | I want to buy a .xxx domain |
| 06:18.20 | Cresl1n | just for kicks |
| 06:18.38 | Qwell | Cresl1n: expensive kick |
| 06:18.39 | SkyNetishere | like thissu.xxx :) prolly taken |
| 06:18.45 | Qwell | $60 or so I heard |
| 06:18.51 | Cresl1n | yeah |
| 06:19.00 | Cresl1n | matto was telling me about it today |
| 06:19.47 | twisted | xxx.xxx |
| 06:20.10 | twisted | heh, you could be a smartass, get that, and create xxx.xxx.xxx.xxx |
| 06:20.15 | Smi|k | so I want to use low cost VOIP phones |
| 06:20.21 | Qwell | twisted: heh |
| 06:20.22 | Smi|k | and have functions as if they are $500 voip phones |
| 06:20.34 | SkyNetishere | Smi|k grandstream are cheap |
| 06:20.36 | Smi|k | so the functions are on the computer, but the voip phone is $60-$70 basic one |
| 06:20.39 | SkyNetishere | <$80 |
| 06:20.47 | Qwell | twisted: get things like 241.xxx, make IP addresses |
| 06:20.55 | Smi|k | on the computer I want stuff like clicking the name of the person to transfer call to etc..etc.. |
| 06:21.09 | twisted | Qwell, well, sure, but i was thinking anywhere in documentation, you'd get free advertising |
| 06:21.14 | Qwell | heh |
| 06:21.32 | Qwell | especially if documents started underlining .xxx urls |
| 06:21.33 | SkyNetishere | Smi|k you might want to just use softphones then |
| 06:21.45 | SkyNetishere | good headset and poof |
| 06:21.51 | xbmodder | I think I did. |
| 06:21.52 | xbmodder | ~docs |
| 06:22.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 06:22.05 | Qwell | <a href="http://xxx.xxx.xxx.xxx">blah</a> |
| 06:22.48 | SkyNetishere | no ideas on my 404 catch prob? |
| 06:23.40 | Smi|k | fgjuo |
| 06:24.01 | Smi|k | umm, no, really do not want ip phones, they will eat computer CPU and computer will eat phone bandwidth and it just creates issues |
| 06:24.04 | SkyNetishere | how does one guarentee that a _XXXXXXX would be evaluated last? (I am using HEAD/Realtime for my extensions) |
| 06:24.18 | SkyNetishere | Smi|k u say so |
| 06:24.23 | Smi|k | I'd like to have the low cost $60 ip phone, with software that mimics a $500 IP phones functions on the screen of my computer |
| 06:24.27 | Qwell | SkyNetishere: include two different contexts |
| 06:24.41 | Qwell | one with all the stuff you want to match, and the other with the _XXXXXXX |
| 06:24.51 | SkyNetishere | Qwell do contexts have priorities? |
| 06:24.57 | Smi|k | windows pc's are at they desk, cant have calls hang up when computers reboot/crash |
| 06:24.58 | Qwell | SkyNetishere: the order they're included |
| 06:25.01 | Qwell | afaik |
| 06:25.16 | twisted | hey Cresl1n, when are .xxx domains supposed to go up for sale? |
| 06:25.29 | SkyNetishere | Qwell gotcha... I didnt see a mechanism for include in Realtime... thats only in extensions.conf? |
| 06:25.36 | Qwell | no clue |
| 06:25.45 | Smi|k | but has anyone kind of copied the functions of a nice $500+ voip phone on a computer program, and made it work with asterisk and desktop $60 voip phone |
| 06:25.47 | SkyNetishere | file is cool... |
| 06:26.34 | SkyNetishere | Smi|k the problem is going to be integrating a cheap phone with a computer... the phone does the actual sip connection to asterisk... you dont need to hack asterisk you need to hack your phone (or use a softphone) |
| 06:26.35 | Qwell | Smi|k: often, the reason the phones are $60, is because they are cheap and flimsy |
| 06:27.49 | Smi|k | the phone does not integrate with the computer |
| 06:28.13 | Smi|k | the computer tells asterisk route the call from phone (IP HERE) to phone (DID HERE) |
| 06:28.17 | Smi|k | or from IP to IP |
| 06:28.19 | Smi|k | etc..etc.. |
| 06:28.26 | Smi|k | the phone shouldnt need to have any part of the routing |
| 06:28.33 | SkyNetishere | my softphone doesnt use much processor I just placed a call and it used between %3-%5 |
| 06:28.38 | Smi|k | the high end phones tell asterisk server to route the call somewher else |
| 06:28.45 | Smi|k | but a computer desktop can do that too |
| 06:29.02 | Smi|k | SkyNetishere, winxp is weird with memory usage |
| 06:29.17 | Smi|k | could look up customer record and use 98% cpu or memory or whatever and phone call gets laggy |
| 06:30.01 | Smi|k | it just makes no sense to me to have $500 IP phones loaded with features next to computers that could handle the actual commanding of asterisk NO PROBLEM with virtually no CPU |
| 06:30.09 | SkyNetishere | Smi|k I dont know how but it would be cool to see... as far as I am aware SIP would talk to asterisk... What protocol/client do you use to tell asterisk to interupt a call and transfer etc... |
| 06:30.16 | twisted | damn |
| 06:30.24 | twisted | someone's already pre-registered xxx.xxx |
| 06:30.33 | Smi|k | then the phone is there should the computer be down you can still receive calls |
| 06:30.57 | SkyNetishere | how does the pc talk to asterisk? |
| 06:31.09 | Smi|k | SkyNetishere, why should asterisk care if you are talking or not? it cant actually listen to the calls |
| 06:31.19 | twisted | holy shit .xxx's are expensive |
| 06:31.20 | twisted | Service and Fee's:Number of DomainsLine Total |
| 06:31.20 | twisted | Service Fee:15.00 |
| 06:31.20 | twisted | Funding Amount:175.00 |
| 06:31.20 | twisted | Total Amount Due: |
| 06:31.20 | twisted | 80.00 |
| 06:31.30 | Smi|k | SkyNetishere, its just telling asterisk to receive SIP data commands from computer |
| 06:31.35 | Smi|k | its even the same IP address |
| 06:31.36 | cochi | can't it? thought so, esp as there're ways of recording the calls obviously |
| 06:31.42 | SkyNetishere | hmmm |
| 06:31.43 | cochi | in addition IP wasn't p2p, was it |
| 06:31.48 | Smi|k | see what I mean? |
| 06:32.35 | cochi | IP=>SIP |
| 06:32.41 | Smi|k | ? |
| 06:32.42 | SkyNetishere | Smi|k not really... sip is session based (hence the name) so you would have to have seperate sessions (one for pc one for phone)... i am not saying it cant be done... i am interested... i wouldnt mind figuring it out too... |
| 06:32.59 | Smi|k | hrm... |
| 06:33.07 | Smi|k | thats why I thought linking into asterisk on a lower level C |
| 06:33.21 | Smi|k | so a client runs to receive commands from the PC |
| 06:33.24 | SkyNetishere | oh... so a seperate control protocol... |
| 06:33.26 | Smi|k | makes asterisk think the command is from ip phone |
| 06:33.31 | SkyNetishere | it would be doable i guess. |
| 06:33.53 | SkyNetishere | seems like a lot of code mangling... |
| 06:33.58 | Smi|k | I assume all sip data commands are handled by one particular module of asterisk which pulls the commands apart from the voice in the protocol |
| 06:34.05 | SkyNetishere | do it and open source it i will d/l :) |
| 06:34.11 | Smi|k | so you modify that module to receive the 2nd input |
| 06:34.34 | SkyNetishere | somehow you would at least need to keep track of the callid though |
| 06:34.39 | Smi|k | and then its only one small module that needs to be keept up to date, and probably that module does not change often as SIP is a standard protocol |
| 06:34.58 | cochi | esp standards tend to change quickly ;) |
| 06:35.06 | Smi|k | it LISTENS to sip data cent to phone (caller ID, ani, etc) |
| 06:35.10 | SkyNetishere | you could prefix the callid with some identifier per phone... assign the client to that code and you should be able to see all the calls to that id... |
| 06:35.18 | Smi|k | and it TALKS making asteirsk think it is the phone |
| 06:35.49 | Smi|k | and since its from came IP address, you use ip address as identifier |
| 06:35.53 | SkyNetishere | perhaps someone has already done it... have you googled for it? |
| 06:36.03 | Smi|k | I dont know what to google for, I dont know any programming languages |
| 06:36.17 | SkyNetishere | try 'asterisk remote control' :) |
| 06:36.19 | Smi|k | so I'm looking for someone who knows the language to use to know what I'm talking about and google the right thing to find it :) |
| 06:36.36 | Smi|k | I have found some stupid apps that connect to some admin interface |
| 06:36.44 | SkyNetishere | or 'asterisk virtual' or something like that |
| 06:36.53 | Smi|k | so you install asterisk, then the adminc ontrol thing, and then on top of that is some other thing all together |
| 06:37.12 | SkyNetishere | it would b cool to give to my customers... |
| 06:37.13 | Smi|k | is SIP secure? |
| 06:37.22 | SkyNetishere | Smi|k define secure |
| 06:37.29 | Smi|k | point taken... |
| 06:38.06 | SkyNetishere | isnt g711a secure? heck i never thought of it |
| 06:38.23 | SkyNetishere | encrypted... |
| 06:38.33 | |Vulture| | no... |
| 06:38.42 | Smi|k | ms |
| 06:38.42 | Smi|k | g |
| 06:38.46 | SkyNetishere | i bet anything a cheap $60 phone wont have the ponies to do encryption along with codec |
| 06:38.53 | *** join/#asterisk bjohnson (~bjohnson@ip235-181.tor.istop.com) |
| 06:39.21 | |Vulture| | VPN between 2 * is the only thing I can think that would be secure |
| 06:39.25 | cochi | afaik nothing concerned SIP is encrypted at all. at least i read that some days ago in regards to the "cons" of sip |
| 06:39.39 | twisted | you can do srtp |
| 06:39.54 | Smi|k | help me understand sip |
| 06:40.01 | Smi|k | is it a protocol on the network cables |
| 06:40.05 | tuxinator_linuxM | Hey twisted |
| 06:40.07 | Smi|k | or is it tcpip with sip running on top |
| 06:40.10 | Smi|k | what layer is it on |
| 06:40.16 | twisted | uhm |
| 06:40.46 | twisted | sip is a protocol |
| 06:40.46 | Qwell | Smi|k: same as http, etc |
| 06:40.46 | Smi|k | I see |
| 06:40.46 | cochi | app-level. situated over tcpip |
| 06:40.46 | twisted | that should have been a given |
| 06:40.46 | SkyNetishere | sip=Session Initialazation Protocol (pardon spelling mistakes) |
| 06:40.46 | Qwell | physical layer ;] |
| 06:40.52 | cochi | ;)) |
| 06:40.58 | cochi | "yeah that's our sip cable here" |
| 06:41.01 | SkyNetishere | what are sip alternatives (save iax) h323? argh |
| 06:41.07 | Smi|k | so what if 2 sip phones are hooked in a line |
| 06:41.16 | Qwell | Smi|k: what if? |
| 06:41.18 | twisted | wtf? |
| 06:41.18 | Smi|k | asterisk->sip phone -> sip phone |
| 06:41.25 | Smi|k | network in, network out, network in |
| 06:41.25 | SkyNetishere | Smi|k they are session based |
| 06:41.34 | SkyNetishere | Ok now i am really confused |
| 06:41.38 | Smi|k | and the session is constant |
| 06:42.03 | twisted | Smi|k, go read up on ethernet and tcp/ip |
| 06:42.03 | SkyNetishere | you can have multiple sip clients sharing an ip |
| 06:42.03 | cochi | welcome to my world, sky ;) |
| 06:42.03 | Qwell | and SIP |
| 06:42.03 | SkyNetishere | session by definition is unique |
| 06:42.09 | SkyNetishere | :) |
| 06:42.12 | Smi|k | yes |
| 06:42.12 | Smi|k | ok |
| 06:42.19 | SkyNetishere | i got dinner anyway.... ttys |
| 06:42.20 | Smi|k | so the sip protocol is not secure, and is session based |
| 06:42.34 | Smi|k | so there is no reason the computer cant send commands |
| 06:42.35 | twisted | right, but sip will facilitate for srtp |
| 06:42.53 | twisted | but the sip communication itself (which does not carry the audio) is insecure |
| 06:43.04 | Smi|k | sip does NOT carry the audio at all? |
| 06:43.09 | twisted | no |
| 06:43.10 | Qwell | twisted: srtp? |
| 06:43.15 | twisted | srtp == secure rtp |
| 06:43.17 | Smi|k | I want the SIP session to go to the computer |
| 06:43.18 | Qwell | figured |
| 06:43.20 | Smi|k | and the audio to go to the phone |
| 06:43.26 | Qwell | twisted: sslified? |
| 06:43.27 | Smi|k | can they be split? |
| 06:43.32 | twisted | Smi|k, uhm, sure |
| 06:43.35 | Smi|k | how? |
| 06:43.41 | twisted | Smi|k, go read up on sip |
| 06:43.50 | Qwell | won't the phone need the SIP stuff too? |
| 06:43.59 | Qwell | like, "hey buddy, ring" |
| 06:44.04 | twisted | Qwell, yea, unless your phone will blindly accept rtp |
| 06:44.23 | Smi|k | so I could have the features of a $500 IP phone on my computer |
| 06:44.28 | Smi|k | and use a $60 ip phone on the desk |
| 06:44.47 | twisted | not exactly |
| 06:44.58 | twisted | but using asterisk an the manager interface, sure. |
| 06:45.03 | Smi|k | why not? like what is actually preventing that |
| 06:45.24 | twisted | YOU NEED TO READ UP ON SIP |
| 06:45.34 | twisted | and I need sleep. |
| 06:45.36 | twisted | gnite |
| 06:45.45 | cochi | (that guy's quick) |
| 06:47.20 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 06:48.26 | Smi|k | so manager interface does everyhting a $500 phone would basicly |
| 06:48.43 | Qwell | except last for a while |
| 06:48.56 | Qwell | combined with a cheap phone, that is |
| 06:49.44 | twisted | okay, here's the quick jist of it |
| 06:50.06 | twisted | SIP sets up the calls, directs the audio (media) to where it is going to go, negotiates the codecs involved |
| 06:50.24 | cochi | you're sleeping twisted ;) |
| 06:50.24 | Qwell | media in the above case, being the rtp stream |
| 06:50.33 | twisted | the media (rtp) is then sent to the destination(s) defined in the sip headers |
| 06:51.16 | cochi | mh so source + destination phone get call routing stuff via sip from *. and communicate with each other via rtp? |
| 06:51.19 | twisted | the reason I say you can split the audio and control is because you can. you can send all your control packets to a computer, and direct the audio streams to/from a separate device, as long as that device will accept/transmit blind rtp |
| 06:51.39 | twisted | rtp == audio |
| 06:51.49 | twisted | (and sometimes dtmf) |
| 06:52.02 | twisted | but all of that is contained in the SDP payload of the SIP message |
| 06:52.17 | Qwell | twisted: The third party computer could send new SIP packets to the phone though, right? |
| 06:52.28 | Qwell | and just say, "expect your rtp from this IP" |
| 06:52.32 | twisted | Qwell, sure |
| 06:52.35 | twisted | Qwell, that's what asterisk does |
| 06:52.43 | Qwell | I see |
| 06:52.50 | Qwell | So, the rtp generally comes from the other end? |
| 06:52.56 | Qwell | (assuming no transcoding) |
| 06:52.57 | twisted | rtp requires two paths |
| 06:53.08 | twisted | the sending and receiving |
| 06:53.14 | twisted | it's asyncronous |
| 06:53.20 | Qwell | so it could receive from one place, and send to another? |
| 06:53.25 | twisted | exactly |
| 06:53.34 | Smi|k | it should be able to receive sip data from TWO places, and send it to TWO places |
| 06:53.36 | Smi|k | since it is not secure |
| 06:53.48 | Smi|k | i.e. it doesnt know where its from, part of not being secure |
| 06:53.52 | Qwell | Smi|k: what would that solve though, besides having two calls? |
| 06:53.59 | Smi|k | it would still be ONE call |
| 06:54.04 | twisted | Smi|k, your definition of insecure is wrong |
| 06:54.06 | Smi|k | sip data is just the data that routes the call around |
| 06:54.10 | twisted | Smi|k, insecure means non-encrypted |
| 06:54.23 | twisted | sip data does contain origination/destination data |
| 06:54.24 | Smi|k | and? |
| 06:54.35 | Smi|k | reproduce it, grab it |
| 06:54.37 | twisted | as well as (usually) all paths/routes involved |
| 06:54.48 | Smi|k | if destination data is all you need then no need to ever encrypt :) |
| 06:55.03 | twisted | but that's not all you need |
| 06:55.10 | Smi|k | paths routes are the same as phone and cpu are same ip |
| 06:55.17 | twisted | you need to know the routing path as well |
| 06:55.27 | Smi|k | do explain |
| 06:55.30 | Qwell | Why would you give two devices the same IP/ |
| 06:55.38 | twisted | because the messages will need to follow that path (generally) unless they are strict routed directly back to the origin |
| 06:55.45 | Qwell | That won't do much besides screw up your ARP cache |
| 06:55.48 | twisted | and in tcp/ip you cannot have two devices the same IP |
| 06:56.05 | Qwell | twisted: you *can*, but... |
| 06:56.11 | Smi|k | does the *. server care which ip it gets its sip command from? |
| 06:56.12 | twisted | (at least not on the same physical network) |
| 06:56.15 | Smi|k | or only which session ID |
| 06:56.21 | twisted | Smi|k, both. |
| 06:56.22 | Qwell | twisted: you know as well as I do what would happen... |
| 06:56.41 | Smi|k | hrm |
| 06:56.46 | twisted | the session ID's are generated at the origin |
| 06:57.15 | Smi|k | and then sent out via. ip to something that receives it, and sends back the sort of "ok im here" id |
| 06:57.16 | twisted | the protocol itself is really quite complex |
| 06:57.29 | Smi|k | then its session + okimhere id |
| 06:57.34 | twisted | which is why i recommend you read up on it |
| 06:57.34 | Smi|k | and they talk |
| 06:57.40 | twisted | not quite that simple. |
| 06:58.26 | twisted | normal call setup consists of anywhere between 3-5 messages between two endpoings |
| 06:58.30 | twisted | *endpoints |
| 06:59.00 | twisted | (and by messages i mean packets full of data) |
| 06:59.05 | Smi|k | how about when the command to "transfer call to x2056" is sent |
| 06:59.14 | twisted | there is no command for that |
| 06:59.31 | twisted | you have to use a softswitch/router/proxy to reinvite the call |
| 06:59.53 | Smi|k | dont the snom phones etc..etc.. allow you to transfer call to another extension? |
| 07:00.23 | twisted | yes, but they send an invite with the current media back to the server, which then also reinvites the leg somewhere else |
| 07:00.45 | Qwell | its simply complex |
| 07:00.47 | twisted | the best piece of advice I can offer you is to GO READ UP ON SIP |
| 07:00.50 | Smi|k | so why cant that particular invite be sent from the PC instead? |
| 07:01.02 | twisted | because that's not in the call ath |
| 07:01.03 | twisted | er path |
| 07:01.05 | Smi|k | twisted, because hours of reading and its so complex I am right back where I started |
| 07:01.16 | Qwell | its so complex its simple |
| 07:01.24 | twisted | lol |
| 07:01.30 | Smi|k | why has no one done what I need though |
| 07:01.36 | Qwell | twisted: got a better way of describing it? :) |
| 07:01.58 | twisted | Smi|k, because you haven't paid anyone to do it |
| 07:02.18 | twisted | and your description of what you need is really bizzare. |
| 07:02.26 | Smi|k | twisted, I'll do that, but the problem is I'm not being understood |
| 07:02.28 | Qwell | needlessly complex |
| 07:02.38 | Smi|k | what I want is increadibly simple and only makes sense |
| 07:02.50 | twisted | yeah, in a sales-pitch sort of way |
| 07:03.03 | Smi|k | high resolution XGA monitor on my desktop, snom screen phone has a 2" x 3" black and white LCD |
| 07:03.11 | twisted | OMFG! |
| 07:03.16 | twisted | imagine that! |
| 07:03.23 | twisted | anyway, i'm really off to bed now |
| 07:03.25 | twisted | g'nite |
| 07:03.29 | Smi|k | to take advantage of the monitor I'd like the data that would display on snom to display on XGA |
| 07:03.56 | Smi|k | twisted, know of anyone in this channel who knows SIP protocol and * well enough to pull this off? |
| 07:04.05 | cochi | that's not "take advantage". that'd be if you coupled CRM+CTI |
| 07:04.32 | Smi|k | imagine pressing * on your keyboard |
| 07:04.41 | Smi|k | list of 30 names pops up on phone |
| 07:04.48 | Smi|k | you click one, and your call is gone, transferred away |
| 07:04.59 | Smi|k | the names show up as red and green based on whose on the phone |
| 07:05.25 | Smi|k | I just dont see how it doesnt already exist, lol |
| 07:06.24 | cochi | classical CTI stuff ;) |
| 07:06.45 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
| 07:06.54 | Smi|k | does asterisk have some kind of CTI link? |
| 07:07.44 | Smi|k | maybe all I need is the asterisk@home that has sugarcrm as part of the build, with a new frontend interface for sugar |
| 07:07.47 | cochi | guess so, api's big. but we're having simply the prob of two devices having to share one session (if you do transfer via SIP) or to develop a control protocol for the computer-app which does the think on * |
| 07:08.16 | cochi | i imagine softphones like eyebeam can already transfer that way, they also got friendlist+presence protocol in it |
| 07:08.21 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 07:08.41 | Smi|k | yep |
| 07:08.52 | RoyK | <PROTECTED> |
| 07:08.59 | cochi | don't really get why you're so against softphones, maybe explain? |
| 07:09.10 | Smi|k | on the phone, computer crashes |
| 07:09.17 | Smi|k | its happened to me maybe 20-30 times this year |
| 07:09.24 | Qwell | Thats why we don't use Windows |
| 07:09.26 | Qwell | :p |
| 07:09.29 | Smi|k | call drops and enough $ is lost to buy the snom phone |
| 07:09.34 | cochi | ehm to be honest, i'd dig into your installation then rather ;) |
| 07:09.48 | Smi|k | could happen to anyone in the office, then there is 10 people |
| 07:09.59 | cochi | but if the computer crashes which got the transfer + crm on it, same is |
| 07:10.10 | Smi|k | cochi, really? |
| 07:10.11 | cochi | and without crm at least your money is lost anyway |
| 07:10.24 | Smi|k | if I'm on the phone |
| 07:10.35 | cochi | well if you can just phone and not enter/get data or so... you can't put orders, nor give info |
| 07:10.39 | Smi|k | and say mm, these computers are so slow, as its reboothing |
| 07:10.42 | Smi|k | or work off a paper |
| 07:10.44 | Smi|k | or make small chat |
| 07:10.51 | Smi|k | VS. phone call instantly disconnects |
| 07:11.11 | Smi|k | I can get data without a computer |
| 07:11.15 | cochi | mh k. maybe i'm just too pc centric ;) |
| 07:11.21 | Smi|k | pen/paper write credit card |
| 07:11.27 | Smi|k | boot computer, type it in, click charge |
| 07:11.28 | RoyK | Smi|k: you can do that in the dial plan, divert to someone else if no answer |
| 07:11.50 | cochi | he's not talking of no answer but of call drop |
| 07:11.50 | Smi|k | I just need a GUI on my computer to transfer the desk phone next to me to somewhere else |
| 07:12.09 | Smi|k | so I can use standard IP phone, various ip phones at various desks etc..etc.. |
| 07:12.15 | Qwell | Why not pay the slight extra, and get a phone with a transfer button? |
| 07:12.24 | Smi|k | but one uniform app on computers to control * |
| 07:12.27 | RoyK | cochi: call will not drop if PC is kaputt unless you've made a bad dialplan |
| 07:12.28 | Qwell | a reliable phone with a transfer button |
| 07:12.34 | Smi|k | Qwell, because a transfer button will NOT do what the computer can do |
| 07:12.50 | Smi|k | Qwell, I can build out extensive client lists, employee extensions, etc..etc.. on computer |
| 07:12.50 | Qwell | RoyK: they were discussing softphones before you came in |
| 07:13.16 | cochi | yup |
| 07:13.17 | Smi|k | on soft phone I can use transfer button and write names in small letters under a plastic screen on a white paper |
| 07:13.21 | Smi|k | in my mind that is OLD technology |
| 07:13.32 | Smi|k | why not run a simple cpu app that takes littely no processor or memory to route calls |
| 07:13.36 | Smi|k | and let the ip phone handle all else |
| 07:13.56 | Smi|k | keyboard = phone buttons, monitor = phone LCD, phone = phone |
| 07:14.04 | Qwell | I would personally hate using my computer to control a phone |
| 07:14.09 | cochi | well |
| 07:14.20 | cochi | i must admit that feature demand is far from being "esoteric" or so |
| 07:14.36 | Smi|k | because a keyboard IS buttons, its a whole board of them |
| 07:14.36 | cochi | the arguments're ok |
| 07:14.48 | Smi|k | and a monitor IS a display, 1280x1240 full color |
| 07:15.03 | Smi|k | software app I can continue to develop features on to surpass $500 ip phones |
| 07:15.14 | Smi|k | and if new ip phones come with new features I do not need to replace old hardware |
| 07:15.38 | Qwell | $50 says it wouldn't be released as open source |
| 07:15.47 | Smi|k | need some high speed c++ overlay for monitor |
| 07:15.57 | Smi|k | with a key on the keyboard when pressed it instantly overlays whatever is on screen |
| 07:16.17 | Smi|k | so without using mouse you tap a button, ahve the overlay with active text, select what you need, hit enter, overlay is gone, phone call is transfered |
| 07:16.26 | Qwell | Smi|k: Am I right? |
| 07:16.34 | Smi|k | Qwell, I just need it for me |
| 07:16.40 | Qwell | $50 more says it still wouldn't be, even if he got extensive (free) help from the FOSS community |
| 07:16.40 | Smi|k | could care less on anything else |
| 07:16.57 | Smi|k | I like to use what makes sense, I am not in the software/phone biz |
| 07:17.24 | Smi|k | we make ipod adapters that connect to car cd changer ports to trick the car into thinking the ipod is a factory cd changer |
| 07:17.27 | *** join/#asterisk Broesmeli (~broesme@195.65.2.68) |
| 07:17.28 | cochi | mh quick question around. doesnt the * manager API exactly do what he says? transfer existing calls etc? |
| 07:17.32 | Smi|k | hence the mentality of just making it work |
| 07:18.03 | Smi|k | the * manager api operates on top of asterisk |
| 07:18.16 | Smi|k | I fear using that and hooking into it with all this would create issues with many users at once |
| 07:18.19 | Smi|k | i.e. one for every desk |
| 07:18.31 | Smi|k | also changes with every release of * |
| 07:18.43 | Smi|k | if its pushed in on the SIP level then its done once, and is finished |
| 07:19.10 | cochi | mh |
| 07:19.34 | cochi | well but wasn't the quick usage of the manager API more benefit than trying for some time to do proprietary hacks? |
| 07:19.46 | cochi | in addition manager API uses cleartext commands. quick to change. |
| 07:19.52 | Smi|k | if it was stable for 10+ users |
| 07:19.54 | cochi | and you shouldt upgrade all the time anyway ;) |
| 07:19.58 | cochi | tried it with 10+ users? |
| 07:48.51 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 07:48.51 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 || Everyone test IAXtel! It should be happy now |
| 07:49.15 | SwK | you dont need a but se anymore... most houses have rj=11 customer test jacks in the demarv box now |
| 07:49.36 | opus__ | swk - yeah my neighborhood is like 1920 era |
| 07:49.43 | Zeeek | hey wait I might be able to help |
| 07:50.05 | Zeeek | you use SIP, IAX? |
| 07:50.13 | opus__ | SIP is fine, IAX sure |
| 07:50.35 | Zeeek | you know what? You can get an account with voipbuster for like $1 |
| 07:50.42 | opus__ | word |
| 07:50.51 | fenlander | or for free with voipuser.org |
| 07:51.00 | Zeeek | they have free calls to some countries |
| 07:51.26 | Zeeek | including UK |
| 07:51.33 | opus__ | hmmm. |
| 07:51.38 | opus__ | windows, lemme try it. |
| 07:51.41 | Zeeek | <PROTECTED> |
| 07:52.17 | newl | SwK: most countries do not. |
| 07:52.56 | Zeeek | opus__ you download their softphone, log in once to create an account |
| 07:53.07 | Zeeek | then you can use asterisk with that acct if you like |
| 07:53.15 | opus__ | really |
| 07:53.27 | Zeeek | I think you need to buy 1€ credit |
| 07:53.39 | opus__ | thats the elitest thing ever |
| 07:53.54 | Zeeek | then calls to many places are free |
| 07:55.26 | cochi | "Update : Outbound calls are currently capped to a maximum duration of 10 mins." <- you better talk quick opus |
| 07:55.38 | Zeeek | those are the free ones? |
| 07:55.46 | cochi | yup. is on the announcement |
| 07:55.49 | cochi | http://www.voipuser.org/forum_topic_328.html |
| 07:56.02 | Zeeek | skype is actually pretty reasonable including for world DID |
| 07:56.26 | cochi | ah voipbuster is another one *checking* |
| 07:56.51 | Zeeek | I'd use voiptalk to call the UK |
| 07:57.11 | cochi | i got sipgate for that purpose (got my own number in the uk that way) |
| 07:57.30 | Zeeek | those numbers are very expensive to the caller |
| 07:57.40 | cochi | nope. a real one. no 08xx |
| 07:57.42 | Zeeek | or have we been through this already? :) |
| 07:57.48 | Zeeek | ya |
| 07:58.04 | Makenshi | i got a 0121 from sipgate |
| 07:58.11 | Makenshi | though i still can't use dtmf with it :( |
| 07:58.15 | cochi | i got 01744 or so. St.Helens near Liverpool |
| 07:58.38 | Zeeek | opus__ Ping this: iax.voipbuster.com |
| 07:59.24 | Zeeek | oops, apparently not pingable |
| 07:59.32 | qwertyiop | opus__: i'm trying to waste my gossiptel credit, so i can remove all traces of it from my server, tell me the number and i'll try and put you through. |
| 08:00.11 | opus__ | cochi - can I call you to test this out |
| 08:00.12 | SkyNetishere | is there a fallback to Goto [extension] if extension doesnt exist the man says it always returns 0 even if it doesnt... |
| 08:00.20 | *** join/#asterisk jprao (~prakash@62.116.121.242) |
| 08:00.47 | opus__ | zeeek: yes no ping |
| 08:02.00 | jprao | hi |
| 08:03.51 | Zeeek | anyone know anything about digitmaps? |
| 08:03.54 | cochi | opus, bad idea here. but you can call my * testserver or so. get the "please authenticate" message ;) |
| 08:06.51 | *** join/#asterisk shahanbest (~shahanbes@203.143.48.2) |
| 08:07.14 | cochi | btw. voipbuster: " To counter the misuse of our network, and to reserve capacity for our genuine testers, free calls are limited to 1 minute per call." |
| 08:07.20 | cochi | so all those "free calls" are just crap |
| 08:07.38 | Zeeek | that was during the beta I think |
| 08:07.47 | cochi | apparently they still are in beta |
| 08:07.52 | cochi | at least i am now "beta tester" |
| 08:08.11 | Zeeek | thi is the new improved beta |
| 08:08.29 | cochi | "BUT: To counter the misuse of our network and to reserve capacity ..." <- see? |
| 08:08.29 | shahanbest | hi |
| 08:08.41 | Zeeek | to tell the truth, I commented it out of my iax.conf because they keep dropping into UNREACHABLE too often |
| 08:08.56 | Zeeek | so don't misuse their network :) |
| 08:09.03 | cochi | ;) |
| 08:12.05 | opus__ | zeeek - i think my Dial(IAX/user:password@iax.voipbuster.com/+${EXTEN}) line is busted |
| 08:12.12 | opus__ | do I have it wrong? |
| 08:12.19 | opus__ | <PROTECTED> |
| 08:12.20 | Zeeek | lemmie see |
| 08:12.56 | Zeeek | you must dial 00 in front |
| 08:12.56 | opus__ | it did dial cochi's test number fine though from windows |
| 08:12.59 | opus__ | ok |
| 08:13.32 | Zeeek | replace the + with 00 |
| 08:15.16 | opus__ | hmmm still problems |
| 08:15.17 | opus__ | <PROTECTED> |
| 08:16.21 | Zeeek | opus__ I use the format IAX2/myname@voipbuster/00${EXTEN} |
| 08:16.32 | Zeeek | and voipbuster is an entry in iax.conf |
| 08:16.36 | Zeeek | exit |
| 08:16.38 | Zeeek | oops |
| 08:17.06 | Zeeek | and allow=ulaw |
| 08:21.44 | *** join/#asterisk Mc_Tr (~Mc_Tr@bacterio.knet.es) |
| 08:22.36 | Mc_Tr | has anyone digium TE110P card installed? |
| 08:24.07 | Ahrimanes | i have a te410p |
| 08:24.55 | Mc_Tr | ahrimanes: where are you? |
| 08:25.14 | Ahrimanes | Mc_Tr: denmark |
| 08:25.21 | Mc_Tr | have you problem with te410p card? |
| 08:25.25 | Ahrimanes | Mc_Tr: but i'm not actively using it though |
| 08:25.36 | Mc_Tr | how has yoy zaptel.conf? |
| 08:25.49 | Mc_Tr | Ahrimanes, ok, thanks. |
| 08:26.14 | Mc_Tr | i'm having this error: NOTICE[1558]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 08:26.25 | Mc_Tr | and i don't know waht is this. |
| 08:26.36 | opus__ | zeeek - works line magic |
| 08:26.39 | opus__ | like magic |
| 08:27.55 | Zeeek | great |
| 08:28.05 | Zeeek | you got 10 minutes! |
| 08:28.39 | ptiggerdine | anyone played with rpm's and rebuilding zaptel ? |
| 08:30.11 | opus__ | hmm, net? or per call |
| 08:30.21 | Zeeek | dunno |
| 08:30.37 | opus__ | ok, now i will call somebody in london thanks!!! :) |
| 08:30.53 | qwertyiop | voipbuster only gives you 2 mins per call unless you add credit |
| 08:31.12 | Zeeek | opus__ did you put any money in at all? |
| 08:31.22 | opus__ | nope |
| 08:31.28 | Zeeek | ah. |
| 08:31.36 | Zeeek | drums stop. very bad. |
| 08:32.04 | Zeeek | I think I understand digitmaps now, anyway |
| 08:33.29 | opus__ | has problems dialing to london , germany fine. lame |
| 08:33.46 | Zeeek | they're in germany |
| 08:33.57 | Zeeek | you sure you got the right number? |
| 08:34.00 | opus__ | oh shit, |
| 08:34.05 | opus__ | 'you have run out of credits mate' |
| 08:34.12 | Zeeek | london numbers are often given is shitty international format |
| 08:34.13 | Ahrimanes | hehe |
| 08:34.34 | *** part/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 08:34.34 | cochi | o.O |
| 08:34.35 | Zeeek | you'll have to wait for the FWD XMAS promo :) |
| 08:34.51 | opus__ | no, i will just sign up with my provider tomorrow with ld |
| 08:34.54 | opus__ | i mean int |
| 08:35.05 | cochi | i registered with some new voip provider and got a month free with 250 minutes to pstn o.O |
| 08:35.19 | opus__ | they're in florida so i'll just call in a few hours when the sun hits that spot of the planet |
| 08:35.29 | Zeeek | which trailer park? |
| 08:35.39 | opus__ | looks like orlando, i lived there for two years |
| 08:35.48 | Zeeek | blew away last year |
| 08:35.51 | opus__ | nice place |
| 08:35.53 | opus__ | haha |
| 08:36.06 | opus__ | cochi nice |
| 08:36.50 | opus__ | now i can do jerkey boys world wide |
| 08:39.51 | *** join/#asterisk implicit (~bayan@209.80.0.43) |
| 08:40.05 | timecop | fuck |
| 08:40.08 | timecop | sip registration problems |
| 08:40.10 | timecop | in latest CVS |
| 08:40.16 | timecop | anyone else noticed? |
| 08:40.28 | timecop | to fwd no less |
| 08:40.32 | timecop | do these fuckers ever test anything |
| 08:40.32 | opus__ | try diffing chan_sip with previous version? |
| 08:40.34 | timecop | before committing it |
| 08:40.52 | timecop | sure, mine was about 10 revisions old |
| 08:40.52 | timecop | (= few weeks old) |
| 08:41.01 | opus__ | hmm |
| 08:41.13 | opus__ | does sipp connect fine? |
| 08:41.23 | lehel | people: what is to do? if in my apache2.conf User and Group is Asterisk.. everything is OK, but i have to change that |
| 08:41.43 | timecop | it also loads awfully fast now |
| 08:41.48 | timecop | something is wrong. |
| 08:45.43 | timecop | uh |
| 08:45.46 | timecop | something is really wrong. |
| 08:45.50 | timecop | it loads so fast. |
| 08:45.58 | timecop | h323 used to take like 10 seconds to load. |
| 08:48.31 | *** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146) |
| 08:50.33 | cochi | they removed all functionality to solve complaints about slow startups ;)) |
| 08:52.40 | Zeeek | good plan |
| 08:56.31 | squirrelv5 | hello people, my friend at overseas gave me his sip account in the format myfriend@theirdomain how do i call him? |
| 08:57.27 | Zeeek | Dial(SIP/yourfirend@theirdomain) ? |
| 08:57.54 | squirrelv5 | shoud i create an extension on my pbx? im using xten xlite |
| 08:58.10 | Zeeek | you can dial them directly on X-Lite |
| 08:58.27 | Zeeek | using the PC keyboard |
| 09:01.18 | squirrelv5 | can he call me with mynumber@mydomain.com format too? |
| 09:01.53 | Zeeek | if there is an 'yourname' extension in the context |
| 09:03.05 | squirrelv5 | u mean [myname] in extensions.conf? |
| 09:03.15 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-13.84-47-120.telecom.sk) |
| 09:03.19 | ellvis | hi people |
| 09:03.26 | Zeeek | is this someone who has an account on your * box? |
| 09:04.13 | squirrelv5 | no |
| 09:04.46 | Zeeek | so they would fall into the [default] context or whatever you've called it |
| 09:05.09 | Zeeek | and in that context, you woul dneed an EXTENSION yourname |
| 09:06.05 | squirrelv5 | so, this means i just have to give eg. 11111@mydomain.com as my number to my peers and all wud be fine? |
| 09:06.13 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@mol92-3-82-224-241-3.fbx.proxad.net) |
| 09:06.17 | Zeeek | try it |
| 09:08.09 | squirrelv5 | im confuse, how will the other end know that "mydomain.com" points to my asterisk server? does it need an MX record? like email? |
| 09:09.06 | Zeeek | does ping mydomain.com show the ip of your asterisk box? |
| 09:09.09 | Makenshi | squirrelv5, you need srv records |
| 09:09.14 | Makenshi | and preferable naptr records |
| 09:09.18 | Makenshi | *preferably |
| 09:09.57 | Makenshi | at a minimum, you need _sip._udp.domain.com. in addition, you should also have _sips._tcp and _sip._tcp |
| 09:10.10 | Makenshi | mx ought to be replaced by naptr and srv |
| 09:10.13 | Makenshi | it's so old |
| 09:10.19 | Zeeek | Makenshi I swore I tried this months ago and it worked, just having me@domain.com with a 'me' extension |
| 09:10.46 | Zeeek | I was logged in to the box though |
| 09:11.05 | squirrelv5 | so i need to config my DNS with those records? |
| 09:11.07 | Makenshi | i guess its because some sip clients will happily use an A record |
| 09:11.11 | Makenshi | squirrelv5, yes |
| 09:11.28 | Zeeek | Makenshi I believe this to be true... also srvlookup is an option in * |
| 09:11.32 | Makenshi | squirrelv5, there's a good guide here - http://mit.edu/sip/sip.edu/dns.shtml |
| 09:11.40 | qwertyiop | works fine with just a cname here |
| 09:11.41 | Makenshi | Zeeek, shame, it ought to be mandatory |
| 09:11.45 | squirrelv5 | thanks..ill take a look |
| 09:11.48 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 09:12.25 | squirrelv5 | hey, qwertyiop, what do u mean cname only? |
| 09:12.55 | Zeeek | qwertyiop what sip client? |
| 09:13.01 | squirrelv5 | but mydomain.com is already pointing to www, can i have multiple records for mydomain.com? |
| 09:13.08 | qwertyiop | xten |
| 09:13.17 | Zeeek | yes, that what I used for the test |
| 09:13.27 | Zeeek | plus as I said you can type the address in with X-Lite |
| 09:13.35 | Makenshi | Zeeek, it's like internet explorer in a way |
| 09:13.40 | Makenshi | Zeeek, ie will render any old crap |
| 09:13.41 | Zeeek | no don't go there :) |
| 09:13.47 | Makenshi | so people dont bother conforming to standards |
| 09:13.51 | Makenshi | and it gets out of hand |
| 09:14.11 | blop | is there any _iax txt/srv/naptr records supported? |
| 09:14.17 | Makenshi | blop, yes |
| 09:14.23 | *** join/#asterisk dandre (~dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:14.58 | Makenshi | iirc |
| 09:15.39 | Zeeek | "As a last resort if no SRV records are found a DNS A record request is sent for the domain" |
| 09:15.46 | Zeeek | so it be cool |
| 09:16.29 | Makenshi | i'd like to meet whoever thought that was a good idea |
| 09:16.40 | Zeeek | the mail must get through... |
| 09:17.07 | Makenshi | sure, if its that important, the admin would have set it up right! |
| 09:17.28 | Zeeek | that's one world view among many possible |
| 09:17.45 | squirrelv5 | hey guys, newbie in here, whom to follow? |
| 09:17.48 | Makenshi | the internet was founded on standards |
| 09:17.50 | Zeeek | to file in the chapter "Between Anal and Anarchy" |
| 09:18.32 | Makenshi | Zeeek, i bring you back to my point about internet explorer... |
| 09:18.49 | Zeeek | Makenshi although you are 100% (he said to diffuse a possible argument which has no good reason to happen) eveloution happens because of mutation |
| 09:19.04 | Makenshi | oh i understand there are other viewpoints :) |
| 09:19.08 | squirrelv5 | ok..standards...thanks |
| 09:19.12 | Makenshi | sorry, im not trying to be argumentative |
| 09:19.20 | Zeeek | so the bleeding edge giys who broke standards were "wrong" but they created a lot of stuff that are standards today |
| 09:19.54 | Zeeek | but let's all join hands and repeat FSCKING MS and MSIE MUST DIE! Because they indeed piss me off daily |
| 09:20.07 | Makenshi | :) |
| 09:20.40 | Zeeek | POlus I juts found out some work we had done doesn't display at all in Mozilla browsers. Like these MS guys think MSIE is still 99%... |
| 09:21.04 | Zeeek | I force myself to check our work in MSIE, these jerks should be doing the same with Moz |
| 09:23.13 | Zeeek | hourly troll: anyone going to Madrid next week ? |
| 09:23.54 | *** join/#asterisk waltmaine (~wreed@static-64-222-230-220.port.east.verizon.net) |
| 09:23.56 | Makenshi | sadly not |
| 09:24.34 | *** join/#asterisk potsboy (~chrisg@209.212.122.90) |
| 09:26.01 | *** part/#asterisk waltmaine (~wreed@static-64-222-230-220.port.east.verizon.net) |
| 09:26.54 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 09:27.02 | *** join/#asterisk waltmaine (~wreed@static-64-222-230-220.port.east.verizon.net) |
| 09:27.07 | *** part/#asterisk waltmaine (~wreed@static-64-222-230-220.port.east.verizon.net) |
| 09:31.13 | *** join/#asterisk meppl (mephisto@p54AAD95D.dip.t-dialin.net) |
| 09:33.12 | squirrelv5 | thanks ill try my config |
| 09:35.48 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
| 09:36.35 | *** join/#asterisk mcra (~alutiu@satlink.ro) |
| 09:39.05 | Zeeek | Figure of the week: (from Newsweek) 20 Billion Dollars |
| 09:39.39 | Zeeek | The amount of projected revenues from VoIP services in the US by 2009 |
| 09:40.28 | darkskiez | but when calls are free ? |
| 09:40.50 | Zeeek | 'calls' are old hat |
| 09:41.48 | *** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146) |
| 09:42.00 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 09:42.03 | fantomax1 | hi all |
| 09:42.06 | squirrelv5 | makensi, need help on ur recomendation |
| 09:42.32 | fantomax1 | can anyone tell me how to limit a call in term of time, I would like to limit calls to 30 min |
| 09:42.58 | fantomax1 | having * hanging up after 30 mins |
| 09:43.20 | Zeeek | fantomax1 show application dial |
| 09:43.27 | Zeeek | there is an option |
| 09:43.44 | fantomax1 | where ? |
| 09:43.56 | fantomax1 | Absolutetimeout ? |
| 09:44.03 | Zeeek | read what I typed |
| 09:44.14 | fantomax1 | where can i fing on wiki ? |
| 09:44.32 | qwertyiop | <PROTECTED> |
| 09:44.32 | qwertyiop | <PROTECTED> |
| 09:44.32 | qwertyiop | al. |
| 09:44.50 | Zeeek | SHOW APPLICATION DIAL |
| 09:44.54 | fantomax1 | ok :) |
| 09:44.56 | fantomax1 | thanks |
| 09:45.16 | *** join/#asterisk jixi (~damien@193.190.210.150) |
| 09:45.25 | Zeeek | (or get qwertyiop to enter the rest in) |
| 09:45.58 | qwertyiop | heh, that should do it. |
| 09:46.38 | RoyK | anyone using app_icd here? |
| 09:51.30 | squirrelv5 | haloo, i have a granstream touchtone 486, can i make it as my pstn gateway? |
| 09:51.47 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
| 09:52.38 | *** join/#asterisk mistral (mistral@jstevenson.plus.com) |
| 09:53.42 | Zeeek | <PROTECTED> |
| 09:54.34 | squirrelv5 | what does "not usable via SIP, only used as a fallback for power failure, etc" mean? |
| 09:54.47 | Zeeek | it means, no you can't use it as an FXO |
| 09:54.51 | Zeeek | from SIP |
| 09:54.55 | *** join/#asterisk elric (~m33p@ppp114-10.static.internode.on.net) |
| 09:55.51 | squirrelv5 | what do i need to make outbound pstn call? |
| 09:56.11 | squirrelv5 | we have an existing nortel pabx here in out office |
| 09:56.24 | Zeeek | http://www.voip-info.org/wiki-VOIP+Gateways |
| 09:57.29 | Zeeek | A lot of people like the Sipura 3000 |
| 09:57.58 | squirrelv5 | so my grandstream cant make outbound only inbound? correct? |
| 09:59.08 | Zeeek | it would appear so |
| 10:03.41 | squirrelv5 | im from the phils, anyone knows where can i buy sipura 3000 |
| 10:03.42 | Zeeek | or not |
| 10:04.04 | Zeeek | voxilla? |
| 10:07.03 | SkyNetishere | I dont think there is a fallback to Goto am i missing some really comprehensive doc? |
| 10:07.55 | Zeeek | what situation would produce an invalid context? |
| 10:08.22 | SkyNetishere | not on context (i my application) but no exten |
| 10:08.31 | Zeeek | ok, same question |
| 10:08.34 | *** join/#asterisk jprao (~prakash@62.116.121.242) |
| 10:09.18 | *** join/#asterisk GuERo (~gabriel@82.144.14.178) |
| 10:09.20 | GuERo | hi |
| 10:09.59 | GuERo | how I can remove agent/member in a queue with CLI commands ? |
| 10:10.10 | GuERo | is this posible? |
| 10:10.24 | SkyNetishere | each sip user will have thier own exten with a bunch of conditionals... if they dial 55 that is the same as 'speed dial' ... so _XX then you go to thier exten 'joesPhone' and look at ${DNID} and eval it to whatever num he set up on the website |
| 10:11.05 | SkyNetishere | if they havent set it up or I did something stupid i want to play a "you speed dial is not set up yet blah" |
| 10:11.12 | Zeeek | and the extensions aren't stored in astdb? |
| 10:11.28 | SkyNetishere | mysql (Realtime) |
| 10:11.29 | Zeeek | the speedial codes I mean |
| 10:11.43 | SkyNetishere | if they were then everyone would have the same speed dial |
| 10:11.49 | Zeeek | one of the many, many areas I know nothing about, realtime |
| 10:11.51 | SkyNetishere | or everyone would have to have a seperate context |
| 10:12.12 | SkyNetishere | at this level each line of extensions or sip.conf are just lines in a db |
| 10:12.14 | Zeeek | no the astdb would use a key for each phone callerid or each user |
| 10:12.19 | SkyNetishere | i am doing it to make php talk easy to confnig |
| 10:12.38 | SkyNetishere | where would the speed dials be stored extensions.conf? |
| 10:12.44 | Zeeek | astdb |
| 10:12.58 | Zeeek | SPeedials/2002/12="5551212" |
| 10:13.00 | SkyNetishere | astdb part of normal asterisk? |
| 10:13.05 | Zeeek | yes it is |
| 10:13.12 | SkyNetishere | hrmmm... i must rtfm |
| 10:13.14 | SkyNetishere | thx |
| 10:13.15 | Zeeek | database show |
| 10:14.02 | SkyNetishere | of course mine is empty save dundi/secret |
| 10:14.19 | SkyNetishere | can speeddials be used as transfer and 3way numbers/? |
| 10:14.20 | Zeeek | well it could ahve registered SIP or IAX2 clients in it |
| 10:14.47 | Zeeek | if the extension knows how to do that with logic, sure |
| 10:15.03 | SkyNetishere | oh... ok ... where do i input stuff in astdb? |
| 10:15.36 | RoyK | DB_put? |
| 10:15.41 | Zeeek | you start by reading one or more of the many available docs or examples on the web or the wiki or the mailing list or http://asteriskdocs.org |
| 10:16.09 | SkyNetishere | oh it has to be done with 'asterisk -r'? |
| 10:16.18 | Zeeek | read, read |
| 10:16.23 | SkyNetishere | okok :) |
| 10:16.30 | Zeeek | ~google asterisk astdb |
| 10:16.46 | RoyK | jbot: lart Zeeek for old time's sake |
| 10:17.08 | Zeeek | which will bring up the wiki entry and some other stuff |
| 10:17.40 | Zeeek | such as : http://www.voip-info.org/wiki-Asterisk+Database |
| 10:17.53 | SkyNetishere | :) i am there |
| 10:18.29 | Zeeek | this is not directed at you personally, but it's funny how even my gradma uses "google" as a verb nowadays and yet asterisk users haven't figured out how to search |
| 10:18.58 | SkyNetishere | i was unaware of the noun 'astdb' |
| 10:18.58 | Zeeek | my own best use is google: asterisk anythingyouwannaknow |
| 10:19.14 | Zeeek | more the reason to read a few intros |
| 10:19.35 | Zeeek | but indeed that stuff isn't easily guessable |
| 10:19.51 | Zeeek | astdb is one of the most powerful features for extensions |
| 10:19.52 | SkyNetishere | well i read all of /usr/src/asterisk/doc |
| 10:20.11 | SkyNetishere | grep astdb /usr/src/asterisk/doc/* turns up no matches |
| 10:20.19 | SkyNetishere | i trust inline docs to much |
| 10:20.22 | Zeeek | there are now a zillion newbie forums and some have examples of cookbook like stuff |
| 10:20.44 | SkyNetishere | true |
| 10:20.48 | Zeeek | another trick I used was typing show applications in CLI |
| 10:21.01 | squirrelv5 | what is gsm codec? gsm in celphones? |
| 10:21.03 | Zeeek | looking at that list, were there things I never heard of? What do they do? |
| 10:21.17 | SkyNetishere | HOLY CARP show applications rocks :) |
| 10:21.45 | squirrelv5 | what is gsm codec? gsm in celphones? |
| 10:22.10 | Zeeek | voipnewbie.comI seem to be selling a lot of show applications today - I'm almost out of them |
| 10:22.18 | squirrelv5 | any docs on nortel+* |
| 10:22.34 | Zeeek | google asterisk nortel ? |
| 10:22.58 | Zeeek | mailing list search google: site:lists.digium.com nortel |
| 10:23.44 | Zeeek | funny how all google searches turn up 10 pages of links |
| 10:24.06 | SkyNetishere | 2 bad it doesnt show syntax |
| 10:24.37 | SkyNetishere | oh show application wait |
| 10:24.39 | SkyNetishere | sweet |
| 10:26.09 | SkyNetishere | oh holy cow show application mysql problem bigtime solved |
| 10:26.16 | Zeeek | squirrelv5 someone here once told me that gsm is almost always in softphones while g723 is almost always in hardphones |
| 10:26.18 | SkyNetishere | manuals are just to wordy |
| 10:26.37 | SkyNetishere | hey btw do you non Realtime guys have a MYSQL command? |
| 10:27.08 | Zeeek | I don't |
| 10:27.20 | RoyK | SkyNetishere: there's one in asterisk-addons |
| 10:27.37 | SkyNetishere | oh i loaded the CDR maybe it came from that |
| 10:30.50 | SkyNetishere | ok here is a really dumb q that i will probly get kicked for :) when you pick up a sip line does the dialtone come from asterisk or is it generated by the phone (can asterisk detect an off-hook situation)? |
| 10:30.53 | *** join/#asterisk zoa (zoa@82.103.76.147) |
| 10:35.40 | RoyK | SkyNetishere: dial tone is generated locally. ringing indications etc are generated by asterisk |
| 10:37.36 | SkyNetishere | darn.. |
| 10:44.21 | RoyK | SkyNetishere: wot? |
| 10:45.45 | fantomax1 | thanks |
| 10:46.23 | *** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146) |
| 10:46.38 | squirrelv5 | how to test my * from outside world? |
| 10:47.11 | squirrelv5 | i mean calling an extension thats looks like coming from the outside |
| 10:47.59 | RoyK | what is outside? |
| 10:48.01 | RoyK | isdn? |
| 10:48.03 | RoyK | zap? |
| 10:48.06 | RoyK | sip? |
| 10:48.07 | RoyK | iax? |
| 10:48.10 | RoyK | htcpcp? |
| 10:48.10 | *** join/#asterisk outofjungle (~outofjung@61.17.134.58) |
| 10:48.12 | squirrelv5 | sip? and iax |
| 10:48.31 | squirrelv5 | can someone call me? |
| 10:49.25 | tuxinator_linuxM | I could, but it would only be giberish |
| 10:49.52 | squirrelv5 | i setup my * and my extension is 3511@hcg.com.ph |
| 10:49.54 | tuxinator_linuxM | actually I don't think I am even up for giberish |
| 10:50.34 | tuxinator_linuxM | They put me under today when I went to the doctor, I'm a bit out of it. |
| 10:50.49 | tuxinator_linuxM | sorry |
| 10:51.03 | squirrelv5 | anyone cud help me? |
| 10:52.18 | jerlique | is the commercial version of * from digum that same asterisk thats open source?? |
| 10:57.10 | *** join/#asterisk dln_cds (~viktor@59-114.moldova.md) |
| 11:00.47 | Zeeek | tuxinator_linuxM hahaha |
| 11:01.03 | RoyK | zoa: Guten Tag |
| 11:01.04 | Zeeek | *tnx for the chuckle - I was away |
| 11:01.28 | Zeeek | the commercial version comes with 1.5v batteries |
| 11:01.56 | RoyK | jerlique: yes |
| 11:02.02 | RoyK | jerlique: more or less, prolly |
| 11:02.10 | RoyK | jerlique: they have dual-licenced asterisk |
| 11:03.00 | Zeeek | dualing licenses... |
| 11:03.10 | Zeeek | dueling licenses... |
| 11:03.15 | tzanger | heh |
| 11:03.30 | tzanger | dah da da da da da dah dah.... <bankjo> dah da da da da da dah dah.... |
| 11:04.02 | tzanger | jerlique: think of ABE as CVS HEAD but feature-frozen |
| 11:04.18 | zoa | its not the same |
| 11:04.22 | tzanger | an alternative to STABLE but whose exact source is not available |
| 11:04.22 | zoa | its less than the open source |
| 11:04.27 | zoa | they stripped some things |
| 11:04.45 | Zeeek | pants? shirt? |
| 11:04.49 | tzanger | that's what I mean -- they cut out the more esoteric features and more problematic apps and said "THIS is what you're gonna get. No, yo ucan't have more, no you can't have it different." |
| 11:05.15 | Ahrimanes | so now it cant make coffee?! |
| 11:05.46 | tzanger | Ahrimanes: correct; it can't even make you toast anymore |
| 11:05.52 | Zeeek | dman. |
| 11:05.57 | Zeeek | damn |
| 11:06.13 | Ahrimanes | tzanger: that's just wrong.. |
| 11:06.16 | Zeeek | course the latest cv burns the toast anyway |
| 11:06.41 | Ahrimanes | Zeeek: puts coffee in the toaster and toast in the coffee filter... |
| 11:08.30 | tzanger | heh |
| 11:09.55 | Zeeek | why are there invariably two presentations I want to see at the same time? |
| 11:10.31 | *** join/#asterisk cypromis (~michael@83.149.70.59) |
| 11:11.05 | Ahrimanes | Zeeek: know the feeling |
| 11:11.34 | Zeeek | well I won't be going to astersik en espagnol :) |
| 11:11.55 | Ahrimanes | but why? |
| 11:11.56 | Ahrimanes | hehe |
| 11:12.45 | Zeeek | I'm now curious about Internet connex in that hotel |
| 11:13.42 | Ahrimanes | me too, very much so |
| 11:13.44 | Zeeek | looks like they would have highspeed in the room |
| 11:13.54 | Ahrimanes | but at what cost? |
| 11:14.22 | Zeeek | but the big quest is wifi - and with 800+ rooms how fast would thatbe saturated with asterisk calls :) |
| 11:14.46 | Ahrimanes | hehe well |
| 11:15.05 | Ahrimanes | they'd need many access points just to get coverage through all that concrete anyways.. |
| 11:16.08 | Zeeek | last time I was in Madrid, many, many years ago we got to the hotel at like midnight. There were three or four very nice looking women sitting in the lobby looking like they were waiting for someone |
| 11:16.37 | Ahrimanes | and..? |
| 11:16.40 | Zeeek | I asked one who she was waiting for. "You she smiled. 120000pesetas"" |
| 11:16.51 | Ahrimanes | eheheh |
| 11:16.55 | Zeeek | I'm guessing the amount |
| 11:17.01 | Zeeek | LUNCH HAS ARRIVED!!! |
| 11:17.09 | Ahrimanes | lunch has been eaten |
| 11:17.34 | *** join/#asterisk _BB_ (~bb@ip503d5751.speed.planet.nl) |
| 11:23.49 | gpearson | anyone have recommendations on IAX Service Provider for * |
| 11:25.13 | tzanger | nufone? |
| 11:25.42 | *** join/#asterisk chorix (alejandro1@host124.201-252-204.telecom.net.ar) |
| 11:38.53 | *** join/#asterisk outofjungle (~outofjung@61.17.134.58) |
| 11:49.06 | *** join/#asterisk stkn_ (nobody@stkn.developer.gentoo) |
| 11:53.03 | jerlique | thanks for the answers (had to run off)... |
| 11:53.41 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
| 11:53.42 | jerlique | Is there a development plan for asterisk which is published? |
| 12:00.20 | *** join/#asterisk arctic0 (~aalted@arctic0.active.supporter.pdpc) |
| 12:03.58 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 12:07.45 | *** join/#asterisk __Stu (~stuart@host226.lshift.net) |
| 12:08.11 | __Stu | Hi, I wonder if anyone here can help me |
| 12:08.25 | Mavvie | yay go luigi! |
| 12:08.44 | __Stu | I'm looking to setup an asterisk system using an ISDN2e connection and have been unable to find any info on what cards are supported for this |
| 12:12.01 | *** join/#asterisk cluecon (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 12:14.47 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
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| 12:16.33 | *** join/#asterisk outsidefactor (chrismarti@203-206-241-250.dyn.iinet.net.au) |
| 12:16.44 | sangee | Can i use JSP in AGI? |
| 12:17.35 | *** part/#asterisk outsidefactor (chrismarti@203-206-241-250.dyn.iinet.net.au) |
| 12:18.27 | *** join/#asterisk xcore (~rmaraschi@netuno.promitos.com.br) |
| 12:18.56 | *** join/#asterisk file[class] (~jcolp@66.199.241.90) |
| 12:19.04 | xcore | anybody here can help me |
| 12:19.13 | xcore | i wanna test the asterisk |
| 12:19.30 | file[class] | then test it |
| 12:19.41 | xcore | but i dont understand exactly how it work |
| 12:19.45 | xcore | thats the problem |
| 12:19.47 | xcore | =P |
| 12:19.56 | sangee | i want to connect database in remote side, i like to use JSP, is the anyway i can use JSP in Asterisk? |
| 12:20.05 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 12:20.07 | file[class] | xcore: then go learn |
| 12:20.15 | spyroux | xcore: there is a great handbook on digium.com I think |
| 12:20.26 | cluecon | ~docs |
| 12:20.28 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:20.30 | cluecon | ~cluecon |
| 12:20.31 | jbot | from memory, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 12:20.33 | xcore | spyroux yes, i have this handbook |
| 12:20.54 | cluecon | xcore: start with the wiki. read the docs. this will lead to understanding. |
| 12:21.28 | Delvar | and the light side understanding leads to it does |
| 12:22.07 | xcore | can i use the asterisk just like a "skype" |
| 12:22.25 | Delvar | yes |
| 12:23.43 | cluecon | Delvar: * is the dark side. The Dark Side will seduce you and control you it will. |
| 12:24.14 | Delvar | hehe, astrisk sure has a way of making a bad day for me |
| 12:25.34 | [TK]D-Fender | Peer-Peer telephony..... *shudder* |
| 12:27.17 | xcore | ok yoda |
| 12:28.29 | MikeJ[Laptop] | morning file[class] |
| 12:28.59 | file[class] | hey |
| 12:29.23 | MikeJ[Laptop] | wassup trouble.. |
| 12:30.24 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 12:31.25 | file[class] | helping people |
| 12:35.16 | jerlique | anyone here using SER or another such server for a commercial VOIP server |
| 12:37.37 | file[class] | jerlique: yes. |
| 12:38.20 | zoa | i am |
| 12:41.08 | *** join/#asterisk brettnem (~brettnem@user-0ccsrb6.cable.mindspring.com) |
| 12:41.09 | file[class] | I should log off |
| 12:42.20 | *** join/#asterisk user1fn (~joe@joe.dsl.1fn.net) |
| 12:42.49 | user1fn | hey guys |
| 12:43.19 | user1fn | anyone know if now is a really bad time to update my CVS? |
| 12:43.31 | user1fn | I'm now getting dropped calls and alarms on my pri |
| 12:43.46 | jerlique | fileclass: do you use this as a public server? How do you find it? |
| 12:45.24 | jerlique | Did you consider any other options for this? |
| 12:48.55 | jerlique | zoa where you answering my question? |
| 12:49.22 | *** join/#asterisk Mr3T (~Mr3Tee@ext116.almare.com) |
| 12:49.25 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:53.04 | zoa | yes i was |
| 12:53.04 | zoa | :) |
| 12:54.16 | jerlique | ahh cool |
| 12:54.30 | jerlique | so did you consider any other software for the job? |
| 12:55.22 | MikeJ[Laptop] | docelm0, can you turn off your away message junk... |
| 12:55.59 | MikeJ[Laptop] | jerlique, are you looking for other options becuase you don't like SER?? what are you getting at? |
| 12:56.09 | *** join/#asterisk juanjoc (~jcomellas@38-189-89-200.fibertel.com.ar) |
| 12:56.49 | jerlique | oh not at all... I'm just trying to get a idea of what is around, and what others are using... |
| 12:57.04 | jerlique | are you using voip in a commercial type voip service, something like vonage? |
| 12:59.27 | MikeJ[Laptop] | using, tes |
| 12:59.34 | MikeJ[Laptop] | yes |
| 13:00.11 | jerlique | it appears that SER has no active development on it, is this true? |
| 13:00.27 | zoa | yes i am |
| 13:00.38 | zoa | i dont know, but we do active SER development |
| 13:00.39 | zoa | :) |
| 13:00.49 | zoa | we use a combination of things |
| 13:00.57 | zoa | we use ser for what its best at |
| 13:00.59 | zoa | routing calls |
| 13:01.24 | *** join/#asterisk ArkyLady (ArkyLady@adsl-208-191-243-200.dsl.ltrkar.swbell.net) |
| 13:01.33 | Mr3T | ?? is there any documentation about video conferencing with asterisk ? |
| 13:01.47 | jerlique | and what other software do you use? |
| 13:02.18 | zoa | there is not |
| 13:02.22 | zoa | asterisk |
| 13:02.26 | zoa | and a lot of our own software |
| 13:03.07 | RoyK | Mr3T: voip-info.org is your best friend |
| 13:09.09 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 13:14.46 | Mr3T | RoyK : voip-info.org leads to N count of video projects but.... where is working concepts for sip/h.26x video phones to allow multiparty ( asterisk<->mcu?, iss binds ? any) |
| 13:15.33 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 13:16.53 | gres | hi all. I can't make 2 calls: asterisk->cisco ata 186 :(. Does anybody solve those problem? |
| 13:17.32 | gres | Two calls: cisco ata 186 - > asterisk - > pstn work well. |
| 13:17.42 | *** join/#asterisk Roey (~Roey@Roey.sustaining.supporter.pdpc) |
| 13:17.47 | Roey | EHLO |
| 13:17.48 | gres | Sorry for my English. |
| 13:18.20 | *** join/#asterisk brettnem (~brettnem@user-0ccsrb6.cable.mindspring.com) |
| 13:18.27 | Roey | I looked this up on Google but only found vague references to the * mailing list: are there case studies (with cost figures) of Asterisk in SMBs? |
| 13:20.55 | Nuxi | Why are some dialplan commands in /res and others not? |
| 13:22.20 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 13:23.01 | gres | I make it! I can make two calls: at186 <-> asterisk. |
| 13:23.03 | Roey | [TK]D-Fender: hi |
| 13:23.12 | Hmmhesays | gres: rtfm |
| 13:23.13 | [TK]D-Fender | Good Morning All! Anyone here been involved with testing Digium's new echo-cancelling T1 cards? |
| 13:23.18 | Roey | gres: :) |
| 13:23.22 | [TK]D-Fender | Hey Roey |
| 13:23.24 | Hmmhesays | you cannot make 2 calls on an ata186 if you are using g.729 |
| 13:23.29 | Roey | [TK]D-Fender: I looked this up on Google but only found vague references to the * mailing list: are there case studies (with cost figures) of Asterisk in SMBs? |
| 13:23.41 | Hmmhesays | super mario brothers? |
| 13:24.07 | Roey | small-to-midsized-businesses |
| 13:24.10 | Roey | and super mario brothers, yes. |
| 13:24.14 | Roey | :) |
| 13:24.21 | Roey | that's the first thing I think of when I see that acronym |
| 13:24.51 | tzafrir | Roey, you need a specific site: google |
| 13:25.06 | Roey | tzafrir, tikra ma she katavti |
| 13:25.11 | Roey | tzafrir: i did google |
| 13:25.11 | *** join/#asterisk SwK (~ken@12-219-156-206.client.mchsi.com) |
| 13:25.17 | Hmmhesays | probably most of us in here have production boxes in medium sized biznaz |
| 13:25.54 | Roey | Hmmhesays: ok, I'd like to find out how much it costs to implement an asterisk-based system for a given set of conditions (# of lines, # of offices, etc.) |
| 13:26.27 | Roey | Hmmhesays: our proprietary Nitsuko PBX+phone system cost us $30k back in 2000. |
| 13:26.36 | Roey | and this is for a small business mind you |
| 13:26.41 | Hmmhesays | it's not real hard to figure out <price of hardware> + <your pay> |
| 13:26.49 | cluecon | Roey: fire away with the specs and you will likely get an answer. We are about to implement here for a small call center environment and it will run us about $20,000 |
| 13:27.11 | Roey | *yes* and I'm vanting to know de cost of de said equipment |
| 13:27.20 | Hmmhesays | that's where google comes in |
| 13:27.27 | Roey | Hmmhesays: ARG!!~!!! :) |
| 13:27.31 | Roey | Hmmhesays: I DID google! |
| 13:27.38 | Hmmhesays | for hardware prices? |
| 13:27.42 | Roey | yes |
| 13:27.49 | Roey | the computers, what kind of cards you need, etc. |
| 13:28.03 | cluecon | Roey: how many lines (#pri) how many phones, etc? |
| 13:28.05 | Roey | cluecon: here are our specs (I just want to compare -- not planning on swtiching or anything) |
| 13:28.09 | Roey | cluecon: 6 POTS lines |
| 13:28.17 | [TK]D-Fender | Roey : Since * is a "roll-your-own" kind of solution is hasn't gained the kind of corporate appeal that invites those comparisons yet. Most businesses that sell it are soo specialized that they promote it on their own terms. As for cost it always depends what you want to do. |
| 13:28.22 | Roey | cluecon: about 20 or so connections to phones |
| 13:28.26 | Roey | cluecon: one office for now |
| 13:28.40 | Hmmhesays | 4k |
| 13:28.44 | Roey | cluecon: need for voicemail, too |
| 13:29.10 | Roey | Hmmhesays: can you break down the cost into the price of the parts? |
| 13:29.14 | Roey | Hmmhesays: thanks btw |
| 13:29.33 | [TK]D-Fender | Roey : I am looking to implement it at my company as part of our move to a new office. I'm dealing with 30 phones and the solutions is coming up to 15k$ instead of 27k$ that a Nortel BCM solution ran for. |
| 13:29.42 | Roey | ah |
| 13:29.52 | Roey | hmm, I thought it'd be much less expensive but OK |
| 13:29.59 | Hmmhesays | 4k is as cheap as you'd get for a decent setup |
| 13:30.02 | Hmmhesays | imho |
| 13:30.03 | Roey | I don't know what kind of specialized parts you need |
| 13:30.12 | Roey | Hmmhesays: yeah, 4k sounds reasonable and all (to me at least) |
| 13:30.13 | Hmmhesays | heh, a couple digium cards |
| 13:30.25 | Roey | ok, what are those? what can I google for? |
| 13:30.25 | [TK]D-Fender | For the basics I can recommend * highly, though certain things do fall through the cracks. There is some work to be done on the queue system to give it more corporate appeal. |
| 13:30.45 | Hmmhesays | 2200 for the pc, 600 for the fxo ports, 100 a piece for the phones |
| 13:30.58 | Hmmhesays | *ahem* and call parking |
| 13:31.28 | [TK]D-Fender | Hmmhesays : Talking to me about call parking? |
| 13:31.39 | Roey | [TK]D-Fender: oh, * has a queue system? nice. I wonder if it can handle things like collecting user input (credit card#, or trouble ticket#, say) and passing it to another server to validated or ultimately sent to the tech support person so that they don't have to ask you twice for your info |
| 13:31.46 | Roey | 200 for the PC, ok. |
| 13:31.49 | Hmmhesays | [Tk] yeah.... in reference to corporate appeal |
| 13:31.59 | cluecon | Roey: thats about right. The phones may run you about 200 depending on the features you want out of your phones. |
| 13:32.06 | Roey | Hmmhesays: the fxo ports.. they take in POTS? |
| 13:32.17 | Hmmhesays | by definition yes |
| 13:32.19 | cluecon | Roey: yes, you can collect user input and pass it off to an agi that does the verification. |
| 13:32.44 | Roey | cluecon: have you ever called those slick tech centers where they don't have to ask you twice for your info? |
| 13:33.09 | cluecon | Roey: I'm building a system that asks for the info so that our agents don't have to ask for it. |
| 13:33.13 | Hmmhesays | Roey: there's about eleventy billion ways you could do that |
| 13:33.22 | Roey | cluecon: also, have you seen the directory applications where you input the first coupla letters of the sirname of the person you're trying to reach, and you get the number in return? |
| 13:33.31 | Roey | cluecon: ah |
| 13:33.33 | cluecon | Roey: doing that as well. |
| 13:33.46 | Roey | cluecon: so there is more customized software needed outside of *, isn't there (to do that) |
| 13:33.56 | Hmmhesays | just some programming skills |
| 13:34.07 | Roey | heh |
| 13:34.13 | cluecon | Roey: just roll your own using agi (numerous languages, i use perl) |
| 13:34.21 | cluecon | most of it is very simple. |
| 13:34.23 | Hmmhesays | I just wrote a set of callback agi's for a guy out in st maarten |
| 13:34.44 | Roey | maybe contractors can do this stuff for companies and still come in at $25k where the proprietary comes in at $35k |
| 13:34.46 | Hmmhesays | like 600 lines of php |
| 13:34.49 | Roey | hmm |
| 13:34.51 | Roey | ok |
| 13:34.56 | Hmmhesays | was nothing |
| 13:35.00 | Roey | Hmmhesays: is there a directory for these specialized applications? |
| 13:35.06 | Roey | (to me it sounds exotic and specialized) |
| 13:35.10 | Hmmhesays | AGI |
| 13:35.13 | Roey | AGI? |
| 13:35.19 | Hmmhesays | www.voip-info.org |
| 13:35.20 | Hmmhesays | go there |
| 13:35.21 | Roey | ok |
| 13:35.23 | Roey | ok thanks |
| 13:35.27 | Hmmhesays | np |
| 13:35.34 | cluecon | Roey: I could drop in a 30 phone system based on * for $25,000 and be just fine on the profit side. |
| 13:35.54 | Hmmhesays | haha, yeah by 'fine' you mean about 15k in your pocket |
| 13:36.09 | wasim | more like 20k at current voip handset pricing |
| 13:36.09 | cluecon | Hmm: just don't tell anyone that and I won't have to kill you. |
| 13:36.27 | [TK]D-Fender | Roey : what you are asking is for isn't call queueing but just basical dial plan stuff. Very easy. |
| 13:36.31 | wasim | hell, more like 23k at current voip pricing ;) |
| 13:36.38 | Roey | so, to sum up, the kinds of cool things I've seen include 1) automated telephone directory service; 2) transferring user input to the tech support rep; 3) validating user input |
| 13:36.41 | Hmmhesays | I make about 1k on a small business install, for 5 lines and 8 handsets |
| 13:36.41 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:36.42 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:37.01 | Roey | Hmmhesays: ok, you are such an integrator then? |
| 13:37.15 | Hmmhesays | such a what what? |
| 13:37.25 | Roey | intemegrator |
| 13:37.39 | tzanger | Hmmhesays: :-) |
| 13:37.45 | cluecon | wasim: my systems are based on using pri incoming and IP500 handsets. Not cheap by any means. |
| 13:37.50 | *** join/#asterisk spyroux (~arnaud@cl-16.bru-01.be.sixxs.net) |
| 13:37.58 | Hmmhesays | haha i know what the word is, i'm not sure exactly what you mean in that context though |
| 13:38.03 | [TK]D-Fender | Roey : #1 = Easy, #2 a little tricky depending on how many agents and what means of sending the call to them you intend to do. #3 = Not too hard in AGI |
| 13:38.14 | Hmmhesays | tzanger |
| 13:38.17 | cluecon | Roey: You will find several of us that setup and install * for businesses. |
| 13:38.25 | tzanger | integrator is right |
| 13:38.29 | tzanger | you're a systems integrator |
| 13:38.32 | [TK]D-Fender | cluecon : I'm planning for IP 600's ;) $$$ |
| 13:38.35 | tzanger | you get all the pieces and integrate them into something useful for someone |
| 13:38.42 | Hmmhesays | haha, to early in the morning for big words |
| 13:39.00 | cluecon | TKD: I would love to have IP600s here, but I'm pushing it to get IP500s. |
| 13:39.14 | Hmmhesays | I understand now |
| 13:39.32 | Hmmhesays | I still have a snom200 sitting on my desk |
| 13:39.39 | Hmmhesays | and I like it |
| 13:39.50 | wasim | i'm getting ywh200d |
| 13:39.54 | bjohnson | any suggestions for Kitchener and London DIDs? |
| 13:39.59 | cluecon | I use BT100s for testing. Not bad, but not great either. |
| 13:40.16 | Hmmhesays | heh, what do you really need for testing though |
| 13:40.25 | Roey | [TK]D-Fender: you wouldn't recommend settng up * for your own small business if you're just starting would you? |
| 13:40.29 | wasim | anybody tried the atcom sets? |
| 13:40.29 | durufle | bj: maybe unlimitel? |
| 13:40.31 | cluecon | Hmm: Not much. |
| 13:40.41 | Hmmhesays | Roey: why wouldn't you? |
| 13:40.46 | Zeeek | wasim I have three |
| 13:40.56 | cluecon | Roey: I would. It is perfect for cost control and management. |
| 13:40.57 | Roey | Hmmhesays: it's not so complicated? |
| 13:41.09 | Hmmhesays | basic asterisk setups are not complicated |
| 13:41.10 | Roey | cluecon: hmm, I'm still wrapping my brain around SIP |
| 13:41.14 | cluecon | Roey: takes 5 minutes to have * working. |
| 13:41.25 | Zeeek | wasim in fact I'm playing with the digitmapping of one right now |
| 13:41.25 | wasim | Zeeek: do you have yuxin sets as well? |
| 13:41.25 | Roey | heh |
| 13:41.29 | Hmmhesays | to set up asterisk you don't need to know a dern thing about sip |
| 13:41.51 | Hmmhesays | in fact you really have to have zero voip knowledge |
| 13:41.52 | Roey | we haven't even set up the call routing feature of our pbx |
| 13:41.54 | wasim | Zeeek: whats the plastic body like? |
| 13:42.03 | Zeeek | wasim these were from "netweb" so I don't know the origins, just that they are that type of PA168s |
| 13:42.04 | [TK]D-Fender | Roey : recommending is a variable idea. Depends how comfotable you and your superiors are with it and if you have a company that can support you in case of failure |
| 13:42.09 | Roey | Hmmhesays: well if you're using voip then you do need to know sip yes? |
| 13:42.22 | Hmmhesays | you could think magic gnomes are carrying the voice from point A to point B |
| 13:42.25 | Zeeek | wasim the bodies or pretty good, no sticky buttons or anything |
| 13:42.27 | bjohnson | durufle: any idea on pricing? they con't have anything on their site. PLUS, looks like they're SIP only |
| 13:42.32 | Roey | [TK]D-Fender: ok, see, what do you do in case of failure.. do you keep one or two POTS lines outside the PBX, just in case? |
| 13:42.52 | Roey | [TK]D-Fender: well we're comfortable with our current setup (especially since it costs so much and we got it only in 2000) |
| 13:42.55 | [TK]D-Fender | Not just that, but to help set things up or repair things if you leave |
| 13:43.02 | Zeeek | wasim the white atcom 360 (?) is a little cheap looking and feeling but works ok |
| 13:43.10 | wasim | Zeeek: i'm thinking 323 |
| 13:43.11 | Roey | [TK]D-Fender: ok, so you DO leave a phone line unconnected to the pbx |
| 13:43.20 | Hmmhesays | what you do in case of failure is have a clone box setup, if you are that paranoid |
| 13:43.25 | Roey | right |
| 13:43.31 | Zeeek | wasim let me look, just a sec |
| 13:43.40 | Roey | Hmmhesays: and someone here did mention that they're working on a heartbeat failover solution |
| 13:43.46 | Roey | s/solution/setup |
| 13:43.56 | Hmmhesays | that way if you have something catastrophic happen that you can't seem to fix...... |
| 13:44.01 | Hmmhesays | Roey, voip-info.org |
| 13:44.05 | Roey | ah, ok |
| 13:44.11 | Hmmhesays | you will see redudant solutions there |
| 13:44.15 | Hmmhesays | *redundant |
| 13:44.19 | Roey | :) |
| 13:44.23 | Roey | thanks |
| 13:44.24 | wasim | Roey: junghanns even has failover PRI switches |
| 13:44.29 | cluecon | Hmm: just setup a vonexus server as the backup :) |
| 13:44.30 | Zeeek | wasim I think I have two white 323 |
| 13:44.49 | Hmmhesays | the new chan_OH323 is pretty good |
| 13:44.59 | Roey | Hmmhesays: that has a directory of integrated solution providers as well? |
| 13:45.30 | Hmmhesays | go there and look |
| 13:45.34 | Roey | ok |
| 13:45.35 | Roey | brb |
| 13:46.05 | gres | Hmmhesays: thanks! Y'll be use g711. :) |
| 13:46.19 | [TK]D-Fender | Roey : Allways keep a basic line available for fall-back, but I meant a service company to support your company if you leave. |
| 13:47.13 | Hmmhesays | gres: np |
| 13:47.32 | Hmmhesays | the ata 188 I believe can handle 2 g.729 calls |
| 13:47.35 | bjohnson | anyone using thinktel.ca? |
| 13:47.39 | Meaty | In a callerID Number, are there a syntax for add an extension number ? |
| 13:47.55 | *** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146) |
| 13:48.24 | *** join/#asterisk justin_time (~justintim@ns-inet.usa-companies.net) |
| 13:48.26 | cluecon | is there a way to have a voicemail recorded to multiple mailboxes at the same time? |
| 13:48.36 | gres | Hmmhesays: Mmm... I'd like to belive. |
| 13:48.37 | tzanger | cluecon: I don't think so, what are you thinking |
| 13:49.35 | Meaty | cluecon: Try to use & between multiple mailboxs |
| 13:50.24 | squirrelv5 | hello, how can i test my installation of *?aside from connecting inside my lan, i want sombody from the net to call me |
| 13:50.33 | cluecon | I have a technical services group of phones. The people at these (me and one other guy) may not answer our phones when they ring as a group. I want to have a general mailbox for technical services that actually records the message into both of our mailboxes. |
| 13:50.33 | zoa | cluecon: i dont think so |
| 13:50.52 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:51.02 | Hmmhesays | hrm, I just realized we have 10 cisco 7940's on the shelf |
| 13:51.18 | RoyK | ~blah |
| 13:51.19 | jbot | somebody said blah was Y |
| 13:51.32 | squirrelv5 | hello, how can i test my installation of *?aside from connecting inside my lan, i want sombody from the net to call me |
| 13:51.54 | Meaty | <PROTECTED> |
| 13:52.00 | *** join/#asterisk arctic0 (arctic0@arctic0.active.supporter.pdpc) |
| 13:52.03 | Meaty | cluecon ^^ |
| 13:52.06 | Hmmhesays | ohhh a 7960 |
| 13:52.17 | spyroux | Hmmhesays: if they take too many space, i can help you :-D |
| 13:52.24 | cluecon | Is the new syntax for Set documented anywhere? |
| 13:52.34 | cluecon | Hmm: I would be happy to take some of those off your hands. |
| 13:52.37 | Hmmhesays | spyroux: got power supplies? |
| 13:52.41 | cluecon | Meaty: wiki? |
| 13:52.46 | *** part/#asterisk Maksim (~max@213.142.207.20) |
| 13:52.54 | spyroux | Hmmhesays: no :-/ |
| 13:53.04 | Hmmhesays | heh |
| 13:53.05 | _BB_ | PSUs are cheap |
| 13:53.10 | _BB_ | $15 a pop |
| 13:53.22 | Hmmhesays | hrm, sip or skinny for these? |
| 13:53.31 | cluecon | Meaty: thx |
| 13:53.36 | squirrelv5 | whats is skinny? |
| 13:53.39 | Hmmhesays | been awhile since i've even touched a cisco phone |
| 13:53.48 | _BB_ | aah but they are very very nice |
| 13:54.02 | _BB_ | got a 7960 at home |
| 13:54.08 | spyroux | http://www.cisco.com/en/US/products/hw/phones/ps379/ps1855/prod_view_selector.html <<< hum ... |
| 13:54.08 | Hmmhesays | what protocol are you using? |
| 13:54.11 | spyroux | :-D |
| 13:54.23 | _BB_ | sip |
| 13:54.45 | Hmmhesays | hrm, I gotta rustle up a power supply for this bad boy |
| 13:55.18 | Roey | Hmmhesays: http://www.digium.com/index.php?menu=fxsvfxo <--- that's very, very understandable |
| 13:55.37 | Hmmhesays | haha Roey: sure is |
| 13:55.51 | Hmmhesays | reminds me of that mentally retarded guy I saw riding the motor scooter down the road yesterday |
| 13:56.03 | Roey | ?! |
| 13:56.09 | Roey | LOL |
| 13:56.20 | [TK]D-Fender | Roey : No, get a T-1 card an a channel bank |
| 13:56.25 | Hmmhesays | Roey: only if you want to pull all your hair out and give yourself an ulcer |
| 13:56.38 | jerlique | speaking of which what channel banks are ppl using |
| 13:56.43 | DeeJayTwo | Roey: A quad t1 should do... |
| 13:56.45 | Hmmhesays | adtran |
| 13:56.49 | DeeJayTwo | 2 ports for your clients (48 lines) |
| 13:56.49 | cluecon | Roey: no, use SIP Phones on the inside (LAN based RJ45) and save yourself some headaches. |
| 13:56.51 | Hmmhesays | Roey: why do you need fxs? |
| 13:56.54 | Hmmhesays | you don't |
| 13:56.56 | DeeJayTwo | and 1 port for a PRI (up to 23 lines) |
| 13:57.00 | [TK]D-Fender | A channel bank can (typically) have up to 24 channels. |
| 13:57.14 | Roey | [TK]D-Fender: we have this switching thingy next to our pbx.. it lets us route ports (on the switching thingy) to office rooms |
| 13:57.17 | Hmmhesays | hrm, can someone look at their 7960 psu and give me the specs? |
| 13:57.34 | [TK]D-Fender | Though for * channel banked phones = *ick* Not unless you really have to. |
| 13:57.34 | Roey | Hmmhesays: that page shows that you use FXS on the PBX to connect to the phones |
| 13:57.39 | *** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
| 13:57.47 | arctic0 | Hi, i've just installed asterisk and sjphone, I have made changes to the (sip.conf, extension.conf) asterisk files. And I have configures sjphone to deal with asterisk. Now i am calling #1000, but sjphone tell me "waiting for proxy registration" all the time. What i'm doing wrong? Any idea? |
| 13:57.52 | Roey | cluecon: oh, right right |
| 13:57.56 | [TK]D-Fender | Roey : just a wiring spaghetti organizer? |
| 13:57.57 | _BB_ | 7960 PSU is 48vDC |
| 13:58.03 | Hmmhesays | artic0, probably a typo |
| 13:58.11 | Roey | cluecon: so you can hook up that server to a standard switch |
| 13:58.14 | Hmmhesays | 48v at what amperage? |
| 13:58.15 | arctic0 | typo? what's that? |
| 13:58.25 | _BB_ | http://www.voipsupply.com/product_info.php?products_id=191 |
| 13:58.26 | Hmmhesays | in your sip.conf |
| 13:58.26 | spyroux | Hmmhesays: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html << PSU reference |
| 13:58.33 | Hmmhesays | ooh spyroux, thanks |
| 13:58.34 | _BB_ | phones typically take up to about 150mA |
| 13:58.36 | Roey | [TK]D-Fender: we have this thing that looks like spaghetti wires wound around this organizer "breadboard" (it looks like a breadboard really) |
| 13:58.49 | cluecon | Roey: you would need a network connection from the * box to hook into your network. Once that is done, you hook your phones to the network and away you go. |
| 13:58.58 | Roey | cluecon: ok then.. that's it. |
| 13:59.09 | Roey | cluecon: SIP phones are standard, yes? |
| 13:59.19 | Hmmhesays | Roey: you are a funny guy |
| 13:59.33 | cluecon | SIP Phones are pretty common. |
| 13:59.36 | Roey | cluecon: like.. what if they have specialized buttons (for selecting a particular line, or to page everyone on the PA, for example) |
| 13:59.44 | Roey | cluecon: I've seen them at different places |
| 13:59.49 | Hmmhesays | Roey: so you have stumbled upon the downfall of asterisk |
| 13:59.55 | Roey | Hmmhesays: how so?! |
| 13:59.59 | Roey | iax? |
| 14:00.00 | Hmmhesays | getting the little buttons on the phones to do what you want |
| 14:00.03 | Roey | oh |
| 14:00.08 | Roey | Hmmhesays: heh |
| 14:00.11 | cluecon | Roey: that requires some work. |
| 14:00.21 | _BB_ | cluecon: they have an iax ATA...it's a halfway step |
| 14:00.23 | Roey | Hmmhesays: I had assumed that there is a standard protocol for this |
| 14:00.46 | Hmmhesays | Roey: yes and no |
| 14:00.47 | cluecon | _BB_: don't want an ATA, want a phone. (bangs club on table) |
| 14:01.16 | cluecon | Roey: the internal communication protocol of * is IAX. But there aren't any IAX Phones so most everyone uses SIP. |
| 14:01.35 | Roey | IAX and SIP are complements or substitutes? |
| 14:01.37 | _BB_ | there are IAX softphones though...no reason why there can't be a hardphone |
| 14:01.59 | _BB_ | but it's probably better to support one protocol rather than two for the end client |
| 14:02.08 | Delvar | exept FAR less hardware supports IAX :) |
| 14:02.11 | *** join/#asterisk Jeff_BR (Jeff_Brazi@201.30.193.134) |
| 14:02.18 | Hmmhesays | i'd say sip and iax are compliments right now |
| 14:02.19 | cluecon | _BB_: i use iaxComm on my computers. use it my laptop when I'm away from the office. |
| 14:02.41 | Delvar | Sipronix apparently do an aix2 hardware phone |
| 14:02.47 | Roey | *complements |
| 14:02.48 | Roey | ah |
| 14:02.49 | Jeff_BR | i´m with problems on loading wct11xp on FC3 ! any ideas ? Zaptel compiles OK. |
| 14:02.50 | Roey | Hmmhesays: ok |
| 14:02.53 | cluecon | SIP and IAX are just protocols. * supports both. |
| 14:03.11 | cluecon | Jeff: what kind of problems? |
| 14:03.30 | Roey | cluecon: all I know is that SIP tells you the IP address of the phone that you're trying to reach with a phone number (or something like a DNS but for VOIP) |
| 14:03.43 | Roey | and that you can put in a service record in your DNS to point to your SIP server |
| 14:04.05 | Hmmhesays | Roey |
| 14:04.08 | Roey | ya |
| 14:04.10 | Roey | YES I WILL READ |
| 14:04.11 | Roey | ok |
| 14:04.11 | Roey | :) |
| 14:04.14 | Jeff_BR | Well , the module doen´t load : wcte11xp |
| 14:04.16 | Hmmhesays | let me give you a pointer here |
| 14:04.19 | Roey | yeah. |
| 14:04.22 | Roey | voip-blah-blah ok |
| 14:04.30 | Roey | Hmmhesays: I'm reading it now |
| 14:04.31 | cluecon | Jeff: which module are you trying to load? |
| 14:04.33 | Hmmhesays | http://www.protocols.com/pbook/VoIPFamily.htm#SIP |
| 14:04.39 | Jeff_BR | wcte11xp |
| 14:04.39 | Hmmhesays | read that |
| 14:04.44 | Jeff_BR | is for E1 |
| 14:04.51 | Jeff_BR | i have a TE110P |
| 14:05.07 | Roey | Hmmhesays: thanks |
| 14:05.13 | cluecon | Jeff: isn't the module for that wct1xxp? |
| 14:05.15 | Jeff_BR | It works find on 2.4 .. but i´m getting problems here with 2.6 |
| 14:05.18 | Hmmhesays | Roey: np, it's some good general info about voip protocols |
| 14:05.49 | cluecon | Jeff: whats the error? |
| 14:06.02 | Jeff_BR | zaptel 190724 1 wcte11xp |
| 14:06.15 | Jeff_BR | well , now i dont receive an error |
| 14:06.23 | Jeff_BR | but ztcfg reports that .. |
| 14:06.38 | Jeff_BR | line 5: Unable to open master device '/dev/zap/ctl' |
| 14:06.50 | Jeff_BR | and there´s something strange .. |
| 14:07.11 | Jeff_BR | <PROTECTED> |
| 14:07.12 | Jeff_BR | <PROTECTED> |
| 14:07.12 | Jeff_BR | <PROTECTED> |
| 14:07.12 | Jeff_BR | <PROTECTED> |
| 14:07.24 | RoyK | Jeff_BR: documented bug....... |
| 14:07.31 | RoyK | rmmod t1xxp |
| 14:07.33 | Jeff_BR | i suppose .. that .. wcte11xp is for being there. not t1xxp |
| 14:07.37 | RoyK | insmod it again |
| 14:07.39 | RoyK | ztcfg -v |
| 14:07.58 | RoyK | if that doesn't work, power down the server, unplug power, wait a few minutes, power up and try again |
| 14:08.00 | Jeff_BR | [root@voip modules]# rmmod t1xxp |
| 14:08.00 | Jeff_BR | ERROR: Module t1xxp does not exist in /proc/modules |
| 14:08.09 | RoyK | huh? |
| 14:08.10 | RoyK | lsmod |
| 14:08.13 | RoyK | pastebin it |
| 14:08.15 | RoyK | please |
| 14:08.19 | RoyK | ~pastebin |
| 14:08.20 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 14:08.23 | Jeff_BR | Module Size Used by |
| 14:08.50 | [TK]D-Fender | failure |
| 14:09.11 | *** join/#asterisk Jeff_BR (Jeff_Brazi@201.30.193.134) |
| 14:09.14 | *** join/#asterisk GMsoft (~gmsoft@gmsoft.developer.gentoo) |
| 14:09.16 | GMsoft | hi |
| 14:09.18 | Jeff_BR | :( |
| 14:09.22 | Jeff_BR | forgot .. |
| 14:09.24 | Hmmhesays | flood |
| 14:09.30 | Hmmhesays | punishable by death |
| 14:09.32 | Jeff_BR | sorry. |
| 14:09.36 | Jeff_BR | :( |
| 14:09.54 | Jeff_BR | i don´t use IRC .. |
| 14:12.04 | Hmmhesays | Jeff_BR: that's funny |
| 14:12.05 | bkw_ | thats right.. use pastebin.ca to put large ammounts of stuff in here |
| 14:12.14 | bkw_ | your cards aren't getting interrupts |
| 14:12.21 | *** join/#asterisk brettnem (~brettnem@user-0ccsrb6.cable.mindspring.com) |
| 14:12.22 | Hmmhesays | your have the l33t telepathy skills cause I swear you are typing in here |
| 14:13.15 | *** join/#asterisk _omer (~dfsdf@203.215.180.250) |
| 14:13.19 | _omer | hi, |
| 14:13.49 | _omer | which card is for single E1 line? |
| 14:14.05 | *** join/#asterisk asdfblah (~UNIX@pcp04541662pcs.brmngh01.mi.comcast.net) |
| 14:14.12 | Jeff_BR | yes they are |
| 14:14.17 | Jeff_BR | interrupts 5 and 10 |
| 14:14.21 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:14.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:14.27 | Hmmhesays | so, should I bother trying to use skinny on this 7960? |
| 14:14.33 | Hmmhesays | or 'find' the sip ios |
| 14:14.46 | Jeff_BR | both |
| 14:15.04 | Meaty | In a callerID Number, are there a syntax to add an extension number ? |
| 14:15.11 | asdfblah | quick question, if i want to be able to dial into my asterisk box, get a tone, then dial out, what steps do i need to take/ |
| 14:15.39 | _BB_ | skinny works ok |
| 14:15.40 | Jeff_BR | I´ll try to shutdown ... |
| 14:15.57 | _BB_ | sip is a little better with asterisk, but you need the image |
| 14:16.02 | Hmmhesays | asdfblah: identify your origination and termination points |
| 14:16.11 | Hmmhesays | _BB_ that's why i said 'find' |
| 14:16.54 | asdfblah | call from cell, to my broadvoice numer, then on the asterisk box i want to dial out through a POTS line |
| 14:17.06 | _BB_ | it's not publicly avilable without a cisco contract though |
| 14:17.06 | zoa | <Hmmhesays> so, should I bother trying to use skinny on this 7960? ===> look at one of the tutorials on asteriskguru.con |
| 14:17.07 | asdfblah | numbers* |
| 14:17.08 | zoa | .com |
| 14:17.11 | zoa | it says how to use it |
| 14:17.37 | Hmmhesays | BB, no it's not... of course by 'find' i meant contact cisco and purchase |
| 14:17.49 | zoa | Jeff_BR: you also look at asteriskguru.com i wrote something there about cards not taking interrupts or missing interrupts |
| 14:17.54 | *** join/#asterisk file[class] (~jcolp@66.199.241.90) |
| 14:18.00 | file[class] | oh no, I'm alive |
| 14:19.16 | cluecon | ~frag file[class] |
| 14:19.18 | jbot | ACTION readies the nuke launcher and fires some rounds at file[class] |
| 14:19.25 | cluecon | damn, missed. |
| 14:20.14 | Hmmhesays | is there a deadline to register for cluecon? |
| 14:20.36 | file[class] | Hmmhesays: yesterday! oh no!!! |
| 14:21.05 | Hmmhesays | oh no!! lol |
| 14:21.17 | file[class] | I was lying |
| 14:21.27 | *** join/#asterisk jarrod (jarrod@t06700.humbled.org) |
| 14:21.29 | *** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net) |
| 14:22.18 | jarrod | hey is there a way to start recording a convo mid-conversation? |
| 14:22.22 | file[class] | cluecon: login on the site and take a gander |
| 14:22.30 | jarrod | with some sort of background extension |
| 14:22.49 | Hmmhesays | redirect to a monitered extension with the manager |
| 14:22.50 | *** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se) |
| 14:23.01 | Hmmhesays | i got a month to try and talk my employer into sending me |
| 14:23.28 | _BB_ | doesnt '*1' do it with the w/W Dial flags? |
| 14:23.47 | cluecon | login at cluecon.com? tried, can't. |
| 14:25.06 | _BB_ | W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk > v1.0.x) |
| 14:25.22 | puppet | _bb_: it works |
| 14:25.31 | puppet | _bb_: you need to activate it tho |
| 14:27.09 | asdfblah | Hmmhesays any ideas? |
| 14:28.51 | *** part/#asterisk malabar (~malabar@164.80-202-124.nextgentel.com) |
| 14:29.14 | Hmmhesays | asdblah: send your incoming call from broadvoice to an extension that takes your fxo line off hook |
| 14:29.51 | asdfblah | got a good website where i can read how to do that sort fo thing? |
| 14:30.09 | Hmmhesays | um... dialplan in the wiki? |
| 14:32.55 | file[class] | cluecon: lemme try to remember the admin pass |
| 14:33.15 | *** join/#asterisk powD (~atob100@83.146.53.34) |
| 14:33.49 | file[class] | meh I don't remember it |
| 14:34.33 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:35.21 | cluecon | file: if i could get brians attention...but I don't want to interrupt tonys conversation. |
| 14:36.19 | [TK]D-Fender | Do *'s logs ever create "historical" copies to archive older messages or do they just grow continuously? |
| 14:36.22 | file[class] | I can log in when I get home |
| 14:37.11 | newl | tkd: they'll grow forever until either a) you handle it manually, or b) use a wonderful tool like logrotate, or c) write something to do it for you. |
| 14:38.00 | Hmmhesays | call a script with cron to archive them nightly/weekly/monthly |
| 14:38.02 | Hmmhesays | whatever |
| 14:38.08 | [TK]D-Fender | newl : Ok, that works. Just need to know the rules :) |
| 14:38.26 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:38.52 | newl | or d) you run out of space hehe |
| 14:38.55 | Hmmhesays | set the logger.conf to log everything, and then watch your disk space magically disappear |
| 14:39.02 | Ariel_ | hello everyone |
| 14:39.16 | Hmmhesays | hey Ariel_ |
| 14:39.19 | *** join/#asterisk santiago (~santiago@208.195.214.92) |
| 14:39.25 | newl | Hmmhesays: easy fix..just whack in another drive or two and grow the volume 8) |
| 14:39.59 | Hmmhesays | <chuckle>@newl |
| 14:39.59 | asdfblah | can anyone recommend a good IP phone for use in the office |
| 14:40.15 | Hmmhesays | asdfblah: what are your ip phone requirements |
| 14:40.58 | Ariel_ | asdfblah, I love the Polycom IP-500/501 or IP-600 great phones. |
| 14:41.08 | Hmmhesays | I tried two tin cans, a piece of string tapping out binary.... but no one could understand me |
| 14:41.30 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
| 14:41.48 | Ahrimanes | for file in `ls -1 /var/log/asterisk`; do rm /var/log/asterisk/$file; ln -s /dev/null /var/log/asterisk/$file; done |
| 14:41.55 | Ahrimanes | will fix it with no added disks.. |
| 14:41.57 | Hmmhesays | speaking of.. I need to get my hands on an IP500 |
| 14:42.57 | tzafrir | Ahrimanes, drop the -l ! |
| 14:43.07 | [TK]D-Fender | Ahrimanes : Curious as to the origin of your nick. That name has some meaning for me going way back.... |
| 14:43.07 | Ahrimanes | tzafrir: it's a 1 not an l |
| 14:43.26 | Ahrimanes | [TK]D-Fender: ever done roleplaying? (not sexually!) |
| 14:43.34 | Hmmhesays | Ariel_ got a recommendation as to where to purchase one? |
| 14:43.34 | [TK]D-Fender | V:TM, thought so :) |
| 14:43.38 | Ahrimanes | [TK]D-Fender: :D |
| 14:43.44 | tzafrir | Ahrimanes, ls |cat |
| 14:43.46 | Hmmhesays | ahaha |
| 14:43.59 | [TK]D-Fender | Very select audience...... |
| 14:44.08 | tzafrir | ls gives a sane output when the output doesn't go to a terminal |
| 14:44.12 | Ariel_ | Hmmhesays, I have gotten mine from atacom and voipsupply |
| 14:44.26 | Ahrimanes | tzafrir: just being cautious |
| 14:44.47 | Hmmhesays | yeah i'll just get it from atacomm then |
| 14:44.51 | Ahrimanes | [TK]D-Fender: hehe would seem so, been ages since i've played last |
| 14:45.02 | [TK]D-Fender | I use the True Name of a certain Anti-diluvian as a P/W here in some places :) |
| 14:45.18 | *** join/#asterisk user1fn (~joe@joe.dsl.1fn.net) |
| 14:45.34 | Ahrimanes | [TK]D-Fender: hehe |
| 14:45.40 | tzafrir | Ahrimanes, not careful in the right way. 4 files in a raw won't harm you, as long as they are separated by whitespaces. a summary of the block numbers will harm you |
| 14:45.53 | user1fn | looking for a pri guru... anyone willing to take a shot? |
| 14:46.10 | Ahrimanes | tzafrir: as i said, it's a 1 (one), not an l (letter l) |
| 14:46.26 | tzafrir | Ahrimanes, for file in `echo /var/log/asterisk/* |grep -v '*'` |
| 14:46.35 | *** join/#asterisk rristroph2 (~rristroph@mail.airlinksystems.com) |
| 14:47.15 | Ahrimanes | tzafrir: yes i could also find about 500 other ways of doing it.. but the one i pasted is generally the one i use |
| 14:47.59 | tzafrir | Ahrimanes, and since you mention find, `find /var/log/asterisk -type f` :-p |
| 14:48.05 | Ahrimanes | tzafrir: haha |
| 14:49.42 | rristroph2 | I am picking out a softphone for use with asterisk in a small office (8 phones). I should work with windows, and I would like it to be open source or have an SDK so that later I can integrate the softphone client with other software. I am considering phonegaim, sipXphone, ans sipXezPhone. Any suggestions ? |
| 14:50.32 | zoa | idefisk |
| 14:50.33 | file[class] | awww I'm away, how nice |
| 14:50.35 | zoa | but its not open source |
| 14:50.42 | zoa | but we would customize if for you needs |
| 14:51.45 | file[class] | zoa zoa zoa |
| 14:52.36 | zoa | yes yes yes ? |
| 14:52.53 | file[class] | hi. |
| 14:53.00 | user1fn | anyone familiar with why HDLC errors would occur on a PRI? |
| 14:53.09 | user1fn | seeing this with recent CVS perhaps? |
| 14:54.02 | rristroph2 | zoa: I will check out idefisk, thanks for the suggestion |
| 14:55.51 | zoa | website just went down, some backup script is causing a load of 300 i think |
| 14:56.02 | bkw_ | file |
| 14:56.04 | bkw_ | oh file |
| 14:57.27 | Ahrimanes | zoa: nice |
| 14:57.34 | cluecon | bkw_: can you check my cluecon registration? |
| 14:58.41 | bkw_ | clue yes |
| 14:58.58 | bkw_ | cluecon, name? |
| 14:59.01 | *** join/#asterisk tskjoedt (~tskjoedt@130.226.169.177) |
| 14:59.03 | file[class] | bkw_: I want my Powerbook :( |
| 14:59.09 | cluecon | registered on friday, but didn't get the confirm...its under Jonathan Dale. |
| 14:59.16 | Ahrimanes | file[class]: i want your powerbook too |
| 14:59.30 | cluecon | file[class]: we all want your Powerbook. |
| 14:59.36 | file[class] | noooooooo |
| 14:59.58 | Ahrimanes | i said it first... lemme have it |
| 15:00.18 | bkw_ | cluecon, ok |
| 15:00.19 | tskjoedt | Hi - anyone's an expert on the sip.conf 'registry' directive? |
| 15:00.35 | asdfblah | Hmmhesays, i need a phone with a lot of programmable buttons and a low price |
| 15:01.08 | Hmmhesays | geebus asdfblah: i asked you that like half an hour ago |
| 15:01.10 | Hmmhesays | lol |
| 15:01.22 | asdfblah | yeah, sorry, stepped away |
| 15:01.22 | Hmmhesays | what is your definition of a low price? |
| 15:01.40 | *** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc) |
| 15:01.47 | rristroph2 | I had presumed that phonegaim was opensource, I guess it isn't . . . |
| 15:01.51 | asdfblah | under 200 would be amazing, but i don't think i'm going to be that lucky |
| 15:01.53 | bkw_ | cluecon, I see it |
| 15:01.58 | Ahrimanes | many features, low price, actually works <- pick any two |
| 15:02.02 | bkw_ | you can login with your user@cluecon.com and password |
| 15:02.22 | Hmmhesays | yeah that's like the rule of people on irc "single, attractive, mentally stable" <-- pick two |
| 15:02.31 | zoa | 16:59:41 up 161 days, 19:49, 5 users, load average: 188.29, 138.99, 74.17 |
| 15:02.34 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 15:02.54 | Ahrimanes | zoa: ah, load is comming down? hehe |
| 15:02.57 | asdfblah | i need to be able to see the status of other extensions |
| 15:03.24 | Ahrimanes | asdfblah: sip show peer XXXXXX ? |
| 15:03.30 | Hmmhesays | asdfblah: on the phone? |
| 15:03.41 | Qwell | zoa: wow, whats with your load? |
| 15:03.41 | asdfblah | Hmmhesays, preferably |
| 15:03.48 | sangee | I want to write AGI in java, but I am new to AGI where can get samples programs? |
| 15:03.55 | Juggie | www.voip-info.org |
| 15:03.58 | *** join/#asterisk roche (~roche@216.194.173.2) |
| 15:04.05 | sangee | thx |
| 15:04.08 | Hmmhesays | asdfblah: check out www.asternic.org <-- fop, works great with a touch screen |
| 15:04.40 | tskjoedt | d |
| 15:07.05 | roche | When I use pickup group over sip extensions I can not use *8 to answer if the call is from another sip extension, only with a zap channel works :/ |
| 15:07.31 | asdfblah | Hmmhesays, yeah i'm played around with it, but it won't work for my appication |
| 15:07.39 | tzafrir | tskjoedt, did you have a question? if you have, you better ask it. An expert may turn out of the black |
| 15:07.43 | asdfblah | i need phones with leds that show if an extension is active |
| 15:07.51 | tskjoedt | Can anyone answer this question: When I register my asterisk server with another asterisk server in sip.conf like this: 'register => user:secret@server.net/dialtone', where 'dialtone' is the extension I want mapped to 'user' (effectively my SIP number at the other server), how does asterisk know in which context to find 'dialtone'?? |
| 15:07.53 | asdfblah | i'm = i've* |
| 15:08.01 | tskjoedt | tzafrir There it was :-) |
| 15:08.44 | file[class] | away I go |
| 15:08.49 | file[class] | end of classssssss |
| 15:08.54 | *** join/#asterisk sudoer_ (~toy@denali.ccs.neu.edu) |
| 15:09.05 | tzafrir | tskjoedt, IIRC it is irrelevant to the register directive and is set in the sip peer/user for that trunk |
| 15:09.20 | sudoer_ | is there a way to test if several channels are working from asterisk without calling the channels? |
| 15:09.52 | tzafrir | tskjoedt, there are actually two contexts: incoming and outgoing. Which one do you refer to? |
| 15:09.53 | sudoer_ | the asterisk box is down, and I can connect via ssh , but I a mnot there and I cant call the server to check |
| 15:10.02 | user1fn | dsfr rocks! |
| 15:10.12 | sudoer_ | the asterisk boz is up I mean, but only one ofthe zaptel channels seems to be working |
| 15:10.19 | sudoer_ | how can I test if both channels are working? |
| 15:10.23 | tskjoedt | It's an incoming context... |
| 15:10.30 | tzafrir | sudoer_, channels of which type? |
| 15:10.37 | sudoer_ | this happens once a month, I suually jsut reset the box, but it is not working now |
| 15:10.48 | sudoer_ | incoming lines on analog |
| 15:10.51 | sudoer_ | zaptel |
| 15:11.08 | tskjoedt | tzafrir I've made up all the contexts myself... |
| 15:11.29 | Zeeek | tzafrir are you going to Madrid? |
| 15:11.35 | tskjoedt | tzafrir The contexts your'e referring to, are they built-in in asterisk? |
| 15:11.37 | blitzrage | Zeeek: are yu? |
| 15:11.38 | tzafrir | Zeeek, sadly, no |
| 15:11.43 | Zeeek | yes blitz, thank you |
| 15:11.48 | sudoer_ | tzafrir: do you have any idea? |
| 15:11.50 | blitzrage | Zeeek: cool! see you there |
| 15:11.57 | tzafrir | I'll be meeting Jack Pulver instead |
| 15:11.59 | Hmmhesays | anyone running the latest firefox? |
| 15:12.06 | Zeeek | tzafrir I asked because I thought I saw your name... must have been somewhere else :) |
| 15:12.06 | sudoer_ | the card is a 4 port tdm04b I thibnk it is callled |
| 15:12.07 | blitzrage | Hmmhesays: I think I am |
| 15:12.13 | asdfblah | 1.0.4? |
| 15:12.16 | Ahrimanes | blitzrage: when are you arriving in madrid? |
| 15:12.17 | Zeeek | blitzrage you are going? When do you arrive? |
| 15:12.20 | tzafrir | Zeeek, My bosses will be there |
| 15:12.21 | Ahrimanes | Zeeek: hehe |
| 15:12.24 | Hmmhesays | www.advancedvoip.com tell me if that renders properly for you |
| 15:12.29 | Zeeek | ah, I thought it was your company |
| 15:12.44 | blitzrage | Zeeek / Ahrimanes: I arrive on Sunday |
| 15:12.49 | tzafrir | Zeeek, you don't confuse me with tzanger by any chance ;-) |
| 15:13.00 | asdfblah | nope |
| 15:13.01 | tskjoedt | I've tried to change the context in the relevant sip peer definition to something obscure, but asterisk still found the right extension. |
| 15:13.07 | Hmmhesays | i confused shido with shido6 the other day |
| 15:13.14 | tskjoedt | tzafrir |
| 15:13.15 | Ahrimanes | blitzrage: wow, early... i arrive tuesday and am trying to find others that are there tuesday evening for some dining and beer |
| 15:13.19 | asdfblah | Hmmhesays, it doens't render properly |
| 15:13.25 | Zeeek | tzafrir no I think I know "who" you are... sort of |
| 15:13.33 | Hmmhesays | ok, i'm going to send them and email |
| 15:13.42 | Zeeek | blitzrage what are you going to do for three days? |
| 15:14.27 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 15:14.36 | tzafrir | tskjoedt, the context is set in the "context" option of the sip peer |
| 15:14.38 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
| 15:15.04 | tzafrir | peer? user? |
| 15:15.13 | seb___ | [channels] |
| 15:15.14 | seb___ | <PROTECTED> |
| 15:15.16 | seb___ | busydetect=1 |
| 15:15.17 | seb___ | busycount=7 |
| 15:15.19 | seb___ | <PROTECTED> |
| 15:15.21 | seb___ | relaxdtmf=yes |
| 15:15.22 | seb___ | callwaiting=yes |
| 15:15.24 | seb___ | callwaitingcallerid=yes |
| 15:15.25 | seb___ | threewaycalling=yes |
| 15:15.27 | seb___ | transfer=yes |
| 15:15.28 | seb___ | cancallforward=yes |
| 15:15.30 | seb___ | <PROTECTED> |
| 15:15.31 | seb___ | usecallerid=yes |
| 15:15.33 | seb___ | <PROTECTED> |
| 15:15.34 | Zeeek | errrr seb___ |
| 15:15.34 | seb___ | echocancel=yes |
| 15:15.36 | seb___ | echocancelwhenbridged=yes |
| 15:15.37 | seb___ | <PROTECTED> |
| 15:15.39 | seb___ | rxgain=0.0 |
| 15:15.40 | seb___ | txgain=0.0 |
| 15:15.42 | seb___ | <PROTECTED> |
| 15:15.44 | seb___ | group=1 |
| 15:15.45 | tskjoedt | tzafrir Yes, I know, I tried to change the 'context' option there, and then it shouldn't have worked, but it still did... But, if you are sure that's how to do it, maybe I made some mistake when testing... |
| 15:15.45 | Ahrimanes | err nopaste plz |
| 15:15.45 | seb___ | pickupgroup=1-4 |
| 15:15.47 | tzanger | wtf "hanguponpolarityswitch" ?? |
| 15:15.48 | seb___ | <PROTECTED> |
| 15:15.49 | seb___ | immediate=no |
| 15:15.51 | seb___ | <PROTECTED> |
| 15:15.53 | tzanger | wtf is wrong with fxs_ks??? |
| 15:15.53 | *** join/#asterisk dwGWF (~dwGWF@63.227.22.62) |
| 15:15.53 | seb___ | context=bell |
| 15:15.55 | seb___ | <PROTECTED> |
| 15:15.56 | tzafrir | sudoer_, you can probably see something in 'zap show channel nnn', but not enough :-( |
| 15:15.57 | Zeeek | hehe |
| 15:15.58 | seb___ | signalling=fxs_ks |
| 15:15.59 | seb___ | callerid=asreceived |
| 15:16.00 | Zeeek | this is why |
| 15:16.01 | seb___ | channel=1 |
| 15:16.03 | Zeeek | we |
| 15:16.04 | seb___ | sorry |
| 15:16.05 | Zeeek | recommend |
| 15:16.07 | Zeeek | pastebin |
| 15:16.12 | Zeeek | ~pastebin |
| 15:16.13 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 15:16.22 | puppet | shit |
| 15:16.23 | dwGWF | is there anything i need to know about faxing getting sent through asterisk? faxing with winfax doesn't work |
| 15:16.28 | Zeeek | everyone can do that once |
| 15:16.28 | puppet | my ISP sucks today |
| 15:16.59 | sudoer_ | tzafrir: yeah, it doesnt help, it seems up |
| 15:17.01 | Zeeek | I wish mine was at least good at that |
| 15:17.05 | Hmmhesays | hrm this dual xeon box on tigerdirect seems to good to be true |
| 15:17.16 | [TK]D-Fender | Totally OT question I just need a hand with : Anyone know how to get the Netbios (samba) client info if you have their IP? |
| 15:17.19 | docelm0 | seb___, Yo mama's car's so ugly, someone broke in just to steal The CLUB. |
| 15:17.41 | Ahrimanes | Hmmhesays: tigerdirect? |
| 15:17.47 | asdfblah | anyone worked wtih the Mitel 5207? |
| 15:17.51 | Hmmhesays | www.tigerdirect.com |
| 15:17.55 | Ahrimanes | ok |
| 15:18.27 | tzafrir | [TK]D-Fender, nmblookup -A IP |
| 15:18.28 | bannerman | I'm using CVS-HEAD-06/05/05, at random times I'm getting segmentation faults.. something like once a day |
| 15:18.49 | Zeeek | blitzrage / Ahrimanes maybe we should plan to meet in the lobby or something at some particular time? |
| 15:19.11 | Ahrimanes | Zeeek: sounds good |
| 15:19.15 | Zeeek | or is there already a plan for tue night? |
| 15:20.05 | blitzrage | Zeeek: I'm sure there will be a time for us to meet up. I'm thinking we should just make an #astricon room so we can all coordinate better like that |
| 15:20.09 | tskjoedt | tzafrir Are you absolutely sure that the registry directive and the sip peer definition are linked through the 'context' option in the peer definition? If so, I'll assume that I made a mistake during testing, and think no more of it :-) |
| 15:20.16 | *** join/#asterisk funxion (~funxion@mtnuser.icgws.com) |
| 15:20.31 | Ahrimanes | blitzrage: good idea.. |
| 15:20.32 | Zeeek | blitzrage good idea - as long as the net access is included int he price of the hotel? |
| 15:20.47 | tzafrir | tskjoedt, I said that the context in the registry directive is *irrelevant* IIRC |
| 15:20.48 | blitzrage | Zeeek: I think it is |
| 15:20.56 | *** part/#asterisk roche (~roche@216.194.173.2) |
| 15:21.12 | Zeeek | I hope it is: I paid $10/day for it in a similar hotel (but the hotel was $40/night priceline) |
| 15:21.17 | *** join/#asterisk Meaty (~cp_simbul@office.abi.ca) |
| 15:21.28 | *** join/#asterisk narcisiss (~narxzilla@131.114.248.237) |
| 15:21.37 | blitzrage | Zeeek: well, there will be enough people also running open access points that it shouldn't be a problem |
| 15:21.39 | [TK]D-Fender | tzafrir : Many thanks |
| 15:21.45 | *** part/#asterisk narcisiss (~narxzilla@131.114.248.237) |
| 15:21.53 | *** join/#asterisk Katty (~root@68.112.15.110) |
| 15:22.03 | Zeeek | blitzrage oh good I can call home :) |
| 15:22.15 | tzafrir | Katty, is the "root" there on purpose? |
| 15:22.29 | Katty | oooh, hold on |
| 15:22.33 | Hmmhesays | http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1332970&CatId=0 |
| 15:22.33 | Ahrimanes | lol |
| 15:22.39 | Hmmhesays | now that... is a good price |
| 15:23.36 | blitzrage | Zeeek: yah, I'll be VoIP'ing it up like theres no tomorrow :) |
| 15:23.58 | Zeeek | hey that's great that you'll be coming from so far, I didn't expect that |
| 15:24.09 | dsfr | dwmw2: take a look at SpanDSP. www.soft-switch.org i believe. |
| 15:24.12 | *** join/#asterisk dwGWF (~dwGWF@63.227.22.62) |
| 15:24.29 | dsfr | bannerman: what is occurring according to debug right before it segfaults? |
| 15:24.44 | blitzrage | welp, moms sent me a gift for grad, so I'm going to pick it up (plus its so damn sunny and nice out!!!) back later |
| 15:25.10 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 15:25.25 | Ahrimanes | hm #astricon is registered, but noone there? |
| 15:26.09 | asdfblah | hen you upload a .wav to an asterisk box for the digital receptionist, where sould the .wav o and what should it be named |
| 15:26.13 | Katty | hmm. |
| 15:27.40 | spyroux | :-D |
| 15:27.51 | tzanger | morning katty |
| 15:27.53 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 15:27.53 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:28.02 | tskjoedt | tzafrir Yes, I noticed. But I don't quite understand why. When someone calls my asterisk server through another asterisk server (that I have registrered with by way of 'register => 36959050:secret@musimi.dk/dialtone', for example), I'm interested in that these people end up in a context that I decide (for example, I don't want them to call out from my server on my bill)... so I thought, there must be some context in play somewhere. Then I looked in |
| 15:28.02 | tskjoedt | the peer definition of the other asterisk server, and noticed the 'context' option you mentioned before, and it fitted the context where the 'dialtone' extension actually was. But when I tried changing this 'context' option to something obscure, the other asterisk server was still able to reach the right extension, i.e. 'dialtone'. I don't get why. Am I off track? |
| 15:28.08 | Zeeek | blitzrage wanna pick up my Polycom and bring it over? |
| 15:28.31 | *** join/#asterisk deg10 (~Denis@200.163.193.247) |
| 15:28.52 | deg10 | I need help on kewrnel 2.6 and CVS zaptel.... |
| 15:29.05 | deg10 | Im getting this message wher ztcfg -vvv |
| 15:29.34 | *** join/#asterisk imcdona (~imcdona@imail.speakeasy.net) |
| 15:29.40 | deg10 | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 15:29.50 | deg10 | Anyone!? |
| 15:30.02 | Juggie | you havnt loaded the modules/drivers for your card |
| 15:30.19 | Katty | Hmmhesays: yes |
| 15:30.20 | Juggie | what distro of linux is it? |
| 15:30.22 | Katty | Hmmhesays: hmmhmmhmm |
| 15:30.23 | deg10 | Is I did: modprobe zaptel and modprobe wcfxs |
| 15:30.27 | deg10 | Fedora Core 2 |
| 15:30.34 | dsfr | asdfblah: place your sounds in /var/lib/asterisk/sounds/ - name it whatever.wav. |
| 15:30.49 | deg10 | Im using a TDM04B with 4 FXO. |
| 15:30.50 | Hmmhesays | *tinkles? tickles |
| 15:30.52 | Katty | Hmmhesays: they're not ticklish |
| 15:31.01 | Hmmhesays | thwarted! |
| 15:31.06 | Katty | DOOM |
| 15:31.11 | Hmmhesays | heh, mine are |
| 15:31.15 | tskjoedt | tzafrir When I register with a peer asterisk server, it _should_ be allowed to call only extensions that are defined in it's peer definition, but it seems not to be the case. |
| 15:31.22 | Hmmhesays | I have a hard time even scratching them |
| 15:31.48 | imcdona | So what do you do if you forget your phone number on IAXTEL? |
| 15:31.59 | sivana | nothing... cuz it's useless |
| 15:32.05 | deg10 | when lsmod, wcfxs is not been used for none. |
| 15:32.18 | anthm | if the context includes any other context it will inherit them |
| 15:32.19 | *** part/#asterisk sylvanus (~sylvanus@jau31-2-82-230-109-158.fbx.proxad.net) |
| 15:32.27 | anthm | you also need to reload |
| 15:32.39 | imcdona | Sivana why is it useless? |
| 15:32.50 | anthm | both sip and iax for sure force you into a specific context on inbound calls |
| 15:32.51 | sivana | imcdona: because it's lagged and nobody uses it |
| 15:33.00 | imcdona | lagged? |
| 15:33.04 | dsfr | sivana: they fixed it up last weekend. |
| 15:33.09 | sivana | dsfr: really |
| 15:33.10 | deg10 | I have the same probelm in two boxes. |
| 15:33.20 | anthm | iax has the * option to let you pick a remote context but it has to be specified in the exten |
| 15:33.32 | Juggie | then i would say deg you did the same thing wrong, twice :) |
| 15:33.32 | deg10 | How may I call Digium support!? |
| 15:33.48 | deg10 | Somebody know them phone number!? |
| 15:33.53 | Juggie | fedora core 2 uses udev |
| 15:33.55 | dsfr | deg10: 1-877-LINUX-ME |
| 15:33.58 | tskjoedt | tzafrir ... hm... OK, I'll assume that it's me who didn't get the testing right. Thanks for your comments. |
| 15:34.00 | Juggie | did you read README.udev? |
| 15:34.18 | Micc | If I'm using ztdummy.ko do I still need to load zaptel.ko and the other 2? |
| 15:34.33 | Zeeek | <anthm> iax has the * option to let you pick a remote context but it has to be specified in the exten |
| 15:34.35 | deg10 | Yes I did, but im not using udev... At least there is no udev related modules or provcess... |
| 15:34.38 | Juggie | you have to give permissions to the modules to create the /dev/.... stuff. |
| 15:34.47 | Juggie | deg10, yeah you are. |
| 15:35.11 | deg10 | So... What I have to do!? |
| 15:35.16 | Zeeek | anthm can you explain specified in the exten |
| 15:35.20 | Juggie | read README.udev |
| 15:35.25 | Juggie | and edit the files it says to edit |
| 15:35.39 | Juggie | unload the modules, reload them, and then run ztcfg -vvv |
| 15:35.42 | deg10 | Ok I will try it out. |
| 15:35.44 | Juggie | try this |
| 15:35.46 | deg10 | Juggie |
| 15:35.50 | Juggie | 'ls -al /dev/zap' |
| 15:35.51 | deg10 | Juggie thanks/. |
| 15:36.02 | deg10 | A moment. |
| 15:36.05 | Juggie | see if you get anything |
| 15:36.05 | *** join/#asterisk jeremywhiting (~jeremy@70-56-99-134.slkc.qwest.net) |
| 15:36.19 | dsfr | deg10: run "make install-udev" in your zaptel source directory. it should place the appropriate udev rules in /etc/udev/rules.d/ for you. |
| 15:37.01 | Juggie | ahh i didnt know about that |
| 15:37.02 | Juggie | nice. |
| 15:38.01 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 15:38.03 | tzanger | dsfr: really... I didn't knwo that |
| 15:38.05 | SpaceBass | howdy |
| 15:38.11 | tzanger | and I'm a veritable zaptel guru :-) |
| 15:38.12 | bannerman | dsfr: I wish I had known that yesterday! |
| 15:38.26 | *** join/#asterisk RoyK (~roy@host-81-191-147-248.bluecom.no) |
| 15:38.44 | SpaceBass | ohhh zaptel guru... just what I need |
| 15:39.10 | SpaceBass | i'm thinking about trying asterisk at my house again... the problem i had before was flashing my zap lines for call waiting |
| 15:39.16 | deg10 | dsfr ill try it too. |
| 15:40.02 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 15:40.02 | perlmonky | with in music on hold is it possible to change the order? It appears to default the the directory handle order... |
| 15:40.19 | Hmmhesays | random |
| 15:40.33 | perlmonky | other than random... say numeric or alpha sort? |
| 15:40.42 | SpaceBass | anyone have a recomendation for a VoIP provider that can port my number? |
| 15:40.49 | SpaceBass | in america |
| 15:42.15 | *** join/#asterisk zaptel (~just@216.194.173.2) |
| 15:42.16 | deg10 | I have /dev/zap directory.... This is what I need!? I have to run make install-udev!? |
| 15:42.55 | zigman | mmh the app_mysql_addon does not compile on cvs ? |
| 15:42.58 | rristroph2 | any one here using the sipXezPhone soft phone ? I am having problems configuring it so it will register with my * |
| 15:43.54 | *** join/#asterisk Jas_Williams (~Jason@host217-44-216-134.range217-44.btcentralplus.com) |
| 15:43.59 | deg10 | Juggie i did a ls -la /dev/zap |
| 15:44.21 | deg10 | Juggie, and I got a lot of directoies. |
| 15:44.26 | deg10 | Im using a TDM04B |
| 15:44.29 | deg10 | 4 FXO. |
| 15:45.37 | deg10 | I think that im not using udev on this system.... |
| 15:46.04 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 15:46.43 | dsfr | deg10: run "ps ax | grep udev". do you see a process called udevd? |
| 15:46.44 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 15:47.09 | deg10 | No I didnt |
| 15:47.40 | deg10 | ps ax |grep udev |
| 15:47.42 | deg10 | <PROTECTED> |
| 15:47.49 | dsfr | deg10: okay, then you're not running udev =P What error are you receiving? |
| 15:48.21 | deg10 | The docs tell mo to do just a modprobe wctdm and not a wcfxs like the older vcersion. |
| 15:48.26 | deg10 | Is thius right!? |
| 15:49.07 | deg10 | modprobe wcfxs |
| 15:49.08 | deg10 | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 15:49.08 | deg10 | FATAL: Error running install command for wcfxs |
| 15:49.10 | dsfr | deg10: Run "modprobe wctdm" if you are running the CVS branches. You'll have to use "modprobe wcfxs" if you are running the tarball Release package. They are the same driver though. |
| 15:49.14 | Qwell | deg10: wctdm |
| 15:49.28 | deg10 | Im usgin the CVS. |
| 15:49.47 | deg10 | I will remove and will try wctdm now. |
| 15:49.54 | dsfr | deg10: Follow this guide http://www.digium.com/index.php?menu=configuration#TDM0XB |
| 15:50.31 | deg10 | Same erros.... |
| 15:50.33 | deg10 | <PROTECTED> |
| 15:50.34 | deg10 | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 15:50.34 | twisted[work] | ooooh |
| 15:50.35 | deg10 | FATAL: Error running install command for wcfxs |
| 15:50.40 | jarrod | hey is there a way to start recording a convo mid-conversation? |
| 15:50.41 | twisted[work] | make sure lspci shows the device |
| 15:50.47 | zoa | jarrod, yes |
| 15:50.49 | zoa | through the manager |
| 15:50.58 | jarrod | zoa: fop? |
| 15:51.04 | deg10 | dsfr I got this docs from Digium... THe error persists.. |
| 15:51.13 | twisted[work] | zoa: iirc, cvs head has the option for one touch record ;) |
| 15:51.25 | twisted[work] | deg10, make sure lspci sees it |
| 15:51.33 | deg10 | twisted[work] it shows.... |
| 15:51.34 | Jas_Williams | deg10: modprobe zaptel then modprobe wctdm |
| 15:51.48 | deg10 | 00:08.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537 |
| 15:51.52 | twisted[work] | ok |
| 15:52.02 | twisted[work] | do an lspci -vv and check the pci ID |
| 15:52.18 | Jas_Williams | deg10: then ztcfg -vvv |
| 15:52.32 | deg10 | lsmod |
| 15:52.33 | deg10 | Module Size Used by |
| 15:52.33 | deg10 | wcfxs 36896 0 |
| 15:52.34 | deg10 | zaptel 240388 1 wcfxs |
| 15:52.36 | deg10 | crc_ccitt 2241 1 zaptel |
| 15:52.36 | deg10 | md5 4161 1 |
| 15:52.36 | deg10 | ipv6 237441 14 |
| 15:52.37 | deg10 | via_rhine 23625 0 |
| 15:52.45 | deg10 | mii 4801 1 via_rhine |
| 15:52.45 | dsfr | Run "lspci -v|grep -A 1 Tiger" - Let me know the 2nd line of output. |
| 15:52.45 | deg10 | dm_mod 56149 0 |
| 15:52.45 | deg10 | joydev 9217 0 |
| 15:52.45 | deg10 | uhci_hcd 31705 0 |
| 15:52.47 | *** mode/#asterisk [+q deg10!*@*] by twisted[work] |
| 15:52.52 | RoyK | ~lart deg10 for not using the pastebin |
| 15:52.54 | RoyK | ~pastebin |
| 15:52.55 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 15:52.58 | twisted[work] | anything longer than 5 lines use pastebin |
| 15:53.06 | *** mode/#asterisk [-q deg10!*@*] by twisted[work] |
| 15:53.32 | *** join/#asterisk roche (~roche@216.194.173.2) |
| 15:53.52 | deg10 | lspci -v|grep -A 1 Jens |
| 15:53.53 | deg10 | 00:08.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537 |
| 15:53.53 | deg10 | <PROTECTED> |
| 15:54.32 | deg10 | sorry about the pastebin |
| 15:54.43 | twisted[work] | deg10, s'ok, you live and learn ;) |
| 15:55.14 | dsfr | Go edit zconfig.h in your zaptel source directory. Uncomment the line "/* #define TDM_REVH_MATCHALL */". Then run "make clean install" in your zaptel source directory. Try running "modprobe wctdm" afterwards. Let me know if that works out for you. |
| 15:55.39 | imcdona | can someone do me a favor and call me via IAXTEL and see if you get my VM?? 700-945-2475 |
| 15:55.41 | dsfr | That define option should be the last option in zconfig.h. So, scroll to the bottom of the file to find it. |
| 15:55.43 | twisted[work] | dsfr, i was gonna say, a9d9 doesn't sound right for some reason |
| 15:55.57 | *** join/#asterisk mAPBIt (mAPBIt@c213-89-241-17.cm-upc.chello.se) |
| 15:56.24 | Jas_Williams | imcdona: Your not registered |
| 15:57.04 | imcdona | jax_williams....I am though "69.73.19.178:4569 imcdona1 24.19.97.94:4569 60 Registered" |
| 15:57.17 | imcdona | wait a sec... |
| 15:57.53 | SpaceBass | anyone used the Hitachi Cable IP-5000 (wifi sip) yet? |
| 15:57.56 | imcdona | now try Jas_willaims |
| 15:57.56 | deg10 | dsfr ok, im comlipping... |
| 15:58.01 | twisted[work] | SpaceBass, yes, they rock. |
| 15:58.03 | deg10 | dsfr oops. |
| 15:58.06 | deg10 | Compiling |
| 15:58.26 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 15:58.32 | twisted[work] | Cresl1n! |
| 15:58.35 | [TK]D-Fender | Anyone here tested the new Digium echo-cancelling T1 card(s)? |
| 15:58.40 | Jas_Williams | imcdona: Goodbye :) only |
| 15:58.46 | Cresl1n | twisted!!! |
| 15:58.54 | twisted[work] | long time no see Cresl1n |
| 15:58.55 | SpaceBass | twisted[work] my big problem with the zyxel was lack of WPA and call waiting support... is the IP-5000 any etteR? |
| 15:59.08 | imcdona | one more time please Jas_williams....I have the console open |
| 15:59.12 | twisted[work] | SpaceBass, WPA i'm not sure on, you'd have to doublecheck, but call waiting, yes. |
| 15:59.36 | Jas_Williams | imcdona: I get a goodbye message only |
| 15:59.38 | Cresl1n | yeppers :-) |
| 15:59.46 | imcdona | Thanks, Jas_williams.....I will troubleshoot my end...thanks |
| 15:59.52 | SpaceBass | twisted[work] ... very new to asterisk, etc.. does call waiting support mean that it will flash a zap channel automaticall? |
| 15:59.59 | twisted[work] | SpaceBass, no. |
| 16:00.04 | Cresl1n | SpaceBass: no way ;-) |
| 16:00.10 | SpaceBass | twisted[work] what about a VoIP provider? |
| 16:00.49 | SpaceBass | or anyone for that matter... |
| 16:00.52 | twisted[work] | SpaceBass, that just means if a second call appearance comes at that phone, you can take it |
| 16:01.04 | imcdona | ok Jas_williams...one more time.....I promise...think I have it fixed now |
| 16:01.37 | SpaceBass | twisted[work] so its up to my configuration to make that 2nd call appear? (basically?) |
| 16:01.37 | deg10 | dsfr IT WORKED OUT |
| 16:01.43 | dsfr | deg10: great |
| 16:02.01 | deg10 | dsfr but i have to modprobe wcfxs.... And not modprobe wctdm... Is this right!? |
| 16:02.11 | deg10 | dsfr What that optin does!? |
| 16:02.18 | twisted[work] | SpaceBass, uhm, huh? |
| 16:02.21 | dsfr | deg10: that is fine. wcfxs and wctdm are the same. |
| 16:02.34 | imcdona | Can someone call my IAXTEL number nad let me know if you get my VM?? 700-945-2475 |
| 16:02.37 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 16:02.37 | deg10 | Ok, anfd qhat about that option on .h !? |
| 16:02.46 | deg10 | and what |
| 16:03.06 | Jas_Williams | imcdona: Phone free ivr |
| 16:03.14 | imcdona | ok...thanks jas... |
| 16:03.38 | SpaceBass | twisted[work] ... sorry, I dont know enough to ask the right questons... i was all set to set up asterisk at home (not @home) a few months ago... biggest problem was that i couldn't flash my zap channels and therefore no call waiting |
| 16:04.06 | SpaceBass | I'm looking for a way to do that, or a VoIP provider that won't kill my TiVos... that and phones that support call waiting |
| 16:05.31 | Hmmhesays | heh |
| 16:05.33 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 16:06.52 | imcdona | OK JAs_williams.....can you do it again? |
| 16:07.33 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:08.17 | *** part/#asterisk imcdona (~imcdona@imail.speakeasy.net) |
| 16:10.47 | deg10 | dsfr THANKS! |
| 16:10.57 | cluecon | imcdona: sound is really bad on my pc at the moment, but I get press 1 to... |
| 16:12.06 | *** join/#asterisk jdg (~jdg@CA03F955.adsl.mana.pf) |
| 16:13.56 | *** join/#asterisk Jeff_BR (~jpcarvalh@201.30.193.134) |
| 16:14.04 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 16:14.12 | Jeff_BR | someone from brazil ? |
| 16:14.30 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
| 16:14.32 | *** join/#asterisk xcore (~rmaraschi@netuno.promitos.com.br) |
| 16:15.56 | xcore | i have an invalid ip, which ports i need redirect to run a asterisk server |
| 16:15.57 | Hmmhesays | some mutherfscker is trying to get me fired |
| 16:16.16 | xcore | to be acessible from the internet |
| 16:16.24 | *** join/#asterisk RChadwell (~rob@rrcs-67-78-145-50.se.biz.rr.com) |
| 16:16.25 | Hmmhesays | I am now angered |
| 16:17.09 | *** join/#asterisk cpatry (~grepmoo@65.39.228.5) |
| 16:17.31 | cluecon | Hmm: on what grounds? |
| 16:17.47 | Hmmhesays | moonlighting |
| 16:17.58 | Hmmhesays | for moonlighting that is |
| 16:18.04 | RChadwell | Is there a good program out there for keeping track of some simple call stats - how many active calls at any one time, how many incoming/outgoing today, etc.??? |
| 16:18.06 | cluecon | Hmm: did you sign a non-compete when you were hired? |
| 16:18.29 | Hmmhesays | no, that's not the issue, I do moonlight... but this person fails to realize I tell my boss about every side job I do |
| 16:18.52 | Hmmhesays | this person went to the next higher person in the chain with 'evidence' of my side projects |
| 16:19.29 | Ariel_ | Hmmhesays, sounds like he is after your job. |
| 16:19.38 | Hmmhesays | He is |
| 16:19.40 | Hmmhesays | twice now |
| 16:19.45 | Hmmhesays | because the higher up boss changed |
| 16:19.49 | cluecon | Hmm: hope your boss defends you. wonder what the motive of this other person really is and if the company would want to keep someone like that around. |
| 16:20.01 | Ariel_ | SpaceBass, you can still use asterisk@home there is ways to create flash as a custom extension setup. |
| 16:21.03 | Hmmhesays | cluecon: I bust my ass for this company, they know that my boss would never allow me to leave because of such a thing |
| 16:21.09 | deg10 | Hi Jeff.... Brasileiro no Freenode! |
| 16:21.35 | deg10 | Jeff_BR Tudo certo!? |
| 16:21.38 | Ariel_ | Hmmhesays, as long as you work for a co. it does not matter everyone is replaceable. |
| 16:21.43 | cluecon | Hmm: busting your ass won't cut it. have you put in those self-destruct codes that activate if you don't use the system for a certain amount of time? |
| 16:22.10 | Hmmhesays | cluecon: sounds about right |
| 16:22.17 | cluecon | would be a shame to see the co. die because you didn't log in but whatever it takes. |
| 16:22.44 | Hmmhesays | haha yeah |
| 16:22.48 | cluecon | ~cluecon |
| 16:22.49 | jbot | well, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
| 16:23.01 | Ariel_ | well it's lunch time. See you in an hour or so. |
| 16:23.53 | Nuxi | M$ is well-known for self-destructing code. |
| 16:24.09 | RChadwell | Is there a good program out there for keeping track of some simple call stats - how many active calls at any one time, how many incoming/outgoing today, etc.??? |
| 16:24.18 | Hmmhesays | it would be so easy, a bash script to format /dev/ |
| 16:24.43 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 16:25.16 | Juggie | dev is dynamic now anyways |
| 16:25.28 | Juggie | so it would not be removeable, not the way it used to be |
| 16:26.03 | Juggie | nuxi, how goes res_php |
| 16:26.10 | Hmmhesays | res_php? |
| 16:26.24 | Nuxi | It's working! I just need to add the * functions to php as an extension. |
| 16:26.37 | Juggie | how is performance? |
| 16:27.04 | Nuxi | Hmmm, how to benchmark... |
| 16:27.40 | *** join/#asterisk gr8nash (~andy@mamabear.si-forest.com) |
| 16:28.06 | Nuxi | cluecon, 0x18 is the sectors per track. No os I've ever found does error checking there. They simple get a divide by zero and crash. |
| 16:29.29 | Nuxi | on a floppy, macs are ok, because they don't read it, they just assume what it should be. |
| 16:30.10 | *** join/#asterisk xcore (~rmaraschi@200.180.23.135) |
| 16:31.18 | *** join/#asterisk Liqnik (~nik@A38c8.a.pppool.de) |
| 16:31.31 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 16:31.56 | Micc | So I guess I can't use SMP on FreeBSD with asterisk. |
| 16:32.14 | *** join/#asterisk brettnem (~brettnem@user-0ccsrb6.cable.mindspring.com) |
| 16:33.26 | *** part/#asterisk Liqnik (~nik@A38c8.a.pppool.de) |
| 16:33.54 | Nuxi | SMP = Sucky Management Portal? ;) |
| 16:34.52 | cluecon | Nuxi: the purpose of causing a crash isn't to make it fatal, just slow em down enough that they can't keep up. |
| 16:35.10 | Nuxi | Juggie, you'll still have to do a custom php build for res_php. you need to --enable-embed=shared |
| 16:35.39 | Nuxi | cluecon, it's not fatal, you just need your own mini-os that can update 0x18. |
| 16:36.17 | *** join/#asterisk Liqnik (~nik@A38c8.a.pppool.de) |
| 16:36.23 | *** part/#asterisk Liqnik (~nik@A38c8.a.pppool.de) |
| 16:36.39 | cluecon | Nuxi: not looking for it to be fatal, just looking to slow them down long enough that I can get away. |
| 16:37.19 | *** join/#asterisk Qorky (Qorky@202.173.160.16) |
| 16:37.45 | Qorky | hello. can ne1 help. I get this when installing asterisk on a fresh install of slackware 10.0 |
| 16:38.03 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
| 16:38.06 | Qorky | ./usr/lib/gcc-lib/i486-slackware-linux/3.3.4/../../../../i486-slackware-linux/bin/ld: cannot find -lXext |
| 16:38.55 | *** join/#asterisk Iceman12C (~Cliff@pool-71-96-177-76.dfw.dsl-w.verizon.net) |
| 16:39.07 | *** join/#asterisk Sephen (~Sephen@proxy5.med-web.com) |
| 16:39.08 | Iceman12C | hello |
| 16:39.18 | Iceman12C | anyone have a working SIP configuration? |
| 16:39.24 | Hmmhesays | is someone actually writing a res_php? |
| 16:39.43 | Juggie | yes |
| 16:39.50 | Hmmhesays | sweet |
| 16:39.56 | *** join/#asterisk hassler (~hassler@r-corp.hcst.com) |
| 16:40.41 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 16:40.41 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 || Everyone test IAXtel! It should be happy now |
| 16:40.57 | Iceman12C | if you have a working SIP configuration, can you please do a 'netstat -tulp' and tell me what the state is for UDP port 5060? |
| 16:41.33 | Nuxi | Hmmhesays, it's being developed with v4, but should work with v5. |
| 16:42.00 | Hmmhesays | yeah v5 is fully backwards compatible, unless you turn that off |
| 16:42.24 | Qorky | [TK]D-Fender: is it looking for X ? |
| 16:42.27 | cluecon | Sephen: are you wanting to initiate a call via a web interface that connects the 600 to another extension? if so, that's easy to do. just use either the Manager API or create a call file based on the web input. |
| 16:42.35 | hassler | anyone have thoughts on best SIP/IAX LD provider(s)? I've tried nufone with acceptable results. Looking specifically at iConnect or voicepulse. Any others? Experiences? |
| 16:42.50 | blop | res_php? |
| 16:43.14 | Hmmhesays | yeah, you can generate a call from the manager in about 10 lines of code |
| 16:43.39 | Iceman12C | anyone? |
| 16:44.01 | Hmmhesays | Iceman12C probably all of us have a working sip config |
| 16:44.06 | [TK]D-Fender | Qorky : Not sure what package series..... |
| 16:44.06 | terrapen | dammit |
| 16:44.06 | hassler | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 16:44.16 | Iceman12C | thanks hassler |
| 16:44.16 | terrapen | Nufone is not letting me dial Uganda today |
| 16:44.23 | [TK]D-Fender | Qorky : I've just never had a problem running Slack 10.0 and 10.1 for it. |
| 16:44.24 | terrapen | i always have trouble making international calls |
| 16:44.32 | Iceman12C | hassler: it doesn't show LISTEN as the state? |
| 16:44.49 | hassler | not on a UDP connection |
| 16:45.00 | cluecon | Iceman: same results here as hassler, no state shown. |
| 16:45.01 | Iceman12C | is asterisk on 5060 TCP? |
| 16:45.23 | terrapen | <PROTECTED> |
| 16:45.25 | terrapen | wtf |
| 16:45.32 | Sephen | Cluecon: You'd have to lift the handset for that to work right? Wouldn't that ring your extension? We're looking for a way basically to tie our CRM's contact list to our phone, so a user can sit at their desk, click on a client to call, and it puts the phone into speaker-phone mode, and dials for us. I can do this with a 7960, but I didn't see a way to do this with the Polys. |
| 16:45.38 | Iceman12C | my problem is that my phone is trying to register with asterisk, but my machine keeps returning a ICMP port unreachable packet |
| 16:45.45 | Iceman12C | iptables is off |
| 16:45.49 | hassler | firewall issue? |
| 16:46.13 | terrapen | can anyone else call uganda today? |
| 16:46.18 | Iceman12C | iptables is off as far as I can tell.. |
| 16:46.44 | hassler | any firewall between your SIP phone and asterisk? |
| 16:46.59 | Iceman12C | nope.. same local network |
| 16:46.59 | cluecon | Sephen: you would have to look into the capabilities of the 600. I haven't used one, but I would think that you are going to be forced to have user interaction unless you have the phone set to autoanswer. |
| 16:47.14 | cluecon | Iceman: what do you get with a nmap on the server IP? |
| 16:47.48 | Iceman12C | running it now |
| 16:47.57 | Iceman12C | when i use ethereal on the server |
| 16:48.08 | Iceman12C | i'm able to see the incoming SIP register packet from the phone |
| 16:48.24 | Iceman12C | and then i see the server immediately send a ICMP port unreachable back to the phone |
| 16:49.22 | Iceman12C | ok, i just did a nmap on the UDP ports of the server |
| 16:49.45 | Iceman12C | all ports are open except 4.. and 5060 is not in those 4 |
| 16:49.58 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 16:52.28 | asdfblah | has anyone worked with the atcom at-320 |
| 16:53.05 | Iceman12C | any ideas? |
| 16:53.42 | cluecon | Iceman, not sure. |
| 16:54.16 | af_ | I am searching .it iax2 termination..... |
| 16:54.57 | cpatry | af_: place a request in -biz, im sure you'll get thousand of replies. |
| 16:55.01 | Iceman12C | oh =/ |
| 16:55.20 | af_ | thanks cpatry |
| 16:55.27 | af_ | nice idea :) |
| 16:55.28 | hassler | I have an at-320 sitting on my desk |
| 16:55.35 | *** join/#asterisk heison (~heison@w3.somanetworks.com) |
| 16:55.38 | Cresl1n | anybody here have any mad DSP skills? |
| 16:56.00 | Cresl1n | hrm.... no #dsp |
| 16:56.36 | durufle | anyone have experience with asterisk and fax functions? |
| 16:56.50 | heison | bkw_: are you around? |
| 16:57.07 | tzanger | Cresl1n: check out #musicdsp on efnet |
| 16:58.03 | Cresl1n | merci :-) |
| 16:58.05 | heison | ~seen JerJer |
| 16:58.07 | jbot | jerjer is currently on #asterisk |
| 16:58.18 | cluecon | bkw is more square than round. |
| 16:58.22 | tzanger | Jerjer is currently in a hot tub with a dozen sweedish swimsuit models |
| 16:58.32 | cpatry | Cresl1n: now you're talking! :) |
| 16:58.49 | Cresl1n | heh |
| 16:58.57 | Cresl1n | ~jerjer |
| 16:58.58 | jbot | somebody said jerjer was the guy who runs nufone |
| 16:59.18 | Cresl1n | ~cpatry |
| 16:59.19 | heison | i'm looking at codec_speex.c, trying to figure out what bitrate it is set to operate... does anyone know what does quality = 8 and complexity = 2 mean? |
| 16:59.28 | Hmmhesays | haha just got a call from my manager.... life is good |
| 16:59.33 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
| 16:59.42 | tzanger | well I might just be shooting in the dark here but I'm pretty sure that sets the quality to 8 and the complexity to 2 |
| 16:59.46 | tzanger | :-) |
| 16:59.49 | cluecon | Hmm: is the other person being escorted out of the building yet? |
| 16:59.51 | dca[laptop] | hey anthm |
| 17:00.09 | Hmmhesays | cluecon: we need him to do the monkey work... what they said to me made me laugh though |
| 17:00.16 | heison | tzanger: yeah, that i can see... but what does 8 and 2 represent? |
| 17:00.21 | cluecon | Hmm: is it repeatable? |
| 17:00.38 | heison | tzanger: it's being used by speex_encoder_ctl() |
| 17:00.52 | anthm | hi |
| 17:01.11 | cluecon | 8 and 2 are empirical measurements that relate specifically to the measured variable but as for what they do, well, i dunno. |
| 17:01.12 | Hmmhesays | yeah.... he went over our managers head and that pissed her off something fierce.... I guess the higher up manager took one look at his 'proof' of my moonlighting, handed it back to him and said "I Don't want this" |
| 17:01.34 | *** join/#asterisk _omer (~dfsdf@203.215.180.250) |
| 17:01.47 | _omer | hello |
| 17:01.47 | [TK]D-Fender | Hmmhesays : No-one likes a tattle-tale..... |
| 17:01.49 | dca[laptop] | anthm: got a sec? got some interesting info about our deployment attempt |
| 17:02.00 | _omer | Can I use Asterisk for calls termination ? |
| 17:02.10 | cluecon | Hmmhesays: exactly. |
| 17:02.18 | cluecon | _omer: what kind of calls termination? |
| 17:02.34 | terrapen | SOMEBODY PLUG THE UGANDAN TELEPHONE LINE BACK IN! |
| 17:02.43 | dca[laptop] | anthm: specifically contrasting RT and astdb, which, unknown to me, has been what causes all my grief |
| 17:03.02 | _omer | cluecon: for example: calls are coming over the IP thru SIP and want to router it to ...lets suppose ..Free world dialup |
| 17:03.06 | tzanger | cluecon: hahahah I'll have to remember that one. "... are imperical measurements that relate specifically to the measured variable..." |
| 17:03.11 | cluecon | terrapen: we're sorry, but Uganda is currently away from the office. If you would like to leave a message, someone will get back with you shortly. |
| 17:03.22 | terrapen | heh |
| 17:03.35 | cluecon | _omer: sure |
| 17:03.40 | anthm | ok i'm kinda busy but i can read |
| 17:03.42 | Hmmhesays | the funniest part was... his proof was a help website I ran about a 1 1/2 ago.. that he had already brought to one boss |
| 17:03.44 | *** join/#asterisk bofh42 (~bofh42@p54821B6F.dip0.t-ipconnect.de) |
| 17:03.45 | anthm | if you care to elaborate |
| 17:03.53 | anthm | i may just forget to look sometimes |
| 17:04.10 | tzanger | Hmmhesays: what trouble are you getting in and out of? |
| 17:04.43 | Hmmhesays | nothing, co-worker trying to convince the boss i'm bad for the company |
| 17:04.46 | cluecon | Hmm: a website? exactly what kind of business are you in? I mean, a website? I run 10 of them out of my house. |
| 17:04.55 | tzanger | Hmmhesays: ooh I love those people |
| 17:04.59 | tzanger | I usually end up getting them fired |
| 17:05.06 | Hmmhesays | a pay for support site |
| 17:05.12 | Hmmhesays | for various ip products |
| 17:05.30 | terrapen | brb reboot |
| 17:05.46 | cluecon | Hmm: thats when the phrase "you get what you pay for" comes in handy. |
| 17:06.18 | Hmmhesays | he would be fired right now if I demanded it.... but he so good and mindless bi@tch work |
| 17:06.25 | Hmmhesays | and/at |
| 17:06.45 | tzanger | Hmmhesays: good now make him clean the mens restroom with a toothbrush... preferably his |
| 17:06.51 | _omer | cluecon: can I get any sample configuration for calls termination? |
| 17:07.06 | Hmmhesays | worse, I'll make him sit and clean old mitel dialers for a few hours |
| 17:07.11 | tzanger | hahaha |
| 17:07.25 | Hmmhesays | we have about 3,000 of them sitting around here just waiting for a good scrubbing |
| 17:07.46 | cluecon | voip-info.org |
| 17:08.07 | *** join/#asterisk pashah (~pashah@ns.itconnection.ru) |
| 17:08.10 | pashah | hello |
| 17:08.54 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 17:08.54 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 || Everyone test IAXtel! It should be happy now |
| 17:09.12 | pashah | I am using M option for Dial, and having problems passing a variable back to context from the macro. what should I read? any pointers? |
| 17:09.56 | *** join/#asterisk Gombi (~Gombi@139-153.mc.royaume.com) |
| 17:10.48 | _omer | Zeeek: thanks |
| 17:12.54 | _omer | Zeeek: is this sample for calls termination ??? receive calls over the IP(SIP) and rout them to SIP Server like Free world dialup |
| 17:13.19 | Hmmhesays | lots of people are doing that |
| 17:13.40 | Gombi | hi everyone. |
| 17:13.47 | Hmmhesays | haha now he's going to be really angry if they send me to cluecon |
| 17:13.58 | *** part/#asterisk Gombi (~Gombi@139-153.mc.royaume.com) |
| 17:14.13 | cluecon | Hmmhesays: you must come to cluecon! You must! |
| 17:14.28 | Hmmhesays | yes I must |
| 17:14.38 | Hmmhesays | I must meet these strange people i talk to on the intarwebs |
| 17:15.09 | cluecon | sweet. core dump. |
| 17:15.21 | Hmmhesays | now that's hot. |
| 17:16.01 | *** join/#asterisk HAiRYANiMAL (~wonderboy@gw0-cust-MOBILE-PRODUCTIVITY-COM.slkc.eli.net) |
| 17:16.08 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 17:16.32 | cluecon | was playing with chanspy and it wigs out. segfault with a core dump and * went bye bye. |
| 17:16.50 | cluecon | maybe that means i need to call it a day. |
| 17:16.57 | *** part/#asterisk Iceman12C (~Cliff@pool-71-96-177-76.dfw.dsl-w.verizon.net) |
| 17:16.58 | PBXtech | that sucks |
| 17:16.58 | PBXtech | :) |
| 17:17.09 | cluecon | good thing its just a dev box. |
| 17:17.14 | *** join/#asterisk midnight- (midnight@64.7.183.85) |
| 17:17.15 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 17:17.31 | PBXtech | never give up soldier |
| 17:18.04 | terrapen | gotta love random mp3 shuffle |
| 17:18.14 | Hmmhesays | chanspy still in development? |
| 17:18.20 | terrapen | "An old man...turned 98...he won that lottery...and died the next day" |
| 17:18.30 | zoa | aha |
| 17:18.35 | zoa | a belgian song |
| 17:18.36 | zoa | :) |
| 17:18.41 | Hmmhesays | if I need to spy on a channel I usually do it the hard way |
| 17:18.45 | Hmmhesays | with a manager redirect |
| 17:18.48 | zoa | when your presidents an actor |
| 17:18.49 | terrapen | "It's a black fly...in your Chardonnay...it's a death row pardon (hard-on?)...five minutes too late |
| 17:18.51 | zoa | someeeehitng wrong |
| 17:18.56 | *** join/#asterisk TESTER2 (~Cyber@modemcable219.42-81-70.mc.videotron.ca) |
| 17:19.15 | zoa | oh no |
| 17:19.17 | TESTER2 | Someone knows where to find a good example of iax trunking with encryption? |
| 17:19.17 | zoa | im wrong |
| 17:19.26 | zoa | tester2 its not working |
| 17:19.30 | zoa | we tried it for a while |
| 17:19.31 | zoa | its broken |
| 17:20.18 | TESTER2 | So what is better: iax alone, iax on a vpn or sip on vpn? |
| 17:20.34 | zoa | iax alone |
| 17:20.35 | bannerman | I'm having real issues with stability with CVS-HEAD from 6/5/05, but I need to be using cvs-head.. |
| 17:20.40 | zoa | dont use encryption |
| 17:20.49 | terrapen | TESTER: which is better? man or woman? |
| 17:21.21 | terrapen | the answer is, whichever you like best |
| 17:22.32 | TESTER2 | I think, I'll begin with a iax alone and when the ecryption will be available, I'll configure it |
| 17:22.45 | TESTER2 | any idea when it will be functionnal? |
| 17:26.45 | *** join/#asterisk xcore (rmaraschin@200.180.23.135) |
| 17:27.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 17:28.17 | xcore | hey people |
| 17:28.25 | xcore | anybody can help me |
| 17:28.45 | xcore | i set a asterisk server in a valid ip |
| 17:28.59 | xcore | but when my friend try to connect with this server |
| 17:29.08 | xcore | the server respond to invalid ip from my friend |
| 17:29.34 | MikeJ[Laptop] | xcore, set up the nat settings right in sip.conf.. |
| 17:29.43 | xcore | hmmmm, |
| 17:29.46 | MikeJ[Laptop] | read the sample config or the wiki for details |
| 17:29.50 | MikeJ[Laptop] | brb |
| 17:29.50 | *** join/#asterisk Brucex (~bruce@200.29.14.68) |
| 17:29.51 | xcore | try to explain me |
| 17:29.53 | Brucex | howdy ! |
| 17:30.07 | xcore | ok ok, i'm a jerk |
| 17:30.09 | tzafrir_laptop | what's the cvs tag for release 1.0.7? |
| 17:30.09 | xcore | =P |
| 17:31.27 | Hmmhesays | hey who would like to send me an IP600 |
| 17:31.34 | tzafrir_laptop | Isn't it V1-0-7? (no) |
| 17:31.42 | *** join/#asterisk Gombi (~Gombi@139-153.mc.royaume.com) |
| 17:31.57 | Gombi | Hi, |
| 17:32.10 | Gombi | my question is about the compatibility of Asterisk. |
| 17:32.21 | hassler | anyone have thoughts on best SIP/IAX LD provider(s)? I've tried nufone with acceptable results. Looking specifically at iConnect or voicepulse. Any others? Experiences? |
| 17:32.29 | Hmmhesays | voipjet |
| 17:32.40 | Hmmhesays | free test account anyway |
| 17:33.37 | Hmmhesays | Hrm, I would like a sip video phone on my desk |
| 17:35.48 | Gombi | I need to know which dialogic Cards will work with Asterisk... We're looking to buy a Intel NetStructure ZT 5085 .. |
| 17:36.00 | TESTER2 | So any idea when the encryption will be available to iax trunking? |
| 17:36.27 | Hmmhesays | when you write it? |
| 17:36.35 | Gombi | but now what i need is to know the Cpci cards that can be used with this box to do Ip Telephony. (SIP) |
| 17:37.24 | crash3m | Cpci? |
| 17:37.29 | Gombi | Right! |
| 17:37.35 | TESTER2 | Hmmhesays: already suppose to be started |
| 17:37.52 | MikeJ[Laptop] | tzafrir_laptop v1-0-7 |
| 17:38.05 | MikeJ[Laptop] | cvs co -r v1-0-7 asterisk |
| 17:38.10 | Gombi | We want to provide a Telephony system that will never go down. so with the Cpci we can insert or change cards! |
| 17:38.13 | crash3m | I'm going o have to guess digium |
| 17:38.20 | Hmmhesays | anyone know if you can get just a video phone from packet8? |
| 17:38.23 | crash3m | dunno if its cpci or not though heh |
| 17:38.30 | Hmmhesays | I don't see an option on their site |
| 17:38.42 | crash3m | Hmmhesays: http://www.google.com/search?hl=en&lr=&q=digium+cpci&btnG=Search |
| 17:38.43 | greg_work | TESTER2: you can use something like openvpn, today |
| 17:39.09 | TESTER2 | yes this is my solution #2 |
| 17:39.18 | Hmmhesays | crash3m: why are you linking me to that? |
| 17:39.21 | Hmmhesays | yeah openvpn works great |
| 17:39.32 | crash3m | Hmmhesays: doh sorry, misfire |
| 17:39.33 | TESTER2 | the problem is that vpn add cpu load and delay |
| 17:39.37 | crash3m | Gombi: ^^^ ;) |
| 17:39.39 | outofjungle | i want to install mod_php 5, how do i unhard mask it? |
| 17:40.04 | Hmmhesays | TESTER2 if you are having delay over an openvpn tunnel you are having issue's |
| 17:40.08 | greg_work | TESTER2: or try zebedee |
| 17:40.31 | Hmmhesays | openvpn works and it works well |
| 17:41.18 | outofjungle | nevermind, i got it |
| 17:41.32 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 17:41.36 | SpaceBass | got (yet) another call waiting question... how does CW work with VoIP providers like BroadVoice, etc... what setup is required on the asterisk side to be able to answer the calls? |
| 17:42.31 | Hmmhesays | call waiting is dependant on how your endpoing handles an incoming call when you already have one active |
| 17:42.39 | Hmmhesays | *endpoing/endpoint |
| 17:42.48 | *** join/#asterisk _DAW (~bob@cable-68-114-110-210.sli.la.charter.com) |
| 17:42.54 | Gombi | If we buy Dialogic Cpci Cards and put it in the intel Netstructure ZT 5085... Is the only condition is that intel had a Linux Driver or there other things that i need to think of.. like Asterisk compatibility .. or asterisk will take linux drivers and works through it? |
| 17:42.57 | SpaceBass | Hmmhesays and by endpoit do you mean phone or device? |
| 17:43.05 | *** join/#asterisk PakiPenguin (~pingu@202.147.163.114) |
| 17:43.29 | Hmmhesays | by endpoint I mean whatever piece of equipment you got that's terminating the ip leg of the call |
| 17:43.36 | SpaceBass | gotcha |
| 17:44.09 | Hmmhesays | like my 2 tin cans and string don't have call waiting at all.... the voice will just barge right through |
| 17:44.22 | _DAW | Hello, Can someone tell me how many mpg123 processes should normally be running with * ? |
| 17:44.24 | Hmmhesays | as many simultaneous calls as you want from the treehouse to the kitchen |
| 17:44.43 | SpaceBass | so assuming we are talking about a certian IP SIP phone with "call waiting" features (assuming a flash button) then does it send some kind of standard message/signal that flashes the line? |
| 17:45.00 | crash3m | _DAW: I'm sure it depends on how many playback sessions are in progress |
| 17:45.25 | Hmmhesays | SpaceBass: usually the flash hook is internal |
| 17:45.42 | *** join/#asterisk mjman (~asterisk@205.158.42.66.ptr.us.xo.net) |
| 17:45.44 | Hmmhesays | sometimes the phones will send out a sip info message |
| 17:46.11 | SpaceBass | Hmmhesays: sorry to be dense, but not sure I follow.. internal to what? the phone? so does that flash hook put one call on hold and answer the other? |
| 17:46.18 | _DAW | crash3m: Just specifing one file in musiconhold.conf seems to spawn two processes. |
| 17:46.32 | Hmmhesays | SpaceBass: pretty much |
| 17:46.58 | *** join/#asterisk samoied (douglas@200.142.138.56) |
| 17:47.00 | dwGWF | why does our T1 here at the office keep cutting people off? we're running Asterisk PBX |
| 17:47.04 | samoied | Hi all! |
| 17:47.11 | mjman | Hi, I am trying to set up an extension, so that if one dials 8555, it will call the voicemail box for their extension. This is what I have: exten => 8555,1,Answer exten => 8555,2,Wait(1) exten => 8555,3,VoicemailMain(${CALLERIDNUM}@default) exten => 8555,4,Macro(hangupcall) |
| 17:47.15 | samoied | I have 8 ISDN phones |
| 17:47.19 | _DAW | crash3m: I have one running six mpg123 processes and it seems to be contributing to some strange behavior. |
| 17:47.19 | mjman | each exten is on a new line |
| 17:47.21 | *** join/#asterisk arctic0 (arctic0@arctic0.active.supporter.pdpc) |
| 17:47.26 | Hmmhesays | mjman www.pastebin.ca |
| 17:47.26 | *** join/#asterisk [hC] (~hardcore@8.10.2.5) |
| 17:47.27 | samoied | how to I connect them with asterisk? |
| 17:47.36 | mjman | Hmmhesays: thanks |
| 17:47.37 | SpaceBass | Hmmhesays thanks for the info! |
| 17:47.44 | SpaceBass | next step is for me to get some hardware to play with I guess |
| 17:47.53 | crash3m | _DAW: 6 mpg123 processes are causing issues? |
| 17:47.53 | Hmmhesays | SpaceBass: np... i hope this isn't considered moonlighting now... i may be fired |
| 17:48.07 | arctic0 | Hi, where can i get sip numbers for testing? |
| 17:48.09 | Hmmhesays | haha i joke |
| 17:48.09 | samoied | Is possible to use a Digium E1 card? |
| 17:48.22 | SpaceBass | Hmmhesays i dont know... i may have to tell your bosss boss, I have proof |
| 17:48.22 | [hC] | Im testing out some CDR mysql billing stuff here, one server seems to do it just fine, my other server doesnt ever write anytihng into the database. The one difference is server A actually has sip phones on it, and server b is just connected to via IAX. does IAX2 not CDR log or something? |
| 17:48.29 | Hmmhesays | PROOF |
| 17:48.33 | samoied | or I have to use a beronet card? |
| 17:48.36 | Hmmhesays | no... please don't. |
| 17:48.41 | SpaceBass | ;) |
| 17:48.43 | mjman | Hi, I am trying to set up an extension, so that if one dials 8555, it will call the voicemail box for their extension. This is what I have: http://pastebin.ca/13736 when I call 8555, however, i am still prompted for my mailbox number. What am i doing wrong/???? |
| 17:49.08 | Hmmhesays | mjman: put your extensions.conf entries up at pastebin.ca then link us so we can look |
| 17:49.10 | SpaceBass | mjman there is a voicemail app and a voicemailmain app |
| 17:49.12 | _DAW | crash3m: Not sure if it is related to the 6 processes or not. I was just wondering if that many was abnormal for only one moh file. |
| 17:49.39 | SpaceBass | I dont know enough to tell you, but you need to pass the calling extension to the voicemail() app i think .. voicemail(s) maybe ? |
| 17:49.44 | crash3m | I'm not sure heh |
| 17:49.59 | Gombi | can someone gime an answer to this big question ? :P |
| 17:50.00 | Gombi | If we buy Dialogic Cpci Cards and put it in the intel Netstructure ZT 5085... Is the only condition is that intel had a Linux Driver or there other things that i need to think of.. like Asterisk compatibility .. or asterisk will take linux drivers and works through it? |
| 17:50.33 | Hmmhesays | haha |
| 17:50.40 | samoied | Anyone know a method to connect ISDN phones with asterisk? |
| 17:50.45 | Hmmhesays | mjman: nevermind I didn't read far enough into your line |
| 17:51.20 | mjman | SpaceBass: passing the caller number to Voicemail just results in asterisk hanging up |
| 17:51.21 | Hmmhesays | mjman, you have voicemail setup properly? |
| 17:51.21 | *** join/#asterisk Tili (Tili@202-133-65-97-dialup.sat.net.pk) |
| 17:51.32 | mjman | Hmmhesays: yes, I have voicemail set up |
| 17:51.52 | samoied | Gombi: You have to writa a asterisk channel to support this |
| 17:52.02 | samoied | s/writa/write/ |
| 17:52.07 | mjman | Hmmhesays: and I can dial into it just fine, but I want to configure it so that it automatically logs into the voicemail box for the current extension |
| 17:52.18 | SpaceBass | mjman I'm no expert |
| 17:52.25 | SpaceBass | rather, just the oppisite |
| 17:52.28 | mjman | haha |
| 17:52.37 | mjman | thats ok, me too |
| 17:52.39 | Hmmhesays | mjman: saydigits(${CALLERIDNUM}); make sure it's trying to log into the right mailbox |
| 17:53.01 | Hmmhesays | quick and dirty way to check |
| 17:53.06 | mjman | Hmmhesays: I will try it now |
| 17:54.31 | *** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
| 17:54.48 | *** join/#asterisk samuel_ (~samuca@200.180.4.168) |
| 17:54.52 | mjman | Hmmhesays: yes, it says the correct digits |
| 17:55.02 | samuel_ | hello!? |
| 17:55.13 | Hmmhesays | mjman: comment out the hangup macro |
| 17:55.21 | mjman | Hmmhesays: ok |
| 17:55.31 | Hmmhesays | why do you have a hangup macro? |
| 17:55.51 | mjman | Hmmhesays: I am using AMP, and its the way it set it up |
| 17:55.52 | samuel_ | anyone can help me.. i`m having some dificult to allow de g723 codec on asterisk..!??! |
| 17:56.09 | Hmmhesays | ahh AMP: the creator of many problems |
| 17:56.14 | mjman | haha |
| 17:56.19 | mjman | it seems pretty good to me |
| 17:56.23 | Hmmhesays | samuel_ I hope you aren't trying to decode g723 |
| 17:56.36 | SpaceBass | amp is pretty, but try and change anything yourself and its a nightmare |
| 17:56.45 | mjman | Hmmhesays: still doesnt work. |
| 17:57.00 | Hmmhesays | not true, I've edited amp generated config files without issue before |
| 17:57.02 | SpaceBass | im torn on using it... makes admin pretty easy for a newbie like me, but it loves overwriting my config files which is obnoxious |
| 17:57.14 | samuel_ | Hmmhesays: no I'm just trying do make the asterisk work with codec g723.. |
| 17:57.34 | Hmmhesays | samuel_ you mean passthru g723? |
| 17:57.36 | samuel_ | Hmmhesays: sorry with my english i'm brazilian.. |
| 17:57.43 | samuel_ | Hmmhesays: yes.. |
| 17:57.52 | Hmmhesays | send me some pictures of pretty brazilian girls and I will help you |
| 17:57.54 | mjman | as long as you don't touch the _additional.conf files, AMP wont cause trouble |
| 17:58.14 | *** part/#asterisk Brucex (~bruce@200.29.14.68) |
| 17:58.19 | Hmmhesays | ;D I joke samuel |
| 17:58.22 | Hmmhesays | Hey Katty |
| 17:58.38 | samuel_ | Hmmhesays: ok.. here have lot of pretty girls, at other time i sendit!! |
| 17:58.52 | zigman | mmh anyone got a sample syntax for DBPUT in cvs ? |
| 17:58.58 | zigman | it is not very well documented |
| 17:59.12 | tzanger | pamples? damn, here I thought I was your one true love, katty. |
| 17:59.24 | Hmmhesays | samuel_: what error is it giving you? |
| 17:59.31 | Katty | tzanger: i love you too. |
| 18:00.02 | tzanger | Katty: but I'm ont your one... *cries* |
| 18:00.11 | samuel_ | Hmmhesays: i'm follow this howto: http://www.readytechnology.co.uk/open/g723.1/INSTALL-041103.txt |
| 18:00.12 | Ariel_ | hello Katty |
| 18:00.30 | samuel_ | and there is a part of howto that i dont understand.. |
| 18:00.38 | Hmmhesays | samuel_ you won't get any help installing that here most likely |
| 18:01.11 | Hmmhesays | you don't need to compile a g723 module if you are only doing passthru calls |
| 18:01.24 | samuel_ | humm |
| 18:01.27 | opus__ | software patents are for facists |
| 18:01.42 | funxion | does anyone know how to make a te100 emulate the following cisco command "isdn channel-id invert extend-bit"? |
| 18:01.44 | Hmmhesays | if you need to transcode g723, then you need a module |
| 18:01.46 | [hC] | how can i tell if the cdr_addon_mysql.so module loaded correctly into asterisk? it doesnt show up on reload, even on the boxes that its working on. |
| 18:01.49 | alt | opus__: I'm not going there except to agree :P |
| 18:01.53 | tzafrir_laptop | Hmmhesays, something I don't understand: aren't those software patents? and if so: do they apply outside the US and Japan? |
| 18:02.10 | samuel_ | Hmmhesays: how i can do this? |
| 18:02.12 | Hmmhesays | they apply anywhere the government will enforce the rules |
| 18:02.14 | Ariel_ | Does any one have a how to for the Polycom setup to work via NAT... Argh I am stumpped on this one. |
| 18:02.28 | Hmmhesays | samuel_ disallow=all allow=g723 |
| 18:02.28 | Gombi | what is a S/WRITA/WRITE ??? |
| 18:02.33 | Hmmhesays | in your sip.conf |
| 18:02.40 | funxion | [hC] it should tell you it loaded on reload |
| 18:02.45 | samuel_ | Hmmhesays: ok.. i have this situation.. |
| 18:02.53 | [hC] | hmm. one box isnt, yet it is infact logging there. |
| 18:03.06 | funxion | weird |
| 18:03.31 | SpaceBass | phone recomendation help... cisco 7912 or polycom sountpoint 600? |
| 18:03.56 | samuel_ | Hmmhesays: I have a client sip g723 and a gateway h323 g723.. |
| 18:04.02 | Ariel_ | SpaceBass, wow between those two The Polycom hands down. |
| 18:04.08 | SpaceBass | really? |
| 18:04.13 | SpaceBass | thanks |
| 18:04.16 | [hC] | ok show modules is showing me that its not loaded on #2, but its in the path.. any way to debug why its not loading? |
| 18:04.18 | Hmmhesays | and you need to convert from sip to h.323? |
| 18:04.34 | TESTER2 | Does the iax protocol send a lot of packets when no communication between peers (keepalive, etc.)? |
| 18:04.40 | tzafrir_laptop | [hC], asterisk -vvvvvvvvvvv |
| 18:04.56 | samuel_ | yes |
| 18:04.57 | *** part/#asterisk Gombi (~Gombi@139-153.mc.royaume.com) |
| 18:04.57 | Hmmhesays | set verbose 11,000,000 |
| 18:05.04 | funxion | [hC] change your logging levels in logger.conf |
| 18:05.06 | SpaceBass | Ariel_ I trust you, but out of curriosity why do you prefer the polycom? |
| 18:05.10 | tzafrir_laptop | on rapid: /etc/init.d/asterisk debug, to avoid permitions issues later |
| 18:05.11 | Hmmhesays | samuel: you don't need a g723 module to do that |
| 18:05.19 | *** join/#asterisk Lee__ (~Lee__@cpe-69-203-211-144.nyc.res.rr.com) |
| 18:05.23 | Hmmhesays | do you have chan_h323 or chan_oh323 installed? |
| 18:05.26 | funxion | does anyone know how to make a te100 emulate the following cisco command "isdn channel-id invert extend-bit"? |
| 18:05.35 | funxion | te110p sry |
| 18:05.37 | *** join/#asterisk cuilean (~cuilean@ool-44c16f4c.dyn.optonline.net) |
| 18:05.38 | samuel_ | Hmmhesays: yes I have this installed and working.. |
| 18:05.43 | Hmmhesays | which one? |
| 18:05.47 | tzafrir_laptop | bye folks |
| 18:05.55 | Ariel_ | The polycom 600 is like the cisco 7960G and polycom includes there sip lisc cisco charge extra for everything. |
| 18:06.13 | SpaceBass | cool... works for me |
| 18:06.14 | samuel_ | Hmmhesays: just allow=g723 on sip.conf and h323.conf and would be work? |
| 18:06.23 | Lee__ | if I say Record(file:gsm) how is the default path determined? |
| 18:06.27 | SpaceBass | the polycom says "6 lines" what does that mean? 6 sip exensions> |
| 18:06.29 | samuel_ | Hmmhesays: let me see.. |
| 18:06.32 | Hmmhesays | you are using chan_h323? not chan_oh323 |
| 18:06.36 | Hmmhesays | is this correct? |
| 18:06.46 | cuilean | Lee--- Working directory of the asterisk process. |
| 18:06.54 | Lee__ | SpaceBass: yes |
| 18:06.54 | Ariel_ | Lee__, look in the asterisk.conf file |
| 18:07.11 | cuilean | Lee-- Or what Ariel said. ;D |
| 18:07.16 | funxion | anyone here have experience conneting PRI to meridian ? |
| 18:07.22 | cuilean | (Littered my box already. ;D) |
| 18:07.22 | Lee__ | Ariel_: that's what I though too but...it isn't honoring it cause it's packaged for Debian. I'm thinkning it might be something at compile time? |
| 18:07.31 | SpaceBass | one polycom 600 on its way to me |
| 18:07.33 | SpaceBass | cool |
| 18:07.55 | samuel_ | Hmmhesays: chan_h323!! |
| 18:08.00 | Hmmhesays | heh, someone got shot in the park yesterday |
| 18:08.00 | funxion | does anyone know how to make a te110p emulate the following cisco command "isdn channel-id invert extend-bit"? |
| 18:08.00 | Hmmhesays | our shootings just went up 100% over the last 2 years |
| 18:08.02 | SpaceBass | now to order the hatchi wireless |
| 18:08.03 | Lee__ | SpaceBass: be warned, it's a demoralizing experience to congfigure the thing |
| 18:08.13 | [hC] | ah. a reload doesnt pick up new modules. you have to actually 'stop' and re load. |
| 18:08.15 | SpaceBass | Lee__ ut-oh... it is bad? |
| 18:08.19 | Lee__ | terrible |
| 18:08.28 | Hmmhesays | samuel_: yes just allow g723 in h323.conf and sip.conf and you should be fine |
| 18:08.29 | funxion | Ariel have you dont it? |
| 18:08.33 | cluecon | meridian to asterisk...sounds fun. |
| 18:08.38 | SpaceBass | great... maybe the cisco was a better idea |
| 18:08.45 | SpaceBass | Lee__ what makes it rough? anything I need to know |
| 18:08.51 | Lee__ | yes, the 79xx is hard but tollerable. |
| 18:09.05 | samuel_ | Hmmhesays: thanks man..!! |
| 18:09.06 | Lee__ | SpaceBass: the phone has lots under the hood and it's all expressed as XML. |
| 18:09.06 | Hmmhesays | I wouldn't want to get hit over the head with one.... that's for sure |
| 18:09.09 | funxion | cluecon do you have any experience with it? |
| 18:09.16 | SpaceBass | Lee__ you saying the hatchi is hard or the polycom? |
| 18:09.17 | Hmmhesays | samuel_: np |
| 18:09.19 | Ariel_ | funxion, that is why I am hiding. |
| 18:09.22 | Lee__ | SpaceBass: Poly |
| 18:09.43 | SpaceBass | Lee__ so not just a matter of plugging it in, giving it an IP and assigning an extension? |
| 18:09.45 | funxion | well I am able to get inbound calls working fine using NI2 but outbound wont work ryte |
| 18:09.46 | dsfr | Lee__: the default path for record is /var/lib/asterisk/sounds/ . you can specify a path for the Record application though such as Record(/mnt/sounds/test.gsm). |
| 18:09.48 | Hmmhesays | xml isn't really that bad |
| 18:09.50 | cluecon | with connecting meridian to asterisk? no, but it shouldn't be too difficult depending on what you want to do. |
| 18:09.52 | samuel_ | Hmmhesays: sendme your mail from pvt, than i will send some pics of brazilian girls... |
| 18:09.54 | funxion | problem is on the meridian side |
| 18:10.03 | Hmmhesays | samuel_: i was just joking around |
| 18:10.09 | Hmmhesays | it's ok, I help you for free today |
| 18:10.10 | mjman | I can't find a reason why this wont work |
| 18:10.12 | funxion | when trying to connect using dms100 I get nothing |
| 18:10.23 | Ariel_ | funkycow, yes I think that manxpower has put something on the wiki about it. |
| 18:10.25 | samuel_ | samuel_: ok! |
| 18:10.30 | samuel_ | thanks.. |
| 18:10.32 | Hmmhesays | mjman: you got the context right? |
| 18:10.32 | funxion | but I need dms100 switchtype because thats the only way our merdian techy knows |
| 18:10.34 | Hmmhesays | np |
| 18:10.46 | mjman | Hmmhesays: yup, all the mailboxes are at default |
| 18:10.49 | Lee__ | dsfr: that's what I thought. The default path in the Debian package is /usr/share/asterisk/sounds, to conform to policy. |
| 18:10.51 | Ariel_ | funxion, , yes I think that manxpower has put something on the wiki about it. |
| 18:11.05 | cuilean | ::Sighs:: I've just been informed that they'd rather replace the Merlin Legend with a Merlin Magix... because then they don't have to buy new handsets. :p |
| 18:11.13 | Lee__ | SpaceBass: no, it's far more complicated than that. |
| 18:11.19 | funxion | I have multiple cisco's hooked up to the meridian but I needded the following command to make it work isdn channel-id invert extend-bit |
| 18:11.26 | SpaceBass | Lee__ might need to rethink my order then |
| 18:11.27 | *** join/#asterisk Gombi (Gombi@138-164.mc.royaume.com) |
| 18:11.28 | cuilean | Going to cost $15,000 to replace the box with a Magix, vs. $12,000 to Asterisk-it. :p |
| 18:11.30 | Gombi | Re all |
| 18:11.45 | opus__ | anyone here a unixOdbc person? |
| 18:11.52 | Gombi | Ok i Will post my question again :) |
| 18:12.01 | Lee__ | SpaceBass: the god news is you can do mass deployment a.la Cisco. So the week it takes to learn how to configure the thing only has to happen once. |
| 18:12.06 | Lee__ | s/god/good/ |
| 18:12.08 | funxion | anyone know how to invert the last octet of the channel id info onzaptel card? |
| 18:12.09 | Hmmhesays | mjman: remove the @context and try |
| 18:12.31 | opus__ | you can mass deploy polycom |
| 18:12.42 | Lee__ | yes, that's what I said. |
| 18:12.44 | mjman | Hmmhesays: already done, doesn't make a difference |
| 18:12.47 | Hmmhesays | but can you do it...... massively? |
| 18:12.52 | SpaceBass | Lee__ well its likely to be the only one I buy... so thats not much consolance |
| 18:13.03 | Lee__ | Hmmhesays: I guess so. how big is your massive? |
| 18:13.17 | cuilean | That's a rather personal question, Lee. ::SMirks:: |
| 18:13.21 | Hmmhesays | as big as my extremely funny jokes today ;D |
| 18:13.42 | Hmmhesays | er.. um |
| 18:13.46 | Lee__ | SpaceBass: if you are just looking for a single Ip phone, I'd recommend the Snom. The most hassel free |
| 18:14.01 | Ariel_ | SpaceBass, polycom's on the same lan are really easy to configure. You can deploy many phones by just setting up your dhcp and getting there mac address into files. |
| 18:14.07 | SpaceBass | Lee__ unfortunatly Im litited to what I can order through my only supplier from work |
| 18:14.09 | Hmmhesays | mjman: set verbose 4 |
| 18:15.00 | Hmmhesays | see what is being output when you call |
| 18:15.19 | mjman | Hmmhesays: Executing VoiceMailMain("SIP/7044-1f88", "7044@default") in new stack |
| 18:15.21 | Gombi | Did someone know Which Dialogic Card CPCI must i use to get an Asterisk System up??? Or someone tryed Asterisk with Dialogic PRI card to do (SIP)?? Or someone can give me an way to get a system with a lots of fiability?? we need the system to never go down" |
| 18:15.21 | SpaceBass | Ariel_ is the actual config done on the phone or through XML config files? |
| 18:15.25 | mjman | Hmmhesays: looks ok to me |
| 18:15.29 | Hmmhesays | then what |
| 18:15.32 | Ariel_ | SpaceBass, both |
| 18:15.37 | Ariel_ | or either |
| 18:15.41 | mjman | then im prompted for the mailbox number |
| 18:15.44 | mjman | and i hang up |
| 18:15.59 | Hmmhesays | what happens if you enter 7044 at that point? |
| 18:16.04 | heison | ~seen bkw_ |
| 18:16.05 | jbot | bkw_ is currently on #asterisk |
| 18:16.13 | funxion | mjman is your callerid the same as your extension? |
| 18:16.17 | mjman | then im prompted for password |
| 18:16.25 | mjman | funxion: yes it is |
| 18:16.34 | funxion | use VoicemailMain,s${CALLERIDNUM} |
| 18:16.48 | mjman | funxion: i'll try that |
| 18:16.52 | SpaceBass | any of the avaya phones worth looking out? |
| 18:16.58 | funxion | it works for me on extensions that have same callerid as extennum |
| 18:17.04 | SpaceBass | out=at |
| 18:17.16 | Hmmhesays | s just removes the password prompt |
| 18:17.17 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 18:17.19 | Ariel_ | SpaceBass, no |
| 18:17.22 | Hmmhesays | something else isn't right there |
| 18:17.39 | mjman | yep, still doesn't work |
| 18:17.44 | SpaceBass | nortel? |
| 18:17.46 | Hmmhesays | what version? |
| 18:18.00 | NewSole | Anyone know where I can get a SER server |
| 18:18.02 | Ariel_ | mjman, are you voicemail boxes in the default context? |
| 18:18.05 | NewSole | or where to find it |
| 18:18.15 | mjman | Ariel_ yes they are |
| 18:18.20 | brookshire | voip-info.org :) |
| 18:18.21 | opus__ | newsole google for ser sip |
| 18:18.23 | Hmmhesays | the question is, did you set the context in voicemail.conf |
| 18:18.24 | Ariel_ | add @default at the end |
| 18:18.31 | mjman | hmm |
| 18:18.32 | dsfr | Gombi: You should get a Digium TE410P then. |
| 18:18.33 | mjman | let me check |
| 18:18.38 | SpaceBass | Ariel_ |
| 18:18.41 | SpaceBass | oops |
| 18:18.42 | Hmmhesays | Ariel_: http://pastebin.ca/13736 |
| 18:18.45 | Hmmhesays | that's his |
| 18:18.59 | SpaceBass | Ariel_ : ease of use/config cisco 7912 or poly 600? |
| 18:19.09 | Ariel_ | in my view polycom |
| 18:19.25 | *** join/#asterisk Mr_Happy (~Mr_happy@fcirc-adsl.demon.co.uk) |
| 18:19.41 | SpaceBass | any other poly model i should be looking at? |
| 18:19.50 | Hmmhesays | one for my desk? |
| 18:20.00 | Ariel_ | The Cisco's can be a pain due to the firmware and which verison of sip you have. |
| 18:20.05 | Ariel_ | ip-500 |
| 18:20.10 | Ariel_ | or ip-501 |
| 18:20.22 | opus__ | whats the difference between the ip-501 and ip-500 |
| 18:20.31 | opus__ | is it the PoE type? |
| 18:20.37 | SpaceBass | looks like the 600 is my only option... so sticking with it and an hatchi ip-5000 |
| 18:20.57 | Mr_Happy | ip 501 is the current model has more flash memory |
| 18:21.02 | opus__ | why are you going to use the hatchi ip-5000? |
| 18:21.17 | Gombi | dsfr ya but this card is not Cpci... if the cards just bust off i can't change this card without shutting down the system?" |
| 18:21.19 | opus__ | VoCAT5 :) |
| 18:21.36 | cypromis | ip501 can be accesses via https |
| 18:21.41 | SpaceBass | opus__ for not being tied to a wire |
| 18:22.02 | brookshire | dsfr: time for experimentation ;) |
| 18:22.26 | *** join/#asterisk ChArLeS___ (~charles@64.35.168.55) |
| 18:22.32 | SpaceBass | opus__ not a fan of the ip-5000? |
| 18:22.57 | rristroph2 | is anyone here using the sipXphone softphone with * ? I haven't been able to configure it correctly |
| 18:23.02 | opus__ | lost one of my first clients to the hatchi system, don't know that much about it. |
| 18:23.15 | Ariel_ | SpaceBass, any of the wireless phones have problems with sound, connections, and how long the last on a charge. |
| 18:24.02 | ChArLeS___ | Ariel_: wireless sip phone ? |
| 18:24.09 | SpaceBass | opus__ this is for a home system... so cordless is a must... I tried my current analogue phone with a PA-168 (86?) ata and it didn't work. the ATA was junk and didn't recognize 1/2 the digits and sound was intolerable |
| 18:24.14 | Ariel_ | SpaceBass, the only one I have seen that works well most of the time is the expensive Cisco |
| 18:24.20 | ChArLeS___ | Ariel_: how much ? What about using a palm as a sipphone ? |
| 18:24.22 | Ariel_ | ChArLeS___, yes |
| 18:24.49 | opus__ | spacebass oh I thought you were talking about some other device, what is the url to the ip5000? |
| 18:24.50 | Hmmhesays | mjman it doesn't work for me either |
| 18:24.51 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 18:24.54 | SpaceBass | Ariel_ I've played with the zyxel and battery life was better than my Vtec and since I have 4 access points in my house coverage was better two... but it lacked call waiting and lacked WPA |
| 18:24.57 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 18:25.05 | Mr_Happy | anyone know if there is a way to log off all agents on a sip extension something like AgentCallBackLogoff (${CALLERIDNUM}) - if it existed |
| 18:25.08 | Mr_Happy | or limit 1 agent per extension where last agent to log in overrides any previous agents |
| 18:25.32 | Ariel_ | SpaceBass, I have a panasonic KX-TG2267 connected to my sipura 2000 and it works great gives me two lines and it's under 100 dollars |
| 18:25.36 | SpaceBass | opus__ http://www.voipsupply.com/product_info.php?products_id=354 |
| 18:25.44 | mjman | Haha, I figured it out. The phone I was using to test this out doesn't have a mailbox. |
| 18:25.49 | *** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca) |
| 18:25.52 | mjman | I gave it a mailbox, and it works fine |
| 18:25.53 | Hmmhesays | spoke to soon |
| 18:25.56 | SarahEmm | hihi |
| 18:26.00 | Hmmhesays | it works for me fine |
| 18:26.04 | SpaceBass | Ariel_ I am not sure what the deal with the pa-168 was... If I recall you may have even helped me configure it |
| 18:26.05 | Hmmhesays | hellow SarahEmm |
| 18:26.11 | ChArLeS___ | SpaceBass: just use a palm :) |
| 18:26.18 | mjman | thanks guys |
| 18:26.24 | Ariel_ | SpaceBass, I don't know the pa-168 |
| 18:26.24 | opus__ | oh whoah |
| 18:26.29 | SpaceBass | Ariel_ but it was junk... and everyone I called hung up b/c they cou;dnt understand me |
| 18:26.33 | SpaceBass | is there a pa-186? |
| 18:26.39 | SpaceBass | I'd have to go look at what it was |
| 18:26.39 | Ariel_ | For ATA's I use the sipura mainly. |
| 18:26.51 | Hmmhesays | cheap and reliable is never a bad thing |
| 18:27.05 | SpaceBass | ChArLeS___ while I might turn my old pocketpc into a sip phone... no way the wife would go for it as her house phone |
| 18:27.05 | SpaceBass | :) |
| 18:27.14 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:27.47 | opus__ | is there a way to do roaming with these wifi sip phones yet? |
| 18:28.00 | Ariel_ | SpaceBass, I have also my wife using the panasonic via the sipura.... I am able to get good sound all over the house and in the back yard. |
| 18:28.04 | SpaceBass | opus__ roaming, like taking it to starbucks? from what I hear it works in theroy... |
| 18:28.17 | SpaceBass | Ariel_ maybe I'[ll try a sipura |
| 18:28.28 | SpaceBass | but why not just go with the stright IP option? |
| 18:28.36 | opus__ | yes, like taking the call on somebody else's network but having your phone 'vpn' into your own |
| 18:28.50 | SpaceBass | part of the problem is I dropped my vtec and it is dieing and has to be replaced anyway... |
| 18:28.52 | opus__ | and still ring |
| 18:29.33 | SpaceBass | opus__ from what I understand... not a problem... none of the phones support VPN, and they won't support something like tmobile wifi, but if its an open AP, it will work |
| 18:29.39 | ChArLeS___ | Is there a way to set the "Calling party's category" for a ISDN channel on Asterisk ? |
| 18:29.45 | *** join/#asterisk jdg (~jdg@CA03F955.adsl.mana.pf) |
| 18:29.45 | *** join/#asterisk meppl (mephisto@p54AAD95D.dip.t-dialin.net) |
| 18:29.53 | tzanger | "calling party's category" ? |
| 18:30.03 | opus__ | hmmmm |
| 18:30.28 | Ariel_ | ChArLeS___, that's a new one for me as well. |
| 18:30.46 | tzanger | what do you set it to? Animal, Plant, Mineral? |
| 18:31.14 | Micc | So do I just comment out the fxsks=1 line in my zaptel.conf? |
| 18:31.18 | Micc | If I'm using ztdummy |
| 18:31.56 | Ariel_ | Argh they just gave me a bag of Ritz Chips.... Damm you just can't eat just one of these damm things. |
| 18:32.19 | funxion | if I have context=default in my zapata.conf why would asterisk be looking in [ZAP IN] on an inbound call? |
| 18:32.34 | zoa | because you need to restart asterisk |
| 18:32.38 | zoa | or you have two defaults |
| 18:32.55 | Ariel_ | stop asterisk, service zaptel restart then restart asterisk |
| 18:33.41 | tzanger | hahahaha |
| 18:34.46 | *** join/#asterisk gtigene (~chatzilla@70.89.216.41) |
| 18:34.48 | justin_time | Anyone here know *@home very well? |
| 18:34.54 | Ariel_ | the only thing I have in my default context is exten => _.,Congestion |
| 18:35.00 | tzanger | I don't think anyone uses @home very well |
| 18:35.17 | Ariel_ | justin_time, there is a group here that can help #amportal |
| 18:35.17 | blop | i use asterisk at home, but not the "*@home" one :D |
| 18:35.52 | justin_time | After troubleshooting for half a day, I found that it didn't install AGI.pm |
| 18:36.06 | justin_time | So, everytime it was trying to run a perl AGI, it would fail. |
| 18:36.07 | *** join/#asterisk klasstek (~nunyobiz@sta-206-168-218-206.rockynet.com) |
| 18:36.19 | justin_time | Is there an install script I missed somewhere? |
| 18:36.30 | gtigene | So, the command to get CVS HEAD is "cvs checkout -r HEAD asterisk" or just "cvs checkout asterisk"? |
| 18:36.34 | klasstek | matt nicholson /msg me please. |
| 18:36.36 | Ariel_ | justin_time, which build of the iso are you using? |
| 18:37.12 | Ariel_ | gtigene, no -r or head just cvs checkout asterisk |
| 18:37.20 | justin_time | Ariel_: don't recall, where can I find that? Not in /etc/issue or /etc/motd |
| 18:37.26 | gtigene | Ariel, thanks. |
| 18:37.45 | Ariel_ | justin_time, when you downloaded the cd. |
| 18:38.01 | justin_time | Ariel_: just found the iso - it's the 1.0 release |
| 18:38.14 | Ariel_ | get the 1.1 iso |
| 18:38.41 | justin_time | Ariel_: will it upgrade, or do I need to reconfig? |
| 18:38.52 | Ariel_ | reconfigure |
| 18:39.17 | Ariel_ | unless you just make a backup of mysql then restore the mysql part only. |
| 18:39.47 | justin_time | Ariel_: I didn't do too much to the base install, I'll just write it down and reformat |
| 18:40.05 | Ariel_ | justin_time,but there are major differences due to it now has amp 1.10.08 instead 07 |
| 18:40.08 | Hmmhesays | apt-cache search zlib? |
| 18:40.08 | brookshire | hey! i know that! |
| 18:40.24 | *** join/#asterisk Falle (falstaf@voip-forum.se) |
| 18:40.36 | Ariel_ | Hmmhesays, your fast |
| 18:40.56 | anthm | that would work if i was on a box with apt-cache => |
| 18:41.18 | anthm | perhaps you can run that for me and paste the reply? |
| 18:41.23 | Hmmhesays | there's a few zlib references |
| 18:41.30 | Hmmhesays | sure hold on |
| 18:41.32 | anthm | the devel |
| 18:41.38 | anthm | the headers and so |
| 18:41.40 | anthm | etc |
| 18:42.32 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:42.36 | Hmmhesays | zlib1g - compression library - runtime |
| 18:42.46 | *** join/#asterisk outofjungle (~outofjung@61.17.134.58) |
| 18:42.54 | Hmmhesays | zlib1g-dev - compression library - development |
| 18:43.09 | anthm | thx |
| 18:43.28 | Hmmhesays | zlibc is the c library |
| 18:43.40 | harryvv | I am looking for somone to test a voip connection for me. If anyone is interested msg me. |
| 18:44.10 | [TK]D-Fender | I just tried recompiling * with a small mod to "app_queue.c" which didn't seem to take. Any quick reason why? |
| 18:44.18 | Micc | Could someone call into my conference system to see if I still have a delay problem? |
| 18:44.37 | Hmmhesays | url? |
| 18:44.49 | Micc | (888) 206-4443 |
| 18:44.53 | Micc | room #1234 |
| 18:44.55 | Hmmhesays | ahh phone number |
| 18:45.48 | Hmmhesays | woah, yeah I can hear that |
| 18:45.53 | Meaty | :) |
| 18:46.37 | SarahEmm | [TK]D-Fender: err... 'didn't seem to take'? are you sure it compiled right? what fix? |
| 18:47.03 | Hmmhesays | yeah you sounded fine, no delay but I could hear myself faintly delayed |
| 18:47.06 | [TK]D-Fender | I changed "app_queue.c" as follows at line 1324 Per CVS (about a week ago's) |
| 18:47.07 | [TK]D-Fender | <PROTECTED> |
| 18:47.08 | Nugget | it sounded ok to me |
| 18:47.08 | [TK]D-Fender | <PROTECTED> |
| 18:47.08 | [TK]D-Fender | <PROTECTED> |
| 18:47.08 | [TK]D-Fender | <PROTECTED> |
| 18:47.29 | Hmmhesays | it was delayed on his end |
| 18:47.30 | Nugget | but I still suggest moving to linux. I realize that it's distasteful, but if you need ztdummy it's the sane approach |
| 18:47.31 | brookshire | hi.. pastebin thanks! |
| 18:47.53 | jeffik | harryvv: what do you want to test? |
| 18:48.06 | Nugget | I gave up trying to get meetme working on my freebsd asterisk server |
| 18:48.11 | [TK]D-Fender | I added a 2nd verbose just to see if I could (being a hack programmer) trying too see if I could isolate the agent so as to add a "queue_log" entry showing when agents were slacking off. |
| 18:49.01 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 18:49.03 | ramtha | hi |
| 18:49.15 | Hmmhesays | I could use a suggest for a decent sip video phone |
| 18:49.25 | [TK]D-Fender | I see the 1st verbose in the console, but not the 2nd |
| 18:50.37 | ramtha | i get a nummber block from my provider. thos block is terminated for incoming to my zap e1 card. in zapata.conf i have context xyz. in extensions.con i have exten => _X.,1,Dial(SIP/${EXTEN}@x.x.x.x) |
| 18:50.42 | [TK]D-Fender | after my mod I made sure the file has the sa,e permissions and everything and then did |
| 18:50.50 | [TK]D-Fender | "make clean ; make install" |
| 18:50.53 | ramtha | if i call one number it comes into the zap channel and the i get |
| 18:51.16 | Hmmhesays | anyone? |
| 18:51.18 | ramtha | Jun 8 17:48:42 WARNING[9272]: chan_sip.c:1829 sip_write: Asked to transmit frame type 8, while native formats is 0 (read/write = 0/0) |
| 18:51.23 | ramtha | what is the problem? |
| 18:52.35 | Micc | How is the cvs head looking today? |
| 18:52.57 | chorix | Hello everybody. How much per hour and how many hours could cost an asterisk box full install and setup with 100 ip extens? |
| 18:54.04 | *** join/#asterisk mkrufky (~mike@68.160.103.76) |
| 18:54.07 | [TK]D-Fender | JUst looking at other string usage could it be that the %d should have been a %s ? |
| 18:54.47 | SarahEmm | chorix: that varies very widely dpeending who you use. |
| 18:54.56 | [TK]D-Fender | I clearly don't know C worth shit, but did lots in Turbo Pascal & PHP |
| 18:54.59 | Hmmhesays | I second that |
| 18:55.05 | Hmmhesays | chorix |
| 18:55.11 | [TK]D-Fender | Any guesses for my problem SarahEmm? |
| 18:55.24 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 18:55.38 | shmaltz | do I need a dual processer for a single t1? |
| 18:55.39 | brimstone | chorix: look into digium's config packages |
| 18:55.47 | zoa | ijdno you dont |
| 18:55.48 | Hmmhesays | shmaltz: no |
| 18:55.50 | zoa | no you dont |
| 18:56.08 | Mr_Happy | <PROTECTED> |
| 18:56.42 | brimstone | chorix: http://digium.com/index.php?menu=service_category&category=installation maybe |
| 18:56.55 | chorix | I don?t want to config myself, I am looking for a digium certified worker who doit for me, but I don?t know how much cost. |
| 18:57.01 | shmaltz | thanks guys |
| 18:57.15 | [TK]D-Fender | hiya shmaltz |
| 18:57.19 | shmaltz | which processer should I use? |
| 18:57.20 | brimstone | chorix: you can pay digium and they'll assign a tech to config it for you |
| 18:57.24 | shmaltz | hi D-Fender |
| 18:57.38 | chorix | Thanks brimstone |
| 18:57.55 | [TK]D-Fender | shmaltz : Got the queue's working a lot better now and am ironing out some "C" issues. |
| 18:57.58 | dsfr | chorix: it's pretty cheap. |
| 18:58.03 | chorix | Nobody here who can do the work for me? |
| 18:58.04 | shmaltz | nice |
| 18:58.11 | Hmmhesays | chorix: take your pick |
| 18:58.28 | Hmmhesays | if you want to pay is to put 100 peers in your sip.conf that's gonna cost you |
| 18:58.33 | Hmmhesays | *pay us |
| 18:58.41 | shmaltz | anybody here have any recomendations for a phone that works good behind nat similar to Polycom IP 500? |
| 19:00.15 | chorix | Do you know a server host where I can put my * Box and contract a broadband connection in Miami? |
| 19:00.32 | Hmmhesays | like a colo? |
| 19:00.40 | funxion | chorix gisnet |
| 19:00.45 | funxion | look it up |
| 19:00.49 | funxion | cheap colo |
| 19:00.52 | funxion | Im in miami too |
| 19:01.04 | Hmmhesays | ok food time |
| 19:01.47 | chorix | funxion, what is gisnet? |
| 19:01.49 | funxion | anyone know why an inbound call from a PRI would hang up right after passing call info? |
| 19:01.54 | funxion | its a colo spot |
| 19:03.00 | *** join/#asterisk xcore (rmaraschin@200.180.23.135) |
| 19:03.13 | chorix | As you can see I am not good in english and tech. Can you explain me what a colo spot and gisnet are? |
| 19:03.18 | bkw_ | heison, yes? |
| 19:03.52 | *** join/#asterisk ctooley (~ctooley@pc51.utati.net) |
| 19:04.12 | funxion | chorix a colo spot is a secure facility where you would house your server or rent a server or even just space on a server they would provide power and bandwidth |
| 19:04.27 | funxion | bandwidth is usually cheaper and comes a lot faster when in a colo |
| 19:04.28 | ctooley | I can't figure out why my IAXy is suddenly getting INVAL back on an AUTHREQ even though it's using the same user and secret as are defined in the iax.conf |
| 19:05.07 | Micc | what is chan_modem for? I don't think I need it, can I not load it? |
| 19:05.15 | chorix | funxion And do you know any colo near Fort Lauderdale? |
| 19:05.16 | *** join/#asterisk WSN (~Master@50.198-201-80.adsl.skynet.be) |
| 19:05.27 | WSN | hi |
| 19:05.33 | funxion | publicom |
| 19:05.45 | funxion | there is a ton of them off the golden glades interchange |
| 19:05.52 | WSN | have a question |
| 19:05.52 | funxion | look em up |
| 19:06.10 | funxion | WSN ask your question if somebody can help you they will answer |
| 19:06.40 | chorix | Thanks funxion |
| 19:06.53 | funxion | chorix np |
| 19:07.15 | perlmonky | trying to specify a playlist for moh... |
| 19:07.23 | WSN | anybody any idea what causes a growing delay on outgoing sound with a idsn modem using isdn4linux? |
| 19:07.31 | perlmonky | default => mp3:/var/lib/asterisk/mohmp3-empty,-@ /var/lib/asterisk/mohmp3/playlist |
| 19:07.34 | perlmonky | should work right? |
| 19:07.57 | ctooley | http://pastebin.com/297285 |
| 19:08.06 | ramtha | how can i add a 0 at the bginning of every nummber? |
| 19:08.21 | ramtha | for incomming calls |
| 19:08.43 | funxion | ramtha voip or pstn calls? |
| 19:08.52 | ramtha | voip |
| 19:09.08 | funxion | hmm |
| 19:09.30 | funxion | why? |
| 19:09.44 | ramtha | the calls com in without a leading zero |
| 19:09.52 | ramtha | example: 1634255xxx |
| 19:09.59 | ramtha | it must be 01634255xxx |
| 19:10.09 | funxion | for what reason? |
| 19:10.09 | ramtha | because of our mediation |
| 19:10.10 | shido | add it to the exten |
| 19:10.11 | *** join/#asterisk Excellence (Excellence@64.215.120.254) |
| 19:10.13 | funxion | to be passed on ? |
| 19:10.24 | ramtha | in cdr i must have 0163xxxx |
| 19:10.28 | ramtha | not 163xxx |
| 19:10.42 | Meaty | funxion ! Cool name ! |
| 19:10.46 | funxion | what shido said |
| 19:10.46 | *** join/#asterisk FuriousGeorge (~ads@pool-138-89-15-189.mad.east.verizon.net) |
| 19:10.58 | ramtha | hm? |
| 19:11.01 | Excellence | Who else is going to be at madrid astricon? |
| 19:11.09 | ramtha | how can i add? |
| 19:11.19 | ramtha | with :1 i can cut the first |
| 19:11.22 | funxion | 0${EXTEN} |
| 19:11.32 | ramtha | ic thx a lot |
| 19:11.35 | funxion | np |
| 19:12.11 | funxion | shido do you have any idea why an inbound pri call would hangup just after is passes call info? |
| 19:13.19 | [hC] | huh. asterisk-stat seems pretty cool |
| 19:13.41 | FuriousGeorge | i cant seem to find the corrext syntax to dial zap/1 & sip/user at the same time when i get an icoming call. anyone wanna drop a hint |
| 19:13.42 | tzanger | asterisk-stat? |
| 19:13.58 | [hC] | tzaweb based CDR viewing thingy. |
| 19:14.01 | *** join/#asterisk Iceman12C (~Cliff@pool-71-96-177-76.dfw.dsl-w.verizon.net) |
| 19:14.03 | [hC] | Draws graphs for daily usage, and such |
| 19:14.08 | tzanger | ahh |
| 19:14.15 | ramtha | aeh, i can add the zero for the destination nummber....but i think not for the source nummber...right? |
| 19:14.25 | Iceman12C | Can someone recommend a good web-based GUI frontend for asterisk? |
| 19:14.47 | ramtha | iceman12c: for what? cdr? configuratio? |
| 19:15.05 | Iceman12C | mmm.. configuration |
| 19:15.19 | ramtha | Iceman12c: di you use db ore flat files? |
| 19:15.21 | shido | funxion, sounds like you have problems :) |
| 19:15.36 | funxion | yep |
| 19:16.00 | Iceman12C | ramtha: i'm using flatfiles right now, but I'm willing to switch |
| 19:16.09 | funxion | everything works fine with NI2 but not with dms100 |
| 19:16.34 | FuriousGeorge | anyone know how i can make both a zap and sip chanel ring upon an incoming call? currently the zap channel rings for 15 seconds then gets forewarded. i would like the sip channel to ring the whole time |
| 19:16.45 | ramtha | Iceman12c: ok switch first, than think about a web interface. in my opinion, if you have difficult setups, there is now realy good gui |
| 19:17.18 | Iceman12C | ramtha: wait a second.. what does switching involve? |
| 19:17.20 | ramtha | Furios: you can setup 2 endpints in one extension |
| 19:17.28 | ramtha | i do not now the syntax |
| 19:17.30 | ramtha | ask google |
| 19:18.08 | ramtha | Iceman12c: some guis only can do with files, some only with mysql, some only mysql/postgres |
| 19:18.30 | ramtha | get your basic config working and then choose your interface |
| 19:18.39 | ramtha | we wrote aour own interface |
| 19:18.40 | Iceman12C | is there a good GUI that you recommend i check out? |
| 19:18.42 | ramtha | for our needs |
| 19:19.01 | ramtha | http://www.voip-info.org/wiki-Asterisk+GUI |
| 19:19.04 | ramtha | choose one |
| 19:19.13 | Iceman12C | ok thanks |
| 19:19.33 | FuriousGeorge | ramtha are you talking about groups? b/c they are only across one type of channel. i dont knopw what u mean by endpints or endpoints |
| 19:19.55 | WSN | anybody any idea what causes a growing delay on outgoing sound with a idsn modem using isdn4linux? |
| 19:20.34 | FuriousGeorge | i thought i saw it done with the dial command exten => s,1,dial(zap/1)&dial(sip/user) something like that |
| 19:20.51 | ramtha | yeh, thats what i saw too :) |
| 19:22.03 | iCEBrkr | Is there ANY documentation on EAGI's and how to do something with the audio you can capture, or even how to capture the audio on that line??? |
| 19:22.09 | FuriousGeorge | i wish i could remember where i saw it, im looking all over voip-info right now |
| 19:22.20 | dsfr | FuriousGeorge: exten => s,1,Dial(Zap/1&Sip/user) |
| 19:22.27 | *** join/#asterisk laksla (mglucksman@201.243.103.234) |
| 19:22.46 | FuriousGeorge | dsfr: thanks gonna try |
| 19:24.15 | dsfr | FuriousGeorge: you can do the same method if you want to copy to multiple voicemail boxes using the voicemail application. |
| 19:24.37 | harryvv | Has anyone had experaince with stun before? |
| 19:24.53 | dsfr | FuriousGeorge: like "Voicemail(u1000&1001&1002)" |
| 19:25.12 | *** join/#asterisk sm7xab (~sm7xab@h240n1c1o1095.bredband.skanova.com) |
| 19:25.13 | harryvv | If anyone has had experaince with stun message me. |
| 19:26.43 | *** join/#asterisk sm7syx (~sm7syx@212-162-184-20.skbbip.com) |
| 19:27.40 | FuriousGeorge | dsfr: thanks, i knew i was close, i swear i was looking all voer to find that. take care, gotta run |
| 19:27.42 | *** join/#asterisk asdfblah (~UNIX@pcp04541662pcs.brmngh01.mi.comcast.net) |
| 19:27.54 | asdfblah | anyone here to the asterisk flash operator panel? |
| 19:28.19 | *** part/#asterisk Iceman12C (~Cliff@pool-71-96-177-76.dfw.dsl-w.verizon.net) |
| 19:28.19 | *** join/#asterisk ctooley (~ctooley@pc51.utati.net) |
| 19:28.21 | asdfblah | usa the* |
| 19:28.48 | NewSole | What bandwidth would be needed for 230 g729 calls going at same time |
| 19:28.54 | ctooley | Jun 8 14:26:49 NOTICE[5594]: chan_iax2.c:7014 socket_read: Peer 'asterlink' is now REACHABLE! Time: 61 |
| 19:28.54 | ctooley | cuervo*CLI> " |
| 19:29.07 | ctooley | uhg that didn't work. |
| 19:29.14 | Juggie | NewSole, break out your calculator |
| 19:29.26 | Juggie | 230x8bit(1kb)+overhead |
| 19:29.52 | *** join/#asterisk jjhall (~chatzilla@24-119-114-94.cpe.cableone.net) |
| 19:30.02 | Juggie | at the very least it is 230kb/s but... there is going to be overhead.. so that number could likely double, unless your calls are going over iax |
| 19:30.21 | zoa | there is an online calculator |
| 19:31.06 | sm7xab | Hi there! I've got a small question. Is it possible to use a normal phone modem for incomin analog calls with Asterisk? |
| 19:31.17 | zoa | http://asteriskguru.com/bandwidth_calculator.php |
| 19:31.59 | Gombi | im just posting my question differently. |
| 19:32.41 | Gombi | Is there a way to get asterisk always up (like clusters) or redundant PRI SIP cards. |
| 19:32.47 | laksla | Anyone knows why a call is moved from one group to another on a same span, giving a bad channel error? |
| 19:33.48 | harryvv | need to clearify something. when a sip client is registering on my asterisk box thay are using rtp ports 10000-20000? |
| 19:34.15 | ramtha | rtp.conf? |
| 19:34.17 | jjhall | I have calls that keep coming in from a marketer, with caller ID set to 000-000-0000. I tried setting up a matching line, but it won't catch it. Here is a pastebin with console log, then the extensions.conf snippet. Can somebody please see if I coded something incorrectly? http://pastebin.com/297310 |
| 19:34.29 | harryvv | ramtha: thats already stet from 10k-20k |
| 19:34.42 | harryvv | the firewall is set from 10k-20k |
| 19:35.36 | ramtha | what are you talking about? rtp ports or kb/s? |
| 19:36.06 | harryvv | rtp |
| 19:36.21 | pashah | cheers |
| 19:36.36 | *** part/#asterisk cuilean (~cuilean@ool-44c16f4c.dyn.optonline.net) |
| 19:36.38 | harryvv | Jun 8 12:35:10 NOTICE[3736]: chan_sip.c:8448 handle_request: Registration from '153 <sip:mike@24.81.64.205>' failed for '69.158.9.87' is failing and we do have the username/pass set right. |
| 19:36.57 | JerJer | what realm? |
| 19:37.12 | harryvv | the end user is trying xlite |
| 19:37.20 | ramtha | harryvv: what you then meen with 10k-20k? |
| 19:37.21 | JerJer | put asterisk in the realm |
| 19:37.27 | harryvv | going though there dmz and then though my firewall. |
| 19:37.31 | harryvv | ramtha: yes |
| 19:37.39 | ramtha | hae? |
| 19:37.41 | harryvv | rtp ports 10000-20000 |
| 19:37.43 | ramtha | yes? |
| 19:37.46 | [TK]D-Fender | Anyone awake here who has done "C" coding for * who might have a moment to spare to assist me with a small problem? |
| 19:37.57 | ramtha | i don´t understand your words, sorry :) |
| 19:38.13 | JerJer | you mean C* ? |
| 19:38.26 | harryvv | sorry jerjer what do you mean by realm? |
| 19:38.33 | JerJer | Authentication Realm |
| 19:38.37 | [TK]D-Fender | As in *'s source code. |
| 19:39.06 | [TK]D-Fender | I'm working on "app_queue" and couldn't get one simple thing to work. |
| 19:39.39 | harryvv | on my asterisk box? I dont ever recall configuring that or seeing it. This is a simple asterisk behind this firewall with 5060/10k-20k open and pointed to this server. end user is using xlite via his dmz. |
| 19:39.52 | JerJer | xlite |
| 19:40.01 | Lee__ | does Asterisk have a config option for changing the default location it looks for sound files? |
| 19:40.02 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 19:40.16 | Hmmhesays | asterisk.conf |
| 19:40.30 | MattH | Hi, using the AGI how can I get the caller-id of the person dialing the extension.. just the number not the name.. ie not Joe Smith but rather 221 |
| 19:40.32 | Lee__ | the name of the option, not the file where it's configurable. |
| 19:40.41 | Lee__ | astsounddir? |
| 19:41.13 | Hmmhesays | MattH: strip off the name with your script |
| 19:41.31 | drumkilla | no need |
| 19:41.36 | drumkilla | it's in the var CALLERIDNUM |
| 19:41.38 | harryvv | okay the xlite has my asterisk realm ip setup |
| 19:41.45 | Hmmhesays | good call drumkilla |
| 19:41.51 | brookshire | matt what? |
| 19:41.55 | Lee__ | guess not... |
| 19:41.58 | Lee__ | thanks. |
| 19:42.18 | Hmmhesays | it's in /etc/asterisk.conf is it not? |
| 19:42.18 | MattH | drumkilla: and how do you access the variable CALLERIDNUM from the AGI script? is it passed to the AGI? or do I have to get it? |
| 19:42.37 | Hmmhesays | MattH: there are examples on the wiki |
| 19:42.52 | [TK]D-Fender | drumkilla : any chance for getting a minute or so's help on my coding "challenge"? |
| 19:43.02 | drumkilla | MattH: depends on which AGI bindings you are using :) |
| 19:43.06 | *** join/#asterisk robin_sz (~robin@adsl.redpoint.org.uk) |
| 19:43.11 | robin_sz | meep? |
| 19:43.14 | dsfr | Lee__: you have to modify defaults.h in your Asterisk source directory. Specify that default sounds directory there. You'll have to recompile Asterisk afterwards for the changes to take effect. |
| 19:43.18 | Lee__ | Hmmhesays: the name of the option, not the file where it's configured. I know about asterisk.conf already. |
| 19:43.34 | MattH | Hmmhesays: I'm aware of how to get the calleridnum IN asterisk.. but not sure how to pass it to the AGI script |
| 19:43.42 | Lee__ | dsfr: excellent. I just wanted to confirm that before I go complaining to the Debian package maintainer :) |
| 19:44.04 | Lee__ | too bad it's already in sarge that way. |
| 19:44.12 | drumkilla | [TK]D-Fender: how quick is it? :) |
| 19:44.25 | MattH | ahh |
| 19:44.28 | MattH | hrmm solutions found I believe |
| 19:44.34 | MattH | drumkilla: thanks for the stearing in the right direction |
| 19:44.48 | drumkilla | MattH: you're welcome |
| 19:45.01 | drumkilla | it's going to be however you do a "getvar" in whatever AGI lib you're using |
| 19:45.02 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
| 19:45.17 | MattH | drumkilla: right... I just was having a hangup on getting it passed out.. but found what I needed |
| 19:45.30 | drumkilla | cool |
| 19:45.33 | drumkilla | next! :) |
| 19:46.41 | *** join/#asterisk outofjungle (~outofjung@61.17.134.156) |
| 19:46.58 | *** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || 1.0.8 RC bug #4424 || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 |
| 19:47.09 | *** join/#asterisk gatty (~agatward@beauchief.plus.com) |
| 19:47.58 | [TK]D-Fender | drumkilla : http://pastebin.ca/13749 - I did the following mod, did "make clean ; make install" everyting seems to finish ok, but no message. I get the 1st, but not my addidtion. |
| 19:48.51 | drumkilla | ummm ... code looks fine |
| 19:48.54 | drumkilla | rm /usr/lib/asterisk/modules/* |
| 19:49.01 | drumkilla | and do a make install again ... |
| 19:49.28 | *** join/#asterisk critch (~critch@pcp08699084pcs.500ash01.tn.comcast.net) |
| 19:49.49 | critch | Anyone familiar with telecom in Okinawa? |
| 19:50.24 | [TK]D-Fender | drumkilla : Will try now. I know little about C, Linux, or * coding and am running on instinct. |
| 19:50.31 | drumkilla | haha |
| 19:51.15 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
| 19:51.48 | [TK]D-Fender | ~>~ |
| 19:51.51 | drumkilla | oh, and make sure you actually restart asterisk |
| 19:51.56 | drumkilla | :) |
| 19:52.02 | drumkilla | or unload app_queue.so |
| 19:52.04 | drumkilla | load app_queu.so |
| 19:52.26 | Meaty | I have a weird trouble : I do 1800 calls by my PRI on my server "1" (because long distance is blocked for the moment). A server "2" is connected to server "1" via IAX. And I can do this call by a sip phone and iaxy both registered on server "2". On the CLI of the server "1" i see than both ways call sucessfully the dial command to the 1800 number. With the iaxy, the call is sucesfully, but with the sip phon, i see on the sip : Channel 0/1, span 1 got |
| 19:52.28 | [TK]D-Fender | I downed it and verified the process as gone and am recompiling the whole thing |
| 19:52.43 | drumkilla | cool, let me know how it goes ....... |
| 19:52.45 | Meaty | And the PRIi Hangup |
| 19:52.49 | [TK]D-Fender | Will do in about 2 mins |
| 19:53.36 | *** join/#asterisk WSN (~Master@217.148.166.100) |
| 19:53.38 | WSN | anybody any idea what causes a growing delay on outgoing sound with a idsn modem using isdn4linux? |
| 19:53.51 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 19:56.36 | Gombi | Is there a way to get asterisk always up (like clusters) or redundant PRI SIP cards. |
| 19:56.40 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 19:56.54 | SarahEmm | PRI SIP cards? |
| 19:57.01 | brookshire | PRI SIP? |
| 19:57.21 | *** join/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca) |
| 19:57.57 | [TK]D-Fender | drumkilla : Still nothing :( |
| 19:58.11 | *** join/#asterisk yaaar_work (~chatzilla@lifebook.tranquility.net) |
| 19:58.16 | yaaar_work | word |
| 19:58.24 | ramtha | excel |
| 19:58.31 | WSN | outlook |
| 19:58.34 | ramtha | powerpoint |
| 19:58.34 | yaaar_work | anybody familiar with the grandstream gxp2000? |
| 19:58.42 | yaaar_work | access |
| 19:58.52 | WSN | project :p |
| 19:58.53 | drumkilla | [TK]D-Fender: want me to log in? |
| 19:58.59 | ramtha | xp |
| 19:59.09 | *** part/#asterisk critch (~critch@pcp08699084pcs.500ash01.tn.comcast.net) |
| 19:59.13 | WSN | xp isn't a part of office :p |
| 19:59.13 | [TK]D-Fender | Can't its internal LAN at work :/ dang it |
| 19:59.20 | drumkilla | hrm, ok |
| 19:59.22 | ramtha | right :) |
| 19:59.23 | drumkilla | well. |
| 19:59.26 | *** join/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 19:59.34 | drumkilla | I don't know what to tell you :) |
| 19:59.43 | yaaar_work | i've got a real new (and still somewhat cheesy) asterisk box here, and softphones (x-lite SIP, iaxcomm IAX) are working ok. but i just got this grandstream phone, and i thought i had things setup, but when i dial, i get these errors: |
| 19:59.47 | [TK]D-Fender | Basically just added 1 line to "app_queue.c" and {}ed it. |
| 20:00.02 | drumkilla | yeah, it looks fine |
| 20:00.10 | yaaar_work | "No translator path exists for channel type IAX2 (native 63490) to 256 |
| 20:00.12 | yaaar_work | " |
| 20:00.13 | drumkilla | and you ARE restarting asterisk? |
| 20:00.38 | yaaar_work | "Unable to created channel of type 'IAX2'" |
| 20:00.42 | [TK]D-Fender | I though the %d may be a wrong type or something since I saw that field marked as %s elsewhere (implying string?) |
| 20:00.42 | drumkilla | yaaar_work: you probably have g729 turned on |
| 20:00.48 | yaaar_work | yeah |
| 20:00.54 | yaaar_work | drumkilla: what should i use instead? |
| 20:01.03 | drumkilla | anything else is fine |
| 20:01.11 | drumkilla | [TK]D-Fender: yes, %s is for a string |
| 20:01.16 | drumkilla | which is what you need for your line |
| 20:01.16 | [TK]D-Fender | drumkilla : I did "stop now", "ps -A" and then recompiled "make clean ; make install" |
| 20:01.33 | [TK]D-Fender | drumkilla : I'll try that, brb |
| 20:01.42 | drumkilla | that wouldn't cause the line not to come up |
| 20:01.55 | drumkilla | I don't know what you're missing |
| 20:02.19 | drumkilla | did you save the file with your change in it? ha |
| 20:02.26 | *** join/#asterisk file[desk] (~jcolp@mctn1-1420.nb.aliant.net) |
| 20:02.30 | drumkilla | file[desk]: !!! |
| 20:02.50 | file[desk] | drumkilla: !!! |
| 20:03.48 | Cresl1n | hey peeps :-) |
| 20:03.58 | WSN | My question again : anybody any idea what causes a growing delay on outgoing sound with a idsn modem using isdn4linux? |
| 20:04.09 | drumkilla | Cresl1n: !!!!!!!!!!!!!!!! |
| 20:04.33 | Cresl1n | drumkilla: !!!!!! |
| 20:04.33 | Cresl1n | :-) |
| 20:04.37 | drumkilla | everyone, meet Cresl1n, my hero! |
| 20:04.37 | file[desk] | Cresl1n! |
| 20:04.41 | Cresl1n | file[desk]: !!!!! |
| 20:04.46 | file[desk] | Cresl1n is l33t |
| 20:04.48 | Cresl1n | aww :-) |
| 20:04.51 | drumkilla | Cresl1n is like ... uber 1337 |
| 20:05.22 | Cresl1n | oooooh! |
| 20:05.33 | Cresl1n | thanks ;-) |
| 20:05.34 | *** join/#asterisk maveric (maveric@ip68-3-248-136.ph.ph.cox.net) |
| 20:05.36 | file[desk] | I'm ordering my... Powerbook! |
| 20:05.42 | drumkilla | file[desk]: no!!! |
| 20:05.46 | drumkilla | Cresl1n: hahaha |
| 20:05.48 | file[desk] | yesssssssss |
| 20:05.52 | Cresl1n | it can only do one thread though |
| 20:05.53 | drumkilla | that's what we need, a red bull funnel |
| 20:06.04 | drumkilla | file[desk]: yeah, Macs can't handle threads. |
| 20:06.07 | Cresl1n | heh |
| 20:06.14 | drumkilla | they can only do 1 |
| 20:06.26 | MikeJ[Laptop] | I'm hungry |
| 20:06.34 | drumkilla | you would be! |
| 20:06.47 | file[desk] | hungry... FOR DRUMKILLA! |
| 20:06.49 | harryvv | no two way audio between me and the end user. all my ip ports are open. |
| 20:07.02 | Cresl1n | I had an extra |
| 20:07.11 | drumkilla | harryvv: "SIP happens." |
| 20:07.14 | MikeJ[Laptop] | no.. hungry for food |
| 20:07.18 | Cresl1n | file[desk]: how'd prom go? |
| 20:07.25 | file[desk] | Cresl1n: I'm still in classes |
| 20:07.36 | Cresl1n | file[desk]: ???? |
| 20:07.45 | file[desk] | next week is exams |
| 20:07.45 | harryvv | drumkilla: :) yea heheh i knows it a royal pain. I have forwared all known sip ports to my server. |
| 20:07.50 | file[desk] | then week after is prom/graduation/etc |
| 20:07.51 | Cresl1n | after that is prom? |
| 20:07.52 | Cresl1n | ok |
| 20:07.54 | Cresl1n | nice |
| 20:08.06 | drumkilla | file[desk]: take pictures! |
| 20:08.13 | yaaar_work | is alaw or ulaw better to use between my grandstream phone and *? |
| 20:08.14 | *** join/#asterisk ast_freak (~ast_freak@hades-out.universalsystems.net) |
| 20:08.17 | drumkilla | file at prom, how cute! |
| 20:08.18 | file[desk] | oooooh aaaaaah |
| 20:08.21 | Corydon-w | file[desk]: who are you taking to prom? ;-) |
| 20:08.24 | file[desk] | we're thinking of riding in on a bicycle built for two |
| 20:08.36 | drumkilla | hahahaha |
| 20:08.38 | MikeJ[Laptop] | yaaar_work, it's best to use the same as whatever it will be talking too |
| 20:08.43 | xcore | i still trying resolv the nat problem, i add the nat = yes on [general] but the problem persist |
| 20:08.52 | file[desk] | nobody has ever done it |
| 20:08.53 | xcore | any idea... |
| 20:08.53 | drumkilla | file[desk]: who's your date?! |
| 20:08.55 | [TK]D-Fender | drumkilla : yes I confirmed the file had save with the change, RM'd the modules, and still nothing :/ |
| 20:08.56 | Cresl1n | xcore: nat=never? |
| 20:09.03 | file[desk] | drumkilla: a girl named Jana |
| 20:09.17 | xcore | Cresl1n hmmm, let's try |
| 20:09.21 | gatty | I knew a girl named Jana once |
| 20:09.28 | drumkilla | [TK]D-Fender: I have no clue what you have done :) |
| 20:09.31 | MikeJ[Laptop] | file[desk], does she know how much of a geek you are? |
| 20:09.37 | drumkilla | someone fix [TK]D-Fender 's problem! |
| 20:09.53 | file[desk] | MikeJ[Laptop]: yesssssssss |
| 20:09.58 | MikeJ[Laptop] | wow... nice moves |
| 20:09.59 | drumkilla | [TK]D-Fender: get it on a public IP and I'll fix it :) |
| 20:10.24 | MikeJ[Laptop] | or ssh tunnel to it |
| 20:10.30 | [TK]D-Fender | I added 1 lousy line to app_queue.c @ line 1325 +/- for another Verbose and nothing happens... |
| 20:11.19 | MikeJ[Laptop] | nothing happens as in app_quese no longer works at all, or nothing happens as in you don't see verbose? |
| 20:11.24 | [TK]D-Fender | in /usr/src/asterisk I go through the usual make (with RM'ing the modules as you described) and install and no impact. |
| 20:11.25 | xcore | Cresl1n on the general or in the exten |
| 20:11.45 | [TK]D-Fender | Don't see the verbose. I see the one that should appear before my add-on, but not mine. |
| 20:12.53 | gtigene | What version of the Linux kernel are folks using now? |
| 20:12.59 | MikeJ[Laptop] | cvs diff your app_queue and pastebin it |
| 20:13.06 | MikeJ[Laptop] | cvs diff -u |
| 20:13.25 | gatty | gtigene: my VoIP box is on 2.6.11-gentoo-r9 |
| 20:13.32 | dsfr | gtigene: 2.6 |
| 20:14.06 | Meaty | Anyone can help me with it !! http://pastebin.com/297328 |
| 20:14.25 | *** part/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca) |
| 20:14.53 | *** join/#asterisk rogercharlie (~charliero@c-24-5-73-64.hsd1.ca.comcast.net) |
| 20:15.22 | [TK]D-Fender | MikeJ[Laptop] : I would, but I don't know CVS or any programming/code related tools like it at all and just pastebin'd the mod |
| 20:15.30 | [TK]D-Fender | http://pastebin.ca/13749 |
| 20:15.38 | [TK]D-Fender | Around the line# I mentioned |
| 20:16.48 | [TK]D-Fender | :/ |
| 20:17.17 | Hmmhesays | on the road again, i just can't wait to get on the road again |
| 20:17.20 | rristroph2 | I have an IP hardware phone on one of my extensions that seems to get an echo. Is there a way in asterisk to set some echo cancelation parameters on a per-extension basis ? |
| 20:17.24 | opus__ | hmmm |
| 20:17.35 | opus__ | Anyone here use realtime odbc? |
| 20:17.37 | Hmmhesays | i'm coming up really short googling ip video phones |
| 20:17.38 | *** join/#asterisk jackx (none@dslam7-21-59-81.dyndsl.versatel.nl) |
| 20:17.45 | rogercharlie | has IAX in head been shakey for the last month? |
| 20:18.03 | MikeJ[Laptop] | [TK]D-Fender, you just modified that one file? |
| 20:18.10 | MikeJ[Laptop] | then redid make? |
| 20:18.59 | jackx | When using the AGI. the message "If no channel name is given, hangs up the current channel" revers to the current channel in the docs. Wich channel do they actually mean? |
| 20:19.12 | jackx | the channels wich past the call to the agi script? |
| 20:19.17 | MikeJ[Laptop] | if so, just go to the /usr/src/asterisk/apps dir (or wherever your modified source is) and type cvs diff -u app_queue.c |
| 20:19.20 | [TK]D-Fender | MikeJ[Laptop] : Yup, thats it |
| 20:19.30 | SarahEmm | jackx: yep |
| 20:19.55 | *** join/#asterisk stkn (nobody@stkn.developer.gentoo) |
| 20:19.56 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
| 20:20.29 | jackx | What is no calls are past and want to run the script standalone? |
| 20:22.24 | [TK]D-Fender | MikeJ[Laptop] : http://pastebin.ca/13754 |
| 20:22.36 | xcore | WTF! |
| 20:23.11 | [TK]D-Fender | MikeJ[Laptop] : modified from my original non-working one to simplify test circumstances |
| 20:25.20 | [TK]D-Fender | I'm clearly not doing something very basic properly |
| 20:30.59 | *** join/#asterisk imcdona (~imcdona@imail.speakeasy.net) |
| 20:31.39 | yaaar_work | hey guys. i've got an asterlink account, and when i call out the callerid shows someplace in virginia. they gave me two sections to put into iax.conf ...one with [235454] and one with [asterlink] ....where do i put the callerid="foo" <123> line? |
| 20:31.55 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 20:32.02 | imcdona | Can somone please call me via IAXTEL on 17009452475 and see if you get my voicemail? |
| 20:32.19 | MattH | Hi... I'm trying to get a variable from asterisk into a script I wrote... using PHPAGI... can anyone see what's wrong with this and perhaps point me in the right direction? http://pastebin.com/297336 |
| 20:32.43 | Ariel_ | yaaar_work, in your peer setting |
| 20:33.12 | nestAr | yaaar_work: or you can do a SetCallerID() within your outgoing call logic |
| 20:33.58 | *** join/#asterisk lunchbox08 (~geoff@66-193-73-162.gen.twtelecom.net) |
| 20:34.09 | lunchbox08 | Hello Everyone... |
| 20:34.16 | nestAr | yaaar_work: like so.. |
| 20:34.16 | nestAr | exten => _1NXXNXXXXXX,1,SetCallerID(IgLou Internet <5029663848>) |
| 20:34.17 | nestAr | exten => _1NXXNXXXXXX,2,Dial(Zap/r1/${EXTEN}) |
| 20:34.33 | [TK]D-Fender | ok, well I've got to be going.... maybe another day. |
| 20:34.45 | lunchbox08 | I am having an issue with PlayTones playing back to the PSTN. Is this the common behavior |
| 20:35.44 | jackx | hmm isn't there just a normal protocol voor AGI? |
| 20:36.01 | jackx | tapping the console looks pretty dirty to me |
| 20:36.13 | eKo1 | define 'dirty' |
| 20:36.31 | yaaar_work | Ariel_: i set it in the peer with "callerid="Name" <xxxxxxxxxx>" right? |
| 20:36.37 | yaaar_work | nestAr: cool i'll try that |
| 20:37.15 | Ariel_ | yaaar_work, for outbound it uses peer yes. |
| 20:37.57 | TESTER2 | ${EXTEN:1} remove the fisrt, but I want to replace the first number of the extension with 0 (or add 0) |
| 20:38.27 | maveric | TESTER2 thats easy |
| 20:38.38 | imcdona | Can somone please call me via IAXTEL on 17009452475 and see if you get my voicemail? |
| 20:38.40 | maveric | 0${EXTEN:1} |
| 20:38.43 | lunchbox08 | Should the Application Playtones. Be able to play back (busy or ring) tones back over the PSTN? |
| 20:38.50 | TESTER2 | thanks maveric |
| 20:45.25 | Lee__ | wow. moh with mpg321 sounds AWESOME! |
| 20:46.38 | eKo1 | eh, no. mpg321 sucks. |
| 20:48.15 | Lee__ | awesome in a sound art kind of way. |
| 20:48.31 | Lee__ | like, it'd be hard to reproduce that kind of chaos intentionally :) |
| 20:50.50 | eKo1 | and anyways, your supposed to have mpg123 |
| 20:51.33 | *** join/#asterisk dynaqueue (~wbevan@adsl-63-205-16-227.dsl.mtry01.pacbell.net) |
| 20:52.08 | dynaqueue | Hi all, having a strange problem with DTMF tomes and sip |
| 20:52.21 | dynaqueue | seems that I get double entries on a menu |
| 20:52.47 | dynaqueue | This is with broadvoice as the sip channel |
| 20:52.55 | maveric | make sure the phone you are using isn't hearing the tone again via speaker to mic |
| 20:53.09 | maveric | if you can turn down the volume try that |
| 20:53.56 | dynaqueue | Tried that one, just seems to occur on the number 2, also set the dtmfmode to rfc... inthe sip config still with out joy |
| 20:54.53 | dynaqueue | Works with out a problem with internal calls accessing the same menu |
| 20:55.47 | *** join/#asterisk FuriousGeorge (~brian@ool-43516aa2.dyn.optonline.net) |
| 20:56.44 | PakiPenguin | hello everyone |
| 20:56.49 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 20:56.55 | PakiPenguin | where can i get the latest budgetone firmware? |
| 20:57.01 | dynaqueue | Are there any other possible tweaks that I can do |
| 20:57.15 | ramtha | how can i add a leading zero to the src number of a zap to ip call? |
| 20:57.36 | bjohnson | I'm gonna try unlimitel for Kitchener and London DIDs. Any other suggestions? PS sixtel has lost my business due to non-working DIDs and unanswered service requests |
| 20:58.32 | [hC] | damn. everyone seems to suck lately |
| 20:58.39 | [hC] | even voiopjets call quality has got really bad for me lately |
| 20:58.54 | gatty | PakiPenguin: vendor recommended firmware is from http://www.voiptalk.org/products/gt_update.html |
| 20:58.57 | bjohnson | voipjet's been bad for me for couple of months now |
| 20:59.01 | eKo1 | PakiPenguin: from granstream.com |
| 20:59.18 | [hC] | who is still any good? with a decent price? |
| 20:59.35 | [hC] | for north american outgoing coverage |
| 21:00.00 | harryvv | Its to bad that asterisk could have the ability to sence packet lag and log it..then just before the call is initiated it checks on that networks performance and establishes a call though them or skips to the next network. |
| 21:00.31 | [hC] | yeah, is there any way to set what limits you want to avoid dialing a provider on? for latency? |
| 21:00.36 | citats | harryvv: what like the qualify statement? |
| 21:00.49 | harryvv | I dont know. Its just a idea. |
| 21:01.13 | citats | look at the sample configs at the qualify statements |
| 21:03.29 | jjhall | Do SIP calls get routed to a proper context based on sip.conf friend/user entries now or do we still need to send them to a particular extension based on the register statement? |
| 21:04.02 | *** join/#asterisk doughecka_ (~Tad@doughecka.user) |
| 21:04.34 | rogercharlie | voipjet has been shakey for me as well |
| 21:05.41 | FuriousGeorge | hey all, im trying to log a remote softphone (x-lite) into my asterisk server. the dialplan is ok, but i must be having ant problems cuz i hear no voice. right away, in the sip debug output i notice its broadcasting on 5062 not 5060. i was gonna try fwding port 5062 on both ends. |
| 21:06.12 | FuriousGeorge | i fave 2 questions: does what im saying make sense, and were is 5062 specified? |
| 21:07.17 | Lee__ | can someone here comment on the functionality of the patch to CVS for moh with ogg/vorbis files? |
| 21:07.20 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 21:07.51 | rogercharlie | i would not port forward 5062, xlite has a lotta menus, just find the right entry |
| 21:08.07 | dynaqueue | jjhall for 1.0.7 I have SIP working based on context |
| 21:09.00 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
| 21:09.00 | dynaqueue | Just my damn DTNF that aint working at the moment ! |
| 21:09.10 | *** join/#asterisk mozrat (~mozrat@mozrat01.vm.bytemark.co.uk) |
| 21:10.19 | mozrat | hello chaps - need a bit of advice. My caller ID numbers (from BT in the UK) come through missing the leading 0. I'd like to prefix that caller ID with 90 so people can just hit the redial button and it will just work. Is the best way to do this to use the prefix() cmd in the dial plan or is there somewhere else |
| 21:10.55 | mozrat | Sorry - the SetCallerID command in conjunction with prefix |
| 21:11.13 | FuriousGeorge | mozrat: i know there is a setcallerid app u can invoke in dialplan |
| 21:11.18 | FuriousGeorge | but i never used |
| 21:11.30 | FuriousGeorge | n/m :) |
| 21:11.49 | mozrat | Thanks FuriousGeorge |
| 21:12.41 | *** join/#asterisk abatista (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 21:15.19 | *** part/#asterisk roche (~roche@216.194.173.2) |
| 21:15.25 | *** join/#asterisk RolandHaeder (~root@i528C27DB.versanet.de) |
| 21:15.43 | harryvv | anyone here used xlite before and its ports requirment? |
| 21:16.05 | RolandHaeder | nope |
| 21:16.09 | harryvv | :) |
| 21:16.21 | RolandHaeder | not me :) |
| 21:16.39 | RolandHaeder | I got this here: Jun 8 22:17:31 ERROR[10350]: Unable to register channel '/dev/ttyI0' |
| 21:16.43 | RolandHaeder | hmmm :( |
| 21:16.50 | FuriousGeorge | harryvv: i may be having the same problem, i dunno. it always works insde my lan |
| 21:17.01 | FuriousGeorge | but never get audio outside |
| 21:17.25 | gatty | I've used xlite... you need to allow 8000/UDP both ways through the firewall for audio to work. |
| 21:18.19 | FuriousGeorge | harryvv: i was noticing in sip debug that port 5062 was what was being used. but i cant links into the router on that network to try the change |
| 21:18.36 | FuriousGeorge | gatty: ill try that too |
| 21:19.13 | RolandHaeder | I got OGG stream into asterisk working: musiconhold.conf: default => mp3:/usr/bin/ogg123 -q -d wav -f /dev/stdout http://x.x.x.x:8000/asterisk |
| 21:19.17 | gatty | our config here used 5060/UDP and 8000/UDP both in and out (before we upgraded our firewall code, which is now SIP-aware) |
| 21:19.49 | *** part/#asterisk hassler (~hassler@r-corp.hcst.com) |
| 21:20.33 | *** join/#asterisk Dus10 (~Dus10@68-248-179-130.ded.ameritech.net) |
| 21:20.40 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 21:21.54 | FuriousGeorge | gatty, how come i see this in sip debug: Via: SIP/2.0/UDP 10.0.0.10:5062;branch=z9hG4bK343683CFA61932AA9331836blah.... |
| 21:22.03 | *** part/#asterisk RolandHaeder (~root@i528C27DB.versanet.de) |
| 21:22.06 | FuriousGeorge | the 5062 specifically |
| 21:22.28 | gatty | are you using STUN? |
| 21:22.40 | FuriousGeorge | hmmm.... |
| 21:22.50 | *** join/#asterisk sandstorm (~dxz@195.174.225.113) |
| 21:23.06 | sandstorm | sorry, what is asterisk? |
| 21:23.22 | zoa | haha |
| 21:23.27 | sandstorm | and PBX? |
| 21:23.30 | eKo1 | you didn't just say that did you |
| 21:23.31 | FuriousGeorge | its like that lady in the old movies that transfers calls |
| 21:23.37 | zoa | its a gay porn channel |
| 21:23.38 | eKo1 | * <--- that's an asterisk |
| 21:23.40 | ramtha | lol |
| 21:23.47 | FuriousGeorge | peanut butter and xelly |
| 21:23.48 | sandstorm | :D |
| 21:24.03 | eKo1 | hmm...i thought this was the beastily channel |
| 21:24.08 | sandstorm | i meant not the * :P |
| 21:24.23 | ChkDigit | Why not? |
| 21:24.37 | FuriousGeorge | gatty: i saw that my secondary stun server was set by default, lemme try again |
| 21:24.41 | sandstorm | cuz i know * is asterisk |
| 21:24.54 | sandstorm | i wondered what PBX is :=) |
| 21:25.07 | FuriousGeorge | seriously? |
| 21:25.08 | eKo1 | ~pbx |
| 21:25.10 | jbot | from memory, pbx is a Private Branch eXchange |
| 21:25.10 | ramtha | what you think what it is? |
| 21:25.10 | zoa | its like a telephone thing for in companies |
| 21:25.12 | gatty | PBX = Private Branch eXchange (if I remember rightly) |
| 21:25.20 | eKo1 | ~asterisk |
| 21:25.22 | jbot | [asterisk] a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org |
| 21:25.26 | FuriousGeorge | ~nj |
| 21:25.27 | jbot | rumour has it, nj is home to the Sopranos |
| 21:25.35 | sandstorm | yes im serious that i dont know what pbx is |
| 21:25.44 | sandstorm | sorry for my ignorance |
| 21:25.55 | Hmmhesays | odd there is on debian source package for asterisk |
| 21:25.58 | Hmmhesays | *on/no |
| 21:26.09 | FuriousGeorge | it answers a call for a business and does something with it, but the user doesnt really see it, if its done right. its how a lot of people can share a few lines |
| 21:26.20 | ChkDigit | It is ok. Just normally people join this room for help with asterisk, AFTER they know what it is. |
| 21:26.35 | FuriousGeorge | check out www.wikipedia.com i bet there is something there better than what i said |
| 21:27.01 | sandstorm | like an automatic telephone exchanging system |
| 21:27.07 | sandstorm | right? |
| 21:27.15 | ramtha | yes |
| 21:27.18 | FuriousGeorge | sandstorm: sounds right, i dont know what that is |
| 21:27.29 | sandstorm | so on what platform does asterisk work on ? |
| 21:27.39 | ramtha | linux.. |
| 21:27.47 | eKo1 | any posix compliant platform...yeah right |
| 21:27.59 | FuriousGeorge | cygwin :) |
| 21:28.03 | sandstorm | so all the phone lines are managed by a linux computer? |
| 21:28.10 | FuriousGeorge | exactamundo |
| 21:28.22 | FuriousGeorge | check out the big brains on sandstorm (movie quote) |
| 21:28.43 | sandstorm | im learning fastly :=) |
| 21:28.58 | FuriousGeorge | sandstorm: but its not only "plain old telephone service" lines (pots) its voip lines of many flavors |
| 21:29.26 | FuriousGeorge | <simpsons> me fail english!? that unpossible </simpsons. |
| 21:29.33 | sandstorm | so it manages the next generation phone calls over ip's |
| 21:29.48 | FuriousGeorge | its even got its own flavor of voip which, im comming to learn, is delicious |
| 21:30.27 | FuriousGeorge | sandstorm: yeah if you wanna try it, get some free sip dialtone accounts and a few shareware sip phones |
| 21:30.33 | mozrat | are there any Cisco 7940/60 users here atm? |
| 21:30.36 | jackx | Is there a way of placing a call for an extension in a way that he doesn't have to pick up the phone (so it get automaticly connected) |
| 21:30.40 | sandstorm | how do you interact phone lines and computers? via a modem or a different perhiperal, card or smthing? |
| 21:30.41 | FuriousGeorge | you can even run it on cygwin in windows if thats your bag but i wont recommend it |
| 21:30.59 | mozrat | I'm interested to know if you are able to change the functionality of the soft buttons along the bottom of the display |
| 21:31.10 | sandstorm | *peripheral |
| 21:31.25 | sandstorm | im using Gentoo on my desktop |
| 21:31.35 | gatty | mozrat: not that I've found on mine but hey I'm no expert on those... maybe with some sort of XML services file you can? |
| 21:31.45 | FuriousGeorge | sandstorm: you know in the old movices where the operator would say "ill connect you" by pulling one plug out a machine and sticking it another. kinda like that only you give it a "scrip" to follow for in and out calls |
| 21:31.58 | sandstorm | yes i know that |
| 21:31.59 | FuriousGeorge | and it knows what to do with them |
| 21:32.26 | FuriousGeorge | sandstorm: do you use linux? |
| 21:32.40 | sandstorm | so the incoming call must include a meta information that says which plug will it be put in right |
| 21:32.46 | sandstorm | FuriousGeorge: gentoo linux on desktop |
| 21:32.53 | sandstorm | FuriousGeorge: winxp on laptop |
| 21:33.00 | mozrat | gatty - this is my last attempt to find out I think, I'm fairly sure you can't program them in the SIP image |
| 21:33.05 | mozrat | in the skinny image you certainly can |
| 21:33.08 | mozrat | it's a shame |
| 21:33.16 | FuriousGeorge | sandstorm: no, you determine what to do based on what line it comes in on or what time of the day it is, or what the cid is, or all of the above or other |
| 21:33.17 | doughecka_ | yea, cisco cripples the sip firmwae |
| 21:33.23 | mozrat | doughecka_: aye |
| 21:34.01 | sandstorm | so you define the rules previously and system works with those rules |
| 21:34.10 | FuriousGeorge | think of it as giving that lady a set of rules with different priorities to do her thing |
| 21:34.12 | doughecka_ | although I have gotten ours to lookup client names in a sql database that our CMS software uses, and dail the number out of it |
| 21:34.12 | sandstorm | as ive understood |
| 21:35.27 | FuriousGeorge | i just started using it, so i dont know much, but in one office i have it take an incomming call on a certain line, ring a certain phone for 15 secondds, and then if no one answers it makes a transfer to another building |
| 21:35.42 | FuriousGeorge | and it does outgoing, but thats easier for a small office |
| 21:35.46 | doughecka_ | sandstorm: download asterisk@home |
| 21:35.53 | sandstorm | oh sounds useful. |
| 21:36.00 | FuriousGeorge | doughecka_: ive never used it, but isnt that sorta a cop out?? |
| 21:36.01 | doughecka_ | no need to even know what goes behind the scenes |
| 21:36.19 | doughecka_ | well, you can make custom things inside it, for menus and such |
| 21:36.34 | FuriousGeorge | but thats its own distro right? |
| 21:36.45 | doughecka_ | but for a beginner, it gets it running, then you can go in and poke around and see how it works |
| 21:36.46 | sandstorm | but my phone lines are not enabled for voip feature |
| 21:36.49 | FuriousGeorge | so he would have to nuke his gentoo desktop, no? or come up with another computer |
| 21:36.54 | doughecka_ | it uses cent os 4 and blows an existing install away |
| 21:37.00 | FuriousGeorge | sandstorm: play around with free voip accounts |
| 21:37.10 | FuriousGeorge | and some free softphones |
| 21:37.18 | doughecka_ | but I belive they have a standalone install |
| 21:37.25 | FuriousGeorge | www.xten.com www.sipphone.com for example |
| 21:37.26 | doughecka_ | or install AMP, since thats the major piece |
| 21:37.41 | sandstorm | wont i need a voip enabled phone line? |
| 21:38.03 | FuriousGeorge | sandstorm: you buy a modem board, with daughter modems, on it to handle analog lines |
| 21:38.05 | Hmmhesays | go to the wiki and start reading |
| 21:38.13 | FuriousGeorge | www.voip-supply.com for example tdm400 |
| 21:38.27 | sandstorm | i see |
| 21:38.35 | Hmmhesays | http://www.voip-info.org/wiki-What+is+VOIP |
| 21:38.49 | FuriousGeorge | sandstorm: and most importanly you read a lot |
| 21:38.52 | sandstorm | thats useful for small companies i think |
| 21:38.54 | FuriousGeorge | www.voip-info.org |
| 21:38.58 | FuriousGeorge | !docs |
| 21:39.01 | sandstorm | yes thats the point, i gotta read much |
| 21:39.01 | FuriousGeorge | ~docs |
| 21:39.03 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 21:39.16 | FuriousGeorge | the last two are my fav :) |
| 21:39.23 | *** part/#asterisk lunchbox08 (~geoff@66-193-73-162.gen.twtelecom.net) |
| 21:39.36 | sandstorm | i was just chilling around the channels and aimlessly joined the channel but im learning new things thats good. |
| 21:39.46 | FuriousGeorge | sandstorm: i use it at home to replace a broken doorbell my landlord wont fix |
| 21:39.50 | gatty | which reminds me... my X101P broke and isn't seen on the PCI bus so I've ordered a TDM400P with an FXO module :) |
| 21:39.52 | Hmmhesays | don't be afraid to ask questions you may think are stupid, if you have actually made an attempt to find the answers on your own first |
| 21:40.15 | FuriousGeorge | gatty: exciting times :) |
| 21:40.19 | FuriousGeorge | these are |
| 21:40.26 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 21:40.26 | gatty | just a shame its not in stock :( |
| 21:40.40 | sandstorm | im also using skype for voice chat, thats useful for computer to computer voice chatting. |
| 21:40.49 | Hmmhesays | anyone got a recommendation for a decent video phone? |
| 21:41.05 | FuriousGeorge | omg, i bought from somewhere and asked for 2 day shipping, when i contect them a week later they tell me theyre still out of stock |
| 21:41.09 | Hmmhesays | skype is good for hop on/off calls also, very cheap |
| 21:41.24 | sandstorm | think so. |
| 21:41.28 | FuriousGeorge | never told me nothing, didnt give me a chance to cancel the order, just kinda kept quiet about it |
| 21:41.36 | Hmmhesays | cause they wanted your money |
| 21:41.38 | sandstorm | especially calling someone in america is very cheap |
| 21:41.43 | Hmmhesays | now is no one out there using video phones? |
| 21:41.59 | gatty | dunno why the X101P failed though... its definitely seated and doesn't show up no matter which PCI slot I put it in |
| 21:42.07 | FuriousGeorge | sandstorm: we started using that at work, too. were switching to asterisk and x-ten softphones |
| 21:42.14 | brookshire | cuz it's a x101P |
| 21:42.17 | brookshire | duh |
| 21:42.29 | FuriousGeorge | are those the ones that are intel modems |
| 21:42.29 | Hmmhesays | I suppose I can post on the mailing list |
| 21:42.50 | Hmmhesays | i need some 15,000rpm scsi drives |
| 21:42.56 | Hmmhesays | it takes like 30 seconds to load the mailing list |
| 21:42.57 | FuriousGeorge | dont we all |
| 21:43.11 | sandstorm | FuriousGeorge: oh brilliant information, thanks for your precious informing me about what this is all about |
| 21:43.26 | FuriousGeorge | you can be an * newb like me too one day :) |
| 21:43.28 | Hmmhesays | I see how you are, no joy for the link I gave you |
| 21:43.44 | FuriousGeorge | lol |
| 21:43.48 | sandstorm | :)) |
| 21:44.13 | FuriousGeorge | sandstorm: this is the best link ive gotten in a while www.bash.org |
| 21:44.22 | FuriousGeorge | not as fun as asterisk but definately funnier |
| 21:44.24 | sandstorm | checking out |
| 21:44.24 | Hmmhesays | i've got a few quotes on there |
| 21:44.37 | Hmmhesays | waiting for approval, sometime in 2009 |
| 21:44.41 | FuriousGeorge | i realized i was a geek when i found every single one funny |
| 21:44.47 | FuriousGeorge | but im still in the closet about it |
| 21:45.46 | FuriousGeorge | if i was trying to get on it i would have said "shit my gf is comming up the stairs" and sign off |
| 21:46.14 | Hmmhesays | that's it? |
| 21:46.19 | FuriousGeorge | no |
| 21:46.23 | Hmmhesays | link |
| 21:47.02 | rogercharlie | anyone like their sip termination? |
| 21:47.04 | FuriousGeorge | then it would say "FuriousGeorge has quit IRC blah.blah@ool.net 'Don't tell anyone'" |
| 21:47.10 | sandstorm | :P |
| 21:47.11 | FuriousGeorge | no this is my fake one, it doesnt exit |
| 21:47.17 | Hmmhesays | ok |
| 21:47.41 | FuriousGeorge | im just wasting irc space with that thought |
| 21:47.47 | sandstorm | topic: gf rulez FuriousGeorge :D |
| 21:47.55 | Hmmhesays | that's 1 1/2 seconds of my life i'll never get bck |
| 21:47.57 | Hmmhesays | *back even |
| 21:48.12 | TESTER2 | Jun 8 17:45:24 NOTICE[1394]: chan_iax2.c:5070 authenticate: No way to send secret to peer 'XXX.XXX.XXX.XXX' (their methods: 4) |
| 21:48.24 | sandstorm | cool testing :D |
| 21:49.24 | FuriousGeorge | apparently everyone has heard this one too: there are 10 types of people in this world, those who underestand binary and those who dont" |
| 21:49.53 | rogercharlie | only those who read the list for 5 minutes |
| 21:49.59 | FuriousGeorge | lol |
| 21:50.01 | sandstorm | lol |
| 21:50.40 | Hmmhesays | yeah you're right |
| 21:50.50 | sandstorm | what about this(i love it) : "/* You are not expected to understand this */" - V6 Unix kernel |
| 21:50.54 | sandstorm | :D |
| 21:51.06 | *** join/#asterisk likwid-- (~likwid@nc-63-173-119-75.dyn.sprint-hsd.net) |
| 21:51.59 | pjz | I think I remember xterm code sthat said something like /* If you think you understand this, you're probably wrong. Read it again. */ |
| 21:53.13 | FuriousGeorge | personally i think its funny when asterisk says something like "Yuck.... buffer overrun" or something like that |
| 21:53.49 | TESTER2 | So someone has succesfully use auth=rsa in iax.conf? |
| 21:54.04 | Hmmhesays | you could put your own error messages in the code and be amused for days |
| 21:54.12 | *** join/#asterisk felipeao (~felipeao@200.146.100.6) |
| 21:54.15 | FuriousGeorge | Hmmhesays: OMG gbg |
| 21:54.19 | FuriousGeorge | err, brb |
| 21:55.04 | sandstorm | #99060 , bash.org :))) |
| 21:55.11 | ariel_ | TESTER2, yes |
| 21:55.45 | sandstorm | http://www.bash.org/?99060 :D |
| 21:56.55 | Hmmhesays | label your links |
| 21:57.09 | jackx | is there some kind of pickup method? so i can remotely pickup a phone (wich outputs an dial tone) en then dial the number |
| 21:57.10 | Hmmhesays | can I open that at work? |
| 21:57.28 | TESTER2 | ariel_: could you help me with it? I generate my keys and configure them in iax.conf, but it must be something wrong |
| 21:57.34 | rogercharlie | anyone like their termination anymore, voipjet doesn't always play n n nah nice |
| 21:57.49 | Hmmhesays | vonage is pretty good |
| 21:58.11 | rogercharlie | but they are almost "the man" |
| 21:58.16 | CoaxD | Fuck the man!#$ |
| 21:58.19 | Hmmhesays | naw, they let me use asterisk on their network |
| 21:58.19 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 21:58.21 | CoaxD | err. reflex. sorry. :/ |
| 21:58.50 | rogercharlie | with iax? |
| 21:58.54 | Hmmhesays | sip |
| 21:58.59 | felipeao | I need asterisk to recognize available FXO channels in order for it to just use non-busy Zap channels to make calls. The problem is that asterisk is being able to distinguish a channels thats busy from a channels thats available. Why?? What can I do to solve it? Configs: {extensions.conf}exten => 0,1,Dial(ZAP/g1) {Zapata.conf}signalling = fxs_ks busydetect = yes echocancel = yes callerid=asreceived group => 1 channel => 5-6 |
| 21:59.19 | ariel_ | TESTER2, I use rsa keys for voicepulse and fwd. I have not set it up via my own servers in about a year. |
| 22:00.33 | rogercharlie | feli, the ztcfg -vv looks good before you load ? |
| 22:01.28 | felipeao | roger, I dont even know what this is.. let me check>> |
| 22:01.59 | felipeao | roger, yes it looks just fine |
| 22:02.05 | *** join/#asterisk epac (~epac@65.204.157.77) |
| 22:02.12 | felipeao | any other ideas guys? |
| 22:02.32 | felipeao | btw Im in Brazil :P |
| 22:03.00 | sandstorm | http://www.bash.org/?23601 :D |
| 22:03.22 | rogercharlie | not sure, but change the impedance? |
| 22:03.34 | rogercharlie | something about the signal is diff in diff countries |
| 22:04.06 | *** join/#asterisk sudhir492 (~sudhir@pool-71-114-74-131.washdc.dsl-w.verizon.net) |
| 22:04.07 | felipeao | I tried to use indication's BRazil configs but I get an error on asterisk. I can use FRance's config with no problems, BUT asterisk is still unable to sense when the FXO is busy or available. All the rest works, including when I connect a cable on the fxo |
| 22:04.10 | sudhir492 | hi all |
| 22:04.45 | sudhir492 | anyone knows how to do 3-way calling with Linksys |
| 22:04.49 | rogercharlie | what do you have 2 tdm's for channel 5,6 |
| 22:06.02 | felipeao | whats a tdm? sorry for my "noobness"... Im using them as a group since I want asterisk to just use an FXO channel when this channel is available. I've got 4 FXS and 2FXO in my box |
| 22:06.40 | rogercharlie | i would try with one fxo, one fxs first |
| 22:06.48 | pjz | is there any way to do reverse xfer? |
| 22:06.50 | PBXtech | are the proliant G4 servers still troublesome? |
| 22:07.01 | rogercharlie | then make sure the brazil imedance matches |
| 22:07.22 | felipeao | Ive got no problems making calls or using the FXS.. the only problem is that asterisk is unable to check if the FXO channel is busy or available |
| 22:07.33 | rogercharlie | oh |
| 22:07.37 | felipeao | I can also make calls using the FXO with no problems |
| 22:07.42 | rogercharlie | thats still a lotta cards for one box |
| 22:07.56 | felipeao | just 2 pci slots |
| 22:08.14 | ariel_ | felipeao, are they on there own irq's |
| 22:08.23 | rogercharlie | k, you do have tdm cards |
| 22:08.25 | felipeao | yes they are. |
| 22:08.32 | felipeao | whats a tdm card? |
| 22:08.40 | *** join/#asterisk TedC (~ted@gray.impulse.net) |
| 22:08.55 | ariel_ | the fxs/fxo cards from digium are called TDM400p's |
| 22:08.56 | rogercharlie | quad analog card |
| 22:09.26 | felipeao | oh yeah, I've got a tdm.. |
| 22:11.17 | rogercharlie | you could use failover calling, if one call fails, have the n+101 try the other fxo |
| 22:12.12 | rogercharlie | doesn't check if the line is busy but it keeps hope alive :) |
| 22:12.22 | felipeao | ok, but in this case, what would happen in the fxo thats in use? I mean, both ends would notice that some1 else tried to access that channel and dialed, right? |
| 22:13.10 | rogercharlie | you could disable call waiting on the line, if its on |
| 22:13.27 | rogercharlie | or use the group check for a max of one call per fxo |
| 22:13.51 | FuriousGeorge | how do you guys monitor ur system resources. is there a cli gkrellm type app? |
| 22:14.06 | rogercharlie | I can't believe they changed the syntax on the group checks |
| 22:14.23 | ariel_ | FuriousGeorge, something like top |
| 22:14.37 | ariel_ | for linux |
| 22:14.40 | rogercharlie | took me a month to make them work with queues and the extensions |
| 22:14.47 | FuriousGeorge | yeah, i was hopping for graphs |
| 22:14.50 | rogercharlie | then they change the extension |
| 22:15.10 | FuriousGeorge | framebuffer, ideally |
| 22:15.28 | rogercharlie | err they changed the syntax, warning deprecated dialplan prepare for futility |
| 22:15.45 | ariel_ | rogercharlie, which ones are you talking about. |
| 22:15.54 | ariel_ | using the | instead of , |
| 22:15.59 | felipeao | ý |
| 22:15.59 | felipeao | roger, do u know the syntax of the group check function? |
| 22:16.00 | xmir | |
| 22:17.24 | xmir | whops, sorry. keyboard problems :/ |
| 22:17.39 | felipeao | nvmind, thanks :) |
| 22:17.43 | sandstorm | http://www.bash.org/?77904 :D |
| 22:17.45 | felipeao | I will give it a try |
| 22:17.54 | ariel_ | felipeao, checkgroup and gropucount and setgroup |
| 22:17.57 | harryvv | If anyone has had sucess making xlite work though a remote end router into a asterisk box on the other remote end msg me. |
| 22:18.17 | SkyNetishere | help with MYSQL when I do a query ... select * works fine but select\ 'field1',\ 'field2'\ from.... returns a 1 in the first field and empty in all others |
| 22:18.24 | ariel_ | harryvv, I have it's not that hard to do |
| 22:18.27 | felipeao | ariel_, thx! |
| 22:18.50 | *** part/#asterisk Dus10 (~Dus10@68-248-179-130.ded.ameritech.net) |
| 22:20.05 | ariel_ | felipeao, I spelled groupcount incorrectly. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup |
| 22:20.08 | shido | harryvv, yes |
| 22:20.10 | shido | mine works |
| 22:20.51 | SkyNetishere | nm must escape ,s missed that one :) |
| 22:22.12 | ariel_ | it's time to go home. See you all later. |
| 22:22.18 | felipeao | ariel_, thx, im on it. The only problem is that this command sets the maximum allowed calls/group, though I need asterisk to detect if a channel is already in use, be it on asterisk of not |
| 22:22.42 | TESTER2 | So someone has succesfully use auth=rsa in iax.conf for a friend config? |
| 22:23.03 | harryvv | ariel_ well with all ports his remote end registering on my end we can initiate the call but its a voice passing though the firewall or his xlite is a issue. |
| 22:24.40 | *** join/#asterisk sigterm (sigterm@devious.info) |
| 22:25.06 | TESTER2 | anyway... I'll use md5 it's more simple |
| 22:26.46 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
| 22:27.51 | gtigene | I have always compiled the Linux kernel when updating to a newer version of the kernel but does Asterisk really require this? |
| 22:28.06 | rogercharlie | that setgroup syntax is deprecated, at least in head |
| 22:28.24 | *** join/#asterisk ArkyLady (ArkyLady@adsl-208-191-243-200.dsl.ltrkar.swbell.net) |
| 22:28.26 | rogercharlie | you don't have to recompile kernel! |
| 22:29.12 | *** part/#asterisk GMsoft (~gmsoft@gmsoft.developer.gentoo) |
| 22:29.22 | rogercharlie | but i like to update my kernel sometimes as i put on a new build |
| 22:29.34 | rogercharlie | it makes sure i never know what is really wrong |
| 22:29.59 | TESTER2 | What is the diffenrence between asterisk and a SIP server (example SER)? |
| 22:30.36 | rogercharlie | sip is a proxy, but never is in the middle of a call |
| 22:31.00 | *** join/#asterisk criptos (~criptos@201.128.240.235) |
| 22:31.25 | criptos | is there any way to pickup a line from a iaxy? as *8 at sip or zap channel? |
| 22:31.37 | criptos | I have tried pickup without luck. |
| 22:31.49 | rogercharlie | yah not supposed to work with iax |
| 22:32.13 | rogercharlie | oops meant to say ser is a sip proxy server |
| 22:33.24 | TESTER2 | any link for a complete IVR example? (this is one of the last thing for my asterisk box setup) |
| 22:33.37 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 22:34.18 | rogercharlie | it won't be the last thing |
| 22:34.27 | TESTER2 | I know, but for the moment |
| 22:34.49 | TESTER2 | I want a complete example (entreprise style-like, not just 1..2..3) |
| 22:35.07 | TESTER2 | but it's hard to find |
| 22:36.01 | puppet | IVR? |
| 22:36.27 | TESTER2 | like this site but in more compex: http://users.pandora.be/Asterisk-PBX/IVR.htm |
| 22:36.31 | eKo1 | ~ivr |
| 22:37.21 | TESTER2 | Interactive Voice Response |
| 22:37.25 | puppet | tester2: just make more context |
| 22:37.34 | puppet | tester2: and goto them for new menu |
| 22:38.30 | TESTER2 | I know, but I'll like to see a big one (hundred of lines :) ) |
| 22:38.38 | puppet | write one your self? :o |
| 22:38.50 | puppet | check goto command |
| 22:38.57 | puppet | and make new contexts |
| 22:39.03 | puppet | with s extensions in |
| 22:39.32 | shido | someone needs a SER setup? |
| 22:40.09 | rogercharlie | new contexts are good for star--ing out of an ivr into a specialty ivr |
| 22:42.50 | jackx | where can i find a list of all variables wich can be set ? |
| 22:44.24 | shmaltz | why am I getting this when trying to make a call thru nufone: |
| 22:44.25 | shmaltz | http://pastebin.com/297412 |
| 22:45.15 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 22:45.16 | dalabera | hello |
| 22:45.20 | dalabera | everyone |
| 22:45.32 | *** join/#asterisk escualis (~carlos@200.90.173.215) |
| 22:45.42 | *** part/#asterisk TESTER2 (~Cyber@modemcable219.42-81-70.mc.videotron.ca) |
| 22:46.11 | escualis | Himeko, everyone |
| 22:46.40 | escualis | anyone knows how to fix this.. |
| 22:46.41 | escualis | Jun 8 18:35:49 DEBUG[1332]: Auto destroying call 'F4E9EE159AB74D2DB07BB156C96EFC79@200.90.173.215' |
| 22:46.43 | escualis | please |
| 22:47.32 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:47.40 | dalabera | why this is bothering you? that's a debug message that asterisk use |
| 22:47.45 | harryvv | Looks like i may have accidently disconected :) |
| 22:48.05 | escualis | i've been add a new extension |
| 22:48.11 | harryvv | anyways |
| 22:48.26 | escualis | and i tried one of my friends connect to the server |
| 22:48.29 | harryvv | how many new ip wifi phones been evaluated? |
| 22:48.34 | escualis | but he can't |
| 22:49.17 | *** part/#asterisk sandstorm (~dxz@195.174.225.113) |
| 22:49.36 | harryvv | Jun 8 15:43:45 NOTICE[4333]: chan_iax2.c:6451 socket_read: Peer 'voipjet' is now TOO LAGGED (2079 ms)! |
| 22:49.37 | harryvv | Jun 8 15:43:55 NOTICE[4333]: chan_iax2.c:6445 socket_read: Peer 'voipjet' is now REACHABLE! Time: 45 |
| 22:49.47 | shmaltz | when using iax account from nufone or others and a sip client (spa3000) with asterisk like this: |
| 22:49.48 | shmaltz | Provider < IAX2 > Asterisk < SPA3K) |
| 22:49.50 | shmaltz | I get double ring back tone, anybody knows why? |
| 22:49.56 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 22:50.36 | harryvv | mabey your belcore is set differently |
| 22:51.04 | *** join/#asterisk Katty (Me@0-1pool12-23.nas10.saint-louis1.mo.us.da.qwest.net) |
| 22:52.23 | dalabera | I having problem installing the CDRs, I received the following has anyone seem this? cdr_addon_mysql.c:378 my_load_module: Failed to connect to mysql database DB_*** on localhost. |
| 22:52.49 | dalabera | mysql connection is correct and everything is working, except that asterisk can't connect |
| 22:54.16 | rogercharlie | firewall? table permissions? connecting to port or file? those have all bit me in the bugger |
| 22:54.23 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:54.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:55.38 | shmaltz | JerJer you around? |
| 22:56.03 | shmaltz | ~seen jerjer |
| 22:56.03 | jbot | jerjer is currently on #asterisk. Has said a total of 5 messages. Is idling for 3h 16m 11s |
| 22:59.19 | dalabera | Guys, just to let you know that I have a problem with my perl DBD:mysql install and that was the problem I had |
| 23:02.57 | *** join/#asterisk Jammy (~jammy@24.244.183.236) |
| 23:05.47 | *** join/#asterisk Dovid (~dovi5988@pool-151-198-23-166.mad.east.verizon.net) |
| 23:05.51 | Dovid | hello all |
| 23:06.27 | Dovid | I heard that with the Newer CVS release that there is a new command for the modprobe |
| 23:06.41 | Dovid | its no longer Mocprobe WCFXS. can anyone help |
| 23:06.42 | Dovid | ? |
| 23:07.51 | *** part/#asterisk MasterYoda (~mnicholso@207.111.174.1) |
| 23:07.56 | Dovid | Grrr |
| 23:09.07 | Dovid | having kernel problems have to reformat - never mind |
| 23:10.54 | *** part/#asterisk \Grooby\ (~sl9z@ip24-250-126-171.dc.dc.cox.net) |
| 23:11.02 | blitzrage | Astricon is less than a week away! |
| 23:11.04 | twisted[work] | uhm |
| 23:11.12 | twisted[work] | david, modprobe wctdm |
| 23:11.39 | twisted[work] | and remember, linux commands are case sensitive |
| 23:11.46 | Katty | Dovid: wctdm.o |
| 23:11.56 | blitzrage | twisted[work]: unlike those on OSX? :) |
| 23:11.58 | Dovid | thanks |
| 23:12.13 | blitzrage | twisted[work]: well, its true isn't it? :) |
| 23:12.16 | twisted[work] | oh, Dovid. heh... sorry, misread |
| 23:12.32 | blitzrage | that proves its true! |
| 23:12.40 | twisted[work] | osx == *nix |
| 23:12.40 | Dovid | @twisted[work]: its ok |
| 23:12.47 | blitzrage | twisted[work]: well duh |
| 23:12.53 | twisted[work] | linux == *nix |
| 23:12.56 | blitzrage | twisted[work]: but apparently not case sensitive |
| 23:13.01 | twisted[work] | if a == c and b==c then a==b |
| 23:13.15 | twisted[work] | use a little deductive logic ;P |
| 23:13.48 | blitzrage | twisted[work]: not if its only BASED on a, then a != c, and b != c, thus a != b |
| 23:13.51 | *** join/#asterisk drumkilla (~russell@207.111.174.1) |
| 23:13.51 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 23:13.53 | blitzrage | how ever close it may look :) |
| 23:13.57 | *** join/#asterisk leonk24 (~cg@200.62.182.193) |
| 23:13.59 | blitzrage | drumkilla: booyaka |
| 23:14.10 | leonk24 | helo |
| 23:14.12 | drumkilla | hey blitzrage ! |
| 23:14.15 | escualis | anyone can help me with a little test? |
| 23:14.23 | drumkilla | ack! |
| 23:14.37 | twisted[work] | sorry, drumkilla... walked in at the wrong moment :P |
| 23:14.44 | drumkilla | s'all good |
| 23:14.46 | *** part/#asterisk escualis (~carlos@200.90.173.215) |
| 23:14.51 | leonk24 | help me with a dph-80s dlink ip phone with asterisk |
| 23:15.12 | twisted[work] | 1) do not demand help |
| 23:15.19 | *** mode/#asterisk [+o Cresl1n] by drumkilla |
| 23:15.21 | twisted[work] | 2) tell us what you have tried |
| 23:15.30 | twisted[work] | 3) ASK POLITELY for help |
| 23:15.30 | leonk24 | please of course |
| 23:15.37 | puppet | 3) read docs first |
| 23:15.40 | puppet | ~docs |
| 23:15.41 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 23:15.48 | twisted[work] | puppet, heh |
| 23:15.55 | puppet | twisted[work]: lazy ;P forgot one ;D |
| 23:15.56 | drumkilla | there was already a step 3! |
| 23:15.57 | drumkilla | omg! |
| 23:16.05 | puppet | drumkilla: not my fault! |
| 23:16.08 | *** join/#asterisk h3x (~hex@64.192.116.29) |
| 23:16.09 | h3x | j #uberhack |
| 23:16.12 | h3x | shit |
| 23:16.12 | puppet | drumkilla: im from swedn ! ;D i type slow ;D |
| 23:16.17 | twisted[work] | lol |
| 23:16.23 | drumkilla | puppet, I was in Sweden a couple weeks ago :) |
| 23:16.26 | puppet | and its like late here ;P |
| 23:16.28 | puppet | drumkilla: cool ;D |
| 23:16.38 | puppet | drumkilla: where in sweden? |
| 23:16.39 | drumkilla | 1 in the morning? |
| 23:16.41 | leonk24 | please help me with a dph-80s dlink ip phone with asterisk |
| 23:16.42 | jackx | anyone knows much from the AGI manager? |
| 23:16.44 | twisted[work] | yea, and he failed to bring me back any swedish women or any swedish chocolate |
| 23:16.45 | drumkilla | Stockholm - for VON Europe |
| 23:16.49 | puppet | drumkilla: aye 1 |
| 23:16.51 | puppet | drumkilla: ah cool |
| 23:17.05 | drumkilla | twisted, I have a small Toblerone for you in my laptop bag |
| 23:17.12 | drumkilla | I keep forgetting to give it to you |
| 23:17.19 | puppet | twisted[work]: want me to send chocolate to you? haha |
| 23:17.38 | puppet | small toblerone :o |
| 23:17.39 | blitzrage | drumkilla: I'd have eaten it by now |
| 23:17.40 | leonk24 | I can't make call with this phone.. please help me |
| 23:17.44 | puppet | you know there is like biig toblerone |
| 23:18.11 | drumkilla | puppet, yeah, i know |
| 23:18.14 | *** join/#asterisk a-evol (~a-evol@web2.osloisp.no) |
| 23:18.15 | drumkilla | I got it on my plane, haha |
| 23:18.21 | blitzrage | 7:20 here |
| 23:18.28 | drumkilla | but I told twisted I'd try to get it if there was any way I could |
| 23:18.28 | blitzrage | (in 2 mins) |
| 23:18.29 | a-evol | hi, uhm in amp, what's the user name again for configuring ? |
| 23:18.34 | drumkilla | blitzrage, !!!!!!!!!!!!!!!! |
| 23:18.34 | a-evol | I recall m something |
| 23:18.48 | twisted[work] | drumkilla, you didn't! |
| 23:18.50 | blitzrage | drumkilla: actually... its probably 4:20 in B.C. |
| 23:18.59 | blitzrage | drumkilla: I think their 3 hours behind |
| 23:19.10 | puppet | lool im playign sim city 3000 |
| 23:19.24 | blitzrage | I think they're* |
| 23:19.33 | blitzrage | sheesh... would think I'd have learned grammer by now |
| 23:19.33 | twisted[work] | drumkilla, yay :) |
| 23:19.50 | twisted[work] | drumkilla, but you can get the BIG ones in sweden ;) |
| 23:19.57 | drumkilla | yeah, sorry about that |
| 23:19.58 | twisted[work] | drumkilla, but thanks still ;) |
| 23:20.26 | a-evol | a h |
| 23:20.29 | a-evol | anyone ? |
| 23:21.00 | a-evol | the default login name for AMP's username |
| 23:21.05 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@68.60.76.65) |
| 23:22.26 | blitzrage | twisted[work]: and Canada - I've seen them downstairs in the grocery store |
| 23:23.00 | a-evol | please help me out guys |
| 23:23.07 | a-evol | im totaly new to asterisk |
| 23:23.21 | drumkilla | i think there is a #amp ... |
| 23:23.29 | drumkilla | most people in here don't use it |
| 23:23.35 | a-evol | no |
| 23:23.50 | twisted[work] | blitzrage, downstairs? |
| 23:24.02 | twisted[work] | blitzrage, your groceries have multiple stories? |
| 23:24.05 | blitzrage | twisted[work]: I live in an apartment with lots of businesses attached to it |
| 23:24.27 | twisted[work] | oh |
| 23:24.27 | tzanger | what's that got to do with your groceries telling stories |
| 23:24.29 | blitzrage | twisted[work]: bar, grocery, dentist, dry cleaning, chinese, blockbuster, dollar store |
| 23:24.36 | twisted[work] | blitzrage, heh. |
| 23:24.40 | twisted[work] | wtf is a pample? |
| 23:24.41 | blitzrage | tzanger: shut up j00! |
| 23:24.50 | Katty | tzanger: :< |
| 23:24.53 | blitzrage | twisted[work]: notice I started with the bar :) |
| 23:24.56 | tzanger | I'm hot sweaty and sticky |
| 23:24.59 | twisted[work] | blitzrage, but of course |
| 23:25.00 | Katty | oh |
| 23:25.01 | Katty | k |
| 23:25.12 | blitzrage | twisted[work]: :D |
| 23:25.18 | Katty | twisted[work]: cats pample warm and furry things when happy |
| 23:25.21 | tzanger | Katty: my cat does the same thing... god I can't imagine what he's like, I should shave him |
| 23:25.29 | twisted[work] | Katty, oh.. i thought they cuddled them :P |
| 23:25.49 | twisted[work] | or 'brushed against' as my cats do |
| 23:25.50 | MikeJ[Laptop] | twisted!!!! |
| 23:26.00 | tzanger | kids are cooling off in a cool bath |
| 23:26.03 | tzanger | I should take a cool shower |
| 23:26.03 | twisted[work] | MikeJ[Laptop] ?!? |
| 23:26.07 | tzanger | yeah |
| 23:26.13 | tzanger | I think I'm gonna do that when they go to bed |
| 23:26.33 | tzanger | I gotta be up early and head to Whitby tomorrow early |
| 23:27.03 | tzanger | 200HP drive startup ... one drive driving 4 motors on a cement mixer... "can't go wrong" yet this 4 motor, one drive thing is highly experimental and they didn't put a load reactor on the damn thing |
| 23:27.08 | tzanger | should be interesting |
| 23:27.12 | MikeJ[Laptop] | hehe |
| 23:27.29 | MikeJ[Laptop] | twisted, I thought you were supposed to be working ;) |
| 23:27.47 | tzanger | but, when you're the FAE this is your life :-) |
| 23:28.31 | *** join/#asterisk qwertyiop (~Cain@81-178-196-135.dsl.pipex.com) |
| 23:28.43 | twisted[work] | MikeJ[Laptop], i am, thus my work account |
| 23:28.54 | twisted[work] | i'm waiting on replies to a couple things, and then i have to get back at it |
| 23:30.41 | MikeJ[Laptop] | hehe |
| 23:32.02 | *** join/#asterisk a-evol (~a-evol@web2.osloisp.no) |
| 23:32.09 | a-evol | hi again |
| 23:32.14 | a-evol | anyone here using AMP ? |
| 23:32.46 | qwertyiop | I've got xten set up with asterisk, when i dial someone@sipprovider.com from xten asterisk intercepts it and looks for the extension "someone", is this how its supposed to work? is there a work around so i can dial people direct? |
| 23:33.40 | puppet | lol i suck on sim city :o |
| 23:33.49 | puppet | im - 3226 |
| 23:33.52 | puppet | and i have just started |
| 23:34.37 | a-evol | can't someone please just help me |
| 23:35.06 | *** join/#asterisk egon_L (egon_L@pc-33-19-104-200.cm.vtr.net) |
| 23:35.53 | *** join/#asterisk bdk (~bdk@bdk.caltech.edu) |
| 23:36.43 | Paagrio | hey guys |
| 23:36.50 | Paagrio | is there a complete Asterisk manual you can download anywhere |
| 23:37.02 | Paagrio | (not the handbook) |
| 23:38.08 | JerJer | cvs co asterisk |
| 23:38.23 | JerJer | and open all of the various .c files in your favorite editior |
| 23:43.49 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 23:47.09 | chorix | Hi a-evol, As I see no one can help you at this time |
| 23:47.35 | criptos | ? |
| 23:47.58 | chorix | I saw this chat channel runs better for consulting 3 hours ago |
| 23:48.27 | chorix | Habla alguien en español? |
| 23:49.23 | *** join/#asterisk DaLion (~DaLion@HSE-QuebecCity-ppp3497303.sympatico.ca) |
| 23:49.27 | DaLion | hi all |
| 23:49.29 | *** join/#asterisk meppl (mephisto@p54AAF26B.dip.t-dialin.net) |
| 23:49.43 | DaLion | anyone know how to push SDK to original IP not using the data from headers |
| 23:49.46 | DaLion | meant SDP |
| 23:50.46 | DaLion | symetrick RTP |
| 23:51.13 | JerJer | nat=yes ? |
| 23:51.17 | JerJer | canreinvite=no |
| 23:52.35 | DaLion | canreinvite=no ? |
| 23:52.36 | DaLion | ok |
| 23:52.40 | DaLion | thats on AS |
| 23:53.13 | *** part/#asterisk chorix (alejandro1@host124.201-252-204.telecom.net.ar) |
| 23:53.38 | *** join/#asterisk sigterm (sigterm@devious.info) [NETSPLIT VICTIM] |
| 23:53.46 | DaLion | ok |
| 23:53.50 | DaLion | i still see v=0 |
| 23:53.50 | DaLion | o=root 65201 65201 IN IP4 172.16.0.100 |
| 23:53.50 | DaLion | s=session |
| 23:53.50 | DaLion | c=IN IP4 172.16.0.100 |
| 23:54.38 | *** join/#asterisk Matthew_I (~matthew@64-89-121-30.arpa.kmcmail.net) |
| 23:55.09 | DaLion | IFY..Contact: <sip:asdasde@172.16.0.100>..Content-Length: 0.... |
| 23:55.11 | DaLion | still |
| 23:55.14 | *** part/#asterisk Matthew_I (~matthew@64-89-121-30.arpa.kmcmail.net) |
| 23:55.23 | DaLion | how does new route function work in nat |
| 23:55.26 | DaLion | nat= |
| 23:58.08 | *** join/#asterisk iq (~iq@207-224-101-88.omah.qwest.net) |
| 23:58.32 | iq | hi all |
| 23:58.36 | *** join/#asterisk bdk (~bdk@bdk.caltech.edu) [NETSPLIT VICTIM] |
| 23:59.19 | *** join/#asterisk NightHawke (~NightHawk@66-190-111-175.static.sprn.tx.charter.com) |
| 23:59.28 | *** join/#asterisk brettnem (~brettnem@user-0ccsrb6.cable.mindspring.com) |
| 23:59.34 | NightHawke | i need a hand in setting up ZAP trunks for a FXO card |
| 23:59.48 | NightHawke | extensions actually |