| 00:04.55 | GaVak | What does this mean when trying to dial a zap card? Nov 20 19:04:54 WARNING[3013]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 |
| 00:05.30 | CunningPike | GaVak: Looks like a codec issue |
| 00:06.23 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 00:08.08 | GaVak | hmm, is there any way to check to see if my g729 is loaded properly? |
| 00:10.27 | GaVak | looks like it loads fine when the system starts |
| 00:14.57 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 00:16.47 | *** join/#asterisk billy-jo (n=billy-jo@AClermont-Ferrand-156-1-17-64.w81-251.abo.wanadoo.fr) |
| 00:21.03 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 00:21.16 | GaVak | should I call Digium with Codec problems since I'm using theirs? |
| 00:25.45 | *** join/#asterisk ltd (n=z@202-161-28-21.dyn.iinet.net.au) |
| 00:26.17 | hmmhesays | wow allan thicke is still alive |
| 00:26.18 | TripleFFFF | get a refund get sangoma |
| 00:26.23 | redondos | alamantia: thanks. |
| 00:26.24 | *** part/#asterisk redondos (n=redondos@190.48.14.48) |
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| 00:39.24 | fatpelt | blitzrage: so for your local/ idea to work, do i need to define an ougoing extension? i've tried to do something like this, is it wrong? --cut-- exten => s,1,Goto($CUT(EXTEN,-,2)}\n exten=> _9NXXXXXX,1,Dial(Zap/1/EXTEN) |
| 00:39.40 | fatpelt | blitzrage: that doesn't work though |
| 00:42.56 | hads | You are missing { |
| 00:43.47 | fatpelt | hads: between the $ and the CUT right? |
| 00:43.53 | hads | Yeah. |
| 00:43.54 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 00:44.02 | fatpelt | hads: yeah. it was a typo in ir |
| 00:44.04 | fatpelt | irc |
| 00:44.13 | hads | Sweet |
| 00:44.55 | hads | The second EXTEN needs a ${} around it too. |
| 00:46.24 | fatpelt | hads: exten => s,1,Goto(${CUT(${EXTEN},-,2)}) ... still no No such extension/context 123-xxxxxxx@outgoing |
| 00:47.53 | hads | I haven't been following sorry, I just wanted to point those out mistakes. |
| 00:48.14 | fatpelt | ah. no probs :) |
| 00:48.24 | fatpelt | i'm gonna have to finish this later anyhow |
| 00:48.46 | hads | But I don't think that the EXTEN in the CUT needs the ${}, just the one on line two above. |
| 00:49.14 | fatpelt | oh. i'll add that one later |
| 00:49.24 | CunningPike | fatpelt: Do you need the '-' ? |
| 00:49.38 | fatpelt | CunningPike: it would default to - but i put it just for kicks |
| 00:51.03 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
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| 00:56.01 | Ryanw | how do you tell which codec a channel is using is it the native format, the read format or the write format? |
| 00:56.23 | CoffeeIV_ | I have two * servers have an IAX2 connection between them. Recently, every other call won't connect going between them. There is no message in /var/log/asterisk/full when they don't connect. THen the next call goes through. |
| 01:00.31 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:03.37 | bluregard | Ryanw: show channel <channel_name> |
| 01:04.08 | Ryanw | bluregard, i take it the "Native Format" is the actual codec in use on that channel? |
| 01:04.22 | bluregard | yeah, I think so |
| 01:09.29 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
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| 01:29.23 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:31.57 | KeNroM | question |
| 01:32.10 | dlynes_laptop | answer |
| 01:32.25 | KeNroM | so vicidail is a interface for asterisk? |
| 01:33.34 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 01:33.54 | dlynes_laptop | KeNroM: i don't know...is it? http://www.google.ca/search?client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial_s&hl=en&q=vicidial&meta=&btnG=Google+Search |
| 01:33.55 | KeNroM | ? |
| 01:37.29 | KeNroM | okl |
| 01:37.47 | KeNroM | i need lil help |
| 01:37.49 | dlynes_laptop | ? |
| 01:38.18 | KeNroM | i`m here setting some extensions... |
| 01:38.29 | KeNroM | and when i dail a number it is telling me |
| 01:38.49 | KeNroM | call failed:|not found |
| 01:38.51 | *** join/#asterisk SwK[work] (n=SwK@br0.asteriasgi.com) |
| 01:39.15 | dlynes_laptop | you've got an error in your dial plan, then |
| 01:39.18 | *** join/#asterisk crochat (i=crochat@84-74-145-66.dclient.hispeed.ch) |
| 01:42.24 | KeNroM | no |
| 01:42.35 | knobenheimer | what do you guys recommend for faxing, hylafax or openH323? i need DID support as well |
| 01:42.54 | KeNroM | cause when i use a next extension which was setup 2 months ago |
| 01:42.58 | KeNroM | it works fine |
| 01:44.08 | dlynes_laptop | knobenheimer: how does openh323 give you faxing support? |
| 01:44.58 | knobenheimer | dlynes: well, it says right on the main page "Fax over IP delivery" |
| 01:45.33 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 01:45.56 | dlynes_laptop | knobenheimer: read it a little closer |
| 01:46.08 | dlynes_laptop | knobenheimer: it says the openh323 partners deliver fax over ip |
| 01:46.14 | dlynes_laptop | knobenheimer: it doesn't say openh323 does |
| 01:46.39 | knobenheimer | ahha |
| 01:46.45 | knobenheimer | shoulda looked closer, thanks |
| 01:52.27 | hmmhesays | knobenheimer:L the eternal question |
| 01:54.20 | hmmhesays | knobenheimer: check your pm |
| 02:00.16 | *** join/#asterisk infinity1 (i=foobar@modena.netcal.com) |
| 02:00.31 | infinity1 | can someone send me polycom 2.0.2 software and bootrom? |
| 02:08.53 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:14.29 | infinity1 | quite. wow |
| 02:15.11 | JT | quiet? |
| 02:20.02 | wunderkin | quite quiet? |
| 02:22.01 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 02:25.53 | infinity1 | hanks man |
| 02:26.06 | infinity1 | er /hanks/thanks ;) |
| 02:27.03 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 02:27.09 | *** join/#asterisk Cardoe (n=Cardoe@gentoo/developer/Cardoe) |
| 02:27.23 | Cardoe | Hey guys. |
| 02:27.34 | Cardoe | I'm trying to write a little macro for getting myself football tix. |
| 02:27.42 | Cardoe | Basically to dial over and over again... until I get through |
| 02:28.08 | Cardoe | I've managed to catch all the cases... except where it rings twice and then says "All Circuits are busy" or sometimes it goes to a fast busy. |
| 02:28.52 | *** join/#asterisk alerios (n=alerios@201.244.173.92) |
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| 02:32.43 | *** part/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
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| 02:36.16 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 02:36.25 | Cardoe | http://www.rafb.net/paste/results/n0FqVz96.html |
| 02:36.33 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 02:36.50 | Dovid | ~seen mitcheloc |
| 02:37.15 | jbot | mitcheloc is currently on #asterisk (1d 19h 57m 41s). Has said a total of 1 messages. Is idling for 3h 32m 20s, last said: 'i did, whatchya gonna do about it?!'. |
| 02:37.15 | Dovid | ~seen mitcheloc |
| 02:37.20 | jbot | mitcheloc is currently on #asterisk (1d 19h 57m 46s). Has said a total of 1 messages. Is idling for 3h 32m 25s, last said: 'i did, whatchya gonna do about it?!'. |
| 02:37.20 | Dovid | !seen mitcheloc |
| 02:44.18 | *** join/#asterisk j0 (n=dan@CABLE-72-53-45-212.cia.com) |
| 02:50.31 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 02:53.54 | *** join/#asterisk peka (n=pk@ACD0F08E.ipt.aol.com) |
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| 02:54.40 | killercoder | Hey all |
| 02:54.58 | killercoder | Just finished a cool Asterisk project @ work |
| 02:55.02 | killercoder | wanted to share |
| 02:55.08 | killercoder | any "*" heads in here? |
| 02:55.20 | file | "*" head? wazzat? |
| 02:55.27 | killercoder | Asterisk Head |
| 02:55.33 | killercoder | Also known as asterisk geek |
| 02:55.39 | killercoder | or hairy beast in the corner :) |
| 02:55.42 | file | ah, that makes more sense |
| 02:55.53 | file | no - we got rid of all the asterisk geeks... they were making a mess |
| 02:56.14 | killercoder | hehe |
| 02:58.31 | Cardoe | So can anyone help me out? |
| 02:58.38 | Cardoe | http://www.rafb.net/paste/results/n0FqVz96.html |
| 02:58.45 | Cardoe | Trying to make it constantly redial. |
| 03:00.32 | Nivex | OT: http://www.mattfrye.net/2006/11/gdk_on_patents.html |
| 03:00.59 | *** join/#asterisk Kast (n=Kast@CPE0012173285e1-CM00140454b9ca.cpe.net.cable.rogers.com) |
| 03:01.10 | Kast | How can I block a call(sip) |
| 03:01.17 | Kast | block a number from calling |
| 03:03.44 | SuPrSluG | Kast:put them in a seperate context and don't include them |
| 03:04.02 | Kast | what file? |
| 03:04.26 | Kast | can u elaborate or send me in the right direction |
| 03:06.53 | *** part/#asterisk killercoder (n=killerco@CPE000c41aac393-CM001225009430.cpe.net.cable.rogers.com) |
| 03:10.08 | SuPrSluG | Kast:read up on the include statement. it's where you define who gets to do what. for instance you may have a context for sales and include them in a long distance calls context. |
| 03:11.12 | *** join/#asterisk tim0123 (n=cash247@adsl-71-158-173-186.dsl.rcsntx.sbcglobal.net) |
| 03:11.15 | Kast | thank youy |
| 03:11.21 | tim0123 | hello every one |
| 03:11.41 | Kast | like sip_phones.include |
| 03:11.46 | Kast | that handles my extensions |
| 03:11.59 | tim0123 | I have a cisco 7960 phone and it seems it will not upload a new image |
| 03:12.50 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:17.03 | j0 | does asterisk hold music always have static? :> |
| 03:18.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:18.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:19.13 | file | russellb: bad Russell bad |
| 03:19.15 | peka | just to let you know (because i asked for it yesterday): the defaut value of silencethreshold is 128 in the source code |
| 03:19.40 | SuPrSluG | Kast:usually in extensions.conf include => context |
| 03:20.01 | peka | it is defined as an int but i cant find an upper limit |
| 03:20.24 | russellb | how's it going? |
| 03:20.35 | file | not bad, knocking out bugs |
| 03:20.45 | file | had a wild signed linear endianness problem today >_< |
| 03:21.08 | file | like - shouldn't you be studying and not on IRC? |
| 03:21.19 | russellb | yes. |
| 03:21.29 | russellb | oh well! |
| 03:22.41 | russellb | static int? |
| 03:22.58 | russellb | or ... static FILE *? |
| 03:23.02 | file | if you want. |
| 03:23.05 | file | eeeeeeep |
| 03:24.57 | russellb | file: 4.5 weeks |
| 03:25.04 | file | ZomgZ |
| 03:25.05 | file | RUN |
| 03:26.16 | peka | another question: when i use the "s" flag in VoiceMailMain() to turn off the password prompt, can *anyone* hear my messages or just me when i call from my extension |
| 03:27.05 | russellb | peka: depends on how you set up your dialplan |
| 03:27.20 | peka | k |
| 03:27.26 | russellb | generally you base the mailbox on callerid |
| 03:27.37 | russellb | VoicemailMail(${CALLERID(num)}|s) for example |
| 03:28.53 | russellb | s/Mail/Main/ ... |
| 03:29.09 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:29.12 | peka | perl... |
| 03:30.04 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 03:30.43 | peka | so if i assume noone else has physical access to my phone would it be safe to turn off the pass prompt? |
| 03:31.17 | peka | or is it possible for someone else to "steal" my extension |
| 03:33.09 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 03:35.46 | dioedu | hello all, someone apply patch for the bug #6559 ? EXITEMPTY, EXITXXX events ? |
| 03:36.32 | *** join/#asterisk axisys (n=axisys@69.143.190.152) |
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| 03:50.28 | peka | afk |
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| 03:54.27 | *** join/#asterisk Katty (n=Administ@68-119-251-157.dhcp.cpgr.mo.charter.com) |
| 03:54.31 | Katty | evening. |
| 03:55.21 | dlynes_laptop | perv |
| 03:55.23 | Katty | allo, Corydon |
| 03:55.51 | Katty | today is a Good Day(tm) |
| 03:55.54 | Cardoe | how do you guys feel about openPBX? |
| 03:55.55 | dlynes_laptop | heya katts |
| 03:56.13 | dlynes_laptop | It's a branch of an earlier version of asterisk |
| 03:56.29 | Katty | i've not used it before. |
| 03:56.33 | Katty | i'm still tryin to get asterisk down |
| 03:56.36 | bsdfreak | hi |
| 03:56.51 | Katty | in fact, i'm still tryin to get linux down! |
| 03:56.53 | dlynes_laptop | heya freak |
| 03:57.10 | Katty | file: i have two screens on my linux box :> |
| 03:57.17 | Katty | file: it makes me all giddy inside :>> |
| 03:57.44 | bsdfreak | sup dlynes_laptop |
| 03:57.52 | dlynes_laptop | Katty: so you got your dragging between screens working? |
| 03:58.08 | Katty | dlynes_laptop: nah, but i've not set the resolution and refresh rate the same yet |
| 03:58.41 | bsdfreak | dual monitors ftw |
| 03:58.47 | file | Katty: yay |
| 03:58.57 | bsdfreak | sup file |
| 03:59.09 | file | I am... compiling mplayer from source! weeeeeee |
| 03:59.10 | file | you? |
| 03:59.17 | Katty | file: you brave soul you |
| 03:59.31 | bsdfreak | just relaxing |
| 03:59.41 | bsdfreak | mplayer, yeesh. that'll take a while. |
| 03:59.55 | dlynes_laptop | file: you masochist! |
| 03:59.59 | bsdfreak | haha |
| 04:00.06 | dlynes_laptop | bsdfreak: after he figures out all the interdependencies, after about 60 recompiles :) |
| 04:00.19 | bsdfreak | hehe. |
| 04:00.31 | bsdfreak | usually i just install mplayer from ports. |
| 04:00.45 | dlynes_laptop | i usually install it from source |
| 04:00.56 | dlynes_laptop | otherwise i don't get all the pieces of hte puzzle that i want |
| 04:01.56 | dlynes_laptop | about the only files I can't play without using wine are quicktime 6 files |
| 04:03.20 | justinu|laptop | it takes one command on gentoo too... emerge mplayer |
| 04:05.01 | Katty | justinu|laptop: butbut |
| 04:05.04 | Katty | justinu|laptop: that's no fun |
| 04:05.23 | justinu|laptop | it's fun, it just takes forever |
| 04:05.48 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 04:06.31 | justinu|laptop | if sorting out a serious dependancy tree is your idea of fun, you're a wild one ;) |
| 04:08.16 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
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| 04:08.55 | *** join/#asterisk PMantis (n=pmantis@cpe-69-207-130-14.rochester.res.rr.com) |
| 04:10.23 | FuriousGeorge | hey all. meetme wont relay sip video right? |
| 04:10.54 | Katty | FuriousGeorge: your /nick is /adorable/ |
| 04:11.00 | Katty | FuriousGeorge: just so ya know. |
| 04:11.06 | FuriousGeorge | Katty: thanks |
| 04:11.14 | FuriousGeorge | ~FuriousGeorge |
| 04:11.16 | jbot | i heard furiousgeorge is a knife-fighting (cable) monkey last seen with The Man with the Yellow Bat |
| 04:12.24 | FuriousGeorge | Katty: i hadnt seen you around for a while |
| 04:12.34 | Katty | FuriousGeorge: i've been hiding in file's closet. |
| 04:12.46 | justinu|laptop | any fun? |
| 04:12.48 | Katty | FuriousGeorge: or, perhaps, moving offices to a new location. |
| 04:12.50 | FuriousGeorge | must be crowded in there with the two of you |
| 04:12.55 | Katty | FuriousGeorge: not really. |
| 04:13.15 | FuriousGeorge | Katty: perhaps cozy is more like it |
| 04:13.48 | Katty | FuriousGeorge: mayhaps. |
| 04:15.51 | PMantis | Anyone have some good links on picking a processor/RAM combo for a a server to support 250 concurrent calls? |
| 04:16.41 | PMantis | s/dial/dual |
| 04:20.03 | FuriousGeorge | so anyone know what happens with meetme and sip video? i figure the video rtp doesnt make it past * |
| 04:21.42 | *** part/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 04:24.29 | *** join/#asterisk ManxPower (n=manxpowe@222.sub-70-196-255.myvzw.com) |
| 04:25.16 | PMantis | 250 concurrent calls... how big of a server? |
| 04:31.46 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 04:32.29 | hoobastooba | hey all |
| 04:33.05 | hoobastooba | can anyone tell me how it is possible for a caller to get a busy signal when a call goes into queue? I have agents availalbe. |
| 04:34.22 | hoobastooba | in my database, my lastapp is queue and disposition is busy |
| 04:37.29 | hoobastooba | everyone has gone to bed it seems |
| 04:37.46 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:38.25 | dlynes_laptop | yep yep yep!!! |
| 04:39.26 | hoobastooba | dlynes_laptop: is that supposed to be possible for a queue to get a busy signal? |
| 04:39.46 | dlynes_laptop | i have no idea |
| 04:39.49 | dlynes_laptop | i know nothing about queues |
| 04:42.50 | axisys | hi guys.. is sipura 3000 which is linksys spa-3000 now is the best if i want to use asterisk w/ my existing voip provider? |
| 04:44.00 | dlynes_laptop | axisys: what does a sipura 3000 have to do with a voip provider? |
| 04:44.10 | dlynes_laptop | axisys: sipura 3000 is for a pstn provider |
| 04:45.04 | axisys | my voip is vonage which connects to the wall |
| 04:45.37 | axisys | then all phones in my apt goes to the rest of the walls .. that way all wall jacks are voip |
| 04:45.41 | dlynes_laptop | ah...yeah...vonage is all locked up |
| 04:45.56 | axisys | instead of to the wall can i connect it to sipura? |
| 04:46.11 | dlynes_laptop | axisys: i guess you want a sipura 3000 or a tdm400p or a sangoma a200 then |
| 04:46.47 | axisys | i am bidding on ebay for a sipura 3000.. so i will stick it |
| 04:47.12 | axisys | dlynes_laptop: but i should be able to vonage -> sipura -> phone instead correct? |
| 04:47.24 | axisys | currently vonage -> wall -> phone |
| 04:47.38 | dlynes_laptop | phone -> sipura 3000 -> pap2 -> vonage |
| 04:47.56 | axisys | cool |
| 04:48.08 | dlynes_laptop | i.e. phone -> fxs -> sipura 3000 -> fxo -> fxs -> pap2 -> cat5 -> vonage |
| 04:48.49 | axisys | dlynes_laptop: hmm. i may be little confused |
| 04:48.51 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 04:49.10 | axisys | vonage -> walljack -> phone is how i have setup |
| 04:49.17 | axisys | now |
| 04:49.55 | dlynes_laptop | phone -> walljack -> fxs -> spa-3000 -> fxo -> fxs -> pap2 -> Internet -> vonage |
| 04:50.27 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-0a9d0c9236ddd4d8) |
| 04:51.29 | *** join/#asterisk converx (n=locid@206-248-162-194.dsl.teksavvy.com) |
| 04:51.38 | axisys | why this? fxo -> fxs -> pap2 -> Internet -> vonage |
| 04:51.48 | axisys | fxo -> vonage wont work? |
| 04:52.12 | dlynes_laptop | vonage uses a device called a 'linksys pap2' to give you dial tone |
| 04:52.20 | dlynes_laptop | it has two fxs ports on it |
| 04:52.28 | dlynes_laptop | one labelled line 1, and the other labelled line 2 |
| 04:52.41 | dlynes_laptop | line 1 is typically used for voice, and line 2 is typically used for fax |
| 04:52.51 | axisys | i have that adapter.. i got it when i signed up w/ vonage |
| 04:52.55 | dlynes_laptop | exactly |
| 04:53.01 | axisys | i am looking at it now |
| 04:53.06 | dlynes_laptop | the fxo port is the line in port on the sipura 3000 |
| 04:53.13 | converx | anyone knows how I can set musiconhold in realtime? |
| 04:53.15 | axisys | i have been using vonage for more than a yr |
| 04:53.30 | converx | is there a realtime table for musiconhold.conf file? |
| 04:55.19 | axisys | dlynes_laptop: i have a motorola vt1005V model voice terminal w/ 2 lines.. i only use one .. to activate the second one i have to pay extra |
| 04:55.33 | axisys | i received it from vonage |
| 04:56.15 | dlynes_laptop | ah...i guess they don't use the pap2's in whatever country you live in, then |
| 04:56.24 | axisys | usa |
| 04:56.34 | dlynes_laptop | ok, or they changed the eq they were using :0 |
| 04:56.40 | axisys | i am one of the early subscribers |
| 04:56.46 | dlynes_laptop | they used to use pap2's in canada and the us |
| 04:56.55 | axisys | hmm |
| 04:57.25 | axisys | i could by the linksys model . best buy sells it here.. they call it vonage adapter :P |
| 04:57.36 | axisys | *buy |
| 04:58.05 | dlynes_laptop | yeah...that would be it |
| 04:58.23 | axisys | dlynes_laptop: so i need that? |
| 04:58.33 | converx | is there a mysql table definition for musiconhold.conf config file? |
| 04:58.34 | dlynes_laptop | nope...the motorola probably does the same thing |
| 04:59.02 | dlynes_laptop | converx: it seems there's nobody on right now that uses realtime extensions, or they just don't use moh in realtime |
| 04:59.19 | *** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 04:59.59 | axisys | dlynes_laptop: so can i interpret this part spa-3000 -> fxo -> fxs -> pap2 -> Internet -> vonage |
| 05:00.22 | axisys | as spa-3000 -> fxo -> line 1 of vonage adapter |
| 05:00.43 | dlynes_laptop | as spa3000 -> fxo -> fxs -> vonage adapter -> Internet -> vonage |
| 05:00.44 | dlynes_laptop | yes |
| 05:00.58 | axisys | cool! |
| 05:01.30 | axisys | dlynes_laptop: is this something i need to worry about? http://www.telephreak.org/PAP2/ |
| 05:03.16 | dlynes_laptop | something you need to worry about? |
| 05:03.26 | dlynes_laptop | no...you're using a motorola ata, not a linksys pap2 |
| 05:04.07 | dlynes_laptop | however, you might be able to log into the motorola ata using the default username and password and change the registrar and that kinda thing |
| 05:04.21 | dlynes_laptop | so that it points to your asterisk box |
| 05:04.28 | dlynes_laptop | and then have your asterisk box point to vonage |
| 05:05.10 | dlynes_laptop | if you can do that |
| 05:05.10 | dlynes_laptop | then you don't need the sipura 3000 |
| 05:05.36 | axisys | dlynes_laptop: no way.. r u serious? |
| 05:05.44 | dlynes_laptop | of course i'm serious |
| 05:05.52 | axisys | i though i must need a sipura 3000.. i am such a dump.. |
| 05:06.00 | axisys | i do have wen access to the mot ata |
| 05:06.07 | axisys | http://192.168.102.1 |
| 05:06.09 | axisys | cool |
| 05:06.10 | dlynes_laptop | i would try setting the asterisk box up to use your vonage account though |
| 05:06.19 | dlynes_laptop | and disconnect your motorola ata while you try it |
| 05:06.28 | dlynes_laptop | see if you can register and send calls to vonage through asterisk |
| 05:06.43 | dlynes_laptop | i.e. set up a softphone to register with asterisk and then send calls through the softphone |
| 05:06.48 | dlynes_laptop | if that works |
| 05:06.55 | dlynes_laptop | then you know you can reprogram the motorola |
| 05:06.57 | axisys | wait so i can have a linus server, my asterisk server, point to vonage and do all the magic? |
| 05:07.03 | dlynes_laptop | otherwise, don't touch the motorola |
| 05:07.14 | dlynes_laptop | correct |
| 05:07.25 | axisys | linux* |
| 05:08.20 | axisys | dlynes_laptop: what is a good softphone? i just installed asterisk.. apt-get install asterisk.. done |
| 05:08.28 | axisys | on this laptop |
| 05:08.35 | axisys | ubuntu edgy eft |
| 05:08.46 | *** join/#asterisk edwar64896 (n=medwards@72.83.233.220.exetel.com.au) |
| 05:09.16 | dlynes_laptop | axisys: are you using linux or windows for your desktop? |
| 05:09.44 | axisys | linux.. have not use windows for more than a yr.. and never look back.. never will :-) |
| 05:10.56 | axisys | this laptop linux, the other one mac os x.. third one is openbsd.. but need to change the fan on the hp netserver.. it was a from junk.. works pretty well except the noise |
| 05:11.07 | axisys | other one = mac mini |
| 05:11.28 | axisys | openbsd on hp netserver |
| 05:11.47 | dlynes_laptop | go with eKiga then: http://www.gnomemeeting.org/ |
| 05:12.29 | axisys | i think i already installed it -> sudo apt-get install ekiga |
| 05:12.34 | axisys | let me check |
| 05:12.57 | axisys | yep i have it |
| 05:13.04 | dlynes_laptop | there ya go then |
| 05:13.25 | axisys | so i asterisk.. just call astersik to start? |
| 05:14.26 | axisys | can u help me to get the asterisk start and then have ekiga to register w/ my vonage account? |
| 05:14.51 | dlynes_laptop | nope |
| 05:14.55 | dlynes_laptop | never used ekiga before |
| 05:15.32 | axisys | well i can just start asterisk by calling it correct? |
| 05:15.33 | dlynes_laptop | axisys: /join #ekiga, and then read the topic on how to get to the official irc support channel for ekiga |
| 05:15.43 | vt | the only one that works for me is KPhone |
| 05:15.45 | dlynes_laptop | axisys: yes, but you need to configure your sip channels and your dial plan first |
| 05:16.25 | axisys | dlynes_laptop: hmm.. that is where i need help.. any quick startup.. the manuals are huge. or may be u can help w/ that part?! |
| 05:16.50 | vt | but still, I would like to run linphone but it mess things up with my soundcard |
| 05:17.43 | dlynes_laptop | axisys: gimme a few minutes and i'll give you a stripped down extensions.conf and sip.conf file |
| 05:18.15 | vt | I cannot run kphone without esd, else I got a very baryton voice (41khz sample played as 8khz I think) |
| 05:18.15 | axisys | dlynes_laptop: appreciate it |
| 05:24.47 | ManxPower | idefsk seems interesting. I've not had a chance to play with it much. It does have the most horrible name I've seen in a long time. |
| 05:25.27 | justinu|laptop | idefisk is a great piece of software |
| 05:26.00 | ManxPower | justinu|laptop: what is so great about it over other softphones (other than the fact it uses IAX)? |
| 05:26.38 | justinu|laptop | it's a very simple, reliable IAX softphone |
| 05:26.41 | justinu|laptop | that's it |
| 05:26.58 | justinu|laptop | with a bizzare name |
| 05:27.06 | ManxPower | I did notice it the UI does not try to be some gawdawlful work of art. |
| 05:27.30 | justinu|laptop | great for testing, and troubleshooting |
| 05:28.27 | justinu|laptop | IME, it's just one of those rare pieces of software you seem to have zero issues with |
| 05:29.47 | FuriousGeorge | anyone know if the grandstream video phone's 3 way conferencing will also split the video feed if all parties are other video phones? |
| 05:30.01 | FuriousGeorge | ~s/display/feed |
| 05:30.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:30.27 | justinu|laptop | man... grandstream and videoconferencing... two words I don't want to see in the same sentence. |
| 05:30.40 | FuriousGeorge | justinu|laptop: im open for suggestions |
| 05:30.57 | justinu|laptop | no idea, i just can't imagine a grandstream would be any good at it... their voip phones have some issues |
| 05:31.23 | FuriousGeorge | someone asked me about video conferencing and asterisk looks less and less like a solution |
| 05:31.43 | justinu|laptop | do your homework... |
| 05:31.56 | FuriousGeorge | justinu|laptop: what you think all my info comes from #asterisk |
| 05:32.06 | justinu|laptop | nope ;) |
| 05:32.30 | FuriousGeorge | it actually comes from voip-info |
| 05:32.34 | justinu|laptop | heh |
| 05:32.56 | FuriousGeorge | bbl |
| 05:36.59 | dlynes_laptop | axisys: try this pastebin for a stripped down sample sip.conf and extensions.conf with a little bit of commenting |
| 05:37.02 | dlynes_laptop | axisys: http://pastebin.ca/253873 |
| 05:37.51 | axisys | dlynes_laptop: cool |
| 05:44.32 | axisys | dlynes_laptop: how do start asterisk? |
| 05:45.12 | trelane | ~book |
| 05:45.13 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:45.20 | trelane | axisys, go read it |
| 05:45.50 | dlynes_laptop | axisys: what trelane said |
| 05:46.12 | trelane | dlynes_laptop, if they get more ignorant I'm going to have to get into toll fraud |
| 05:46.14 | trelane | I could make a killing |
| 05:47.24 | axisys | trelane: heh |
| 05:47.32 | axisys | i bought the orielly book |
| 05:47.43 | axisys | i think i can just call asterisk to start |
| 05:48.07 | axisys | (have not open the book yet *sigh* and yes lazy ass) |
| 05:48.07 | trelane | good for you |
| 05:48.17 | axisys | trelane: that will change |
| 05:48.22 | trelane | axisys, asterisk is extremely powerful, and I cannot emphasize this enough |
| 05:48.25 | trelane | extremely dangerous |
| 05:48.36 | trelane | anything that hooks up to the PSTN with the flexibility that Asterisk has is |
| 05:48.55 | trelane | and if you want to talk about some no-sense-of-humor having government types, try the FCC |
| 05:56.26 | j0 | does asterisk hold music have static and fade outs built into it? or is this a common problem with an easy fix? |
| 05:56.28 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 05:57.07 | dlynes_laptop | j0: that's just a j0 problem |
| 05:57.19 | j0 | well darn, i was hoping it was one of the above :) |
| 05:57.31 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 05:58.24 | dlynes_laptop | j0: no such luck |
| 05:58.41 | dlynes_laptop | j0: i've never had static or fade outs on moh |
| 05:58.42 | *** join/#asterisk bulatitoy (n=rmn@adsl-70-231-146-111.dsl.snfc21.sbcglobal.net) |
| 05:59.34 | j0 | could it be silence detection? hmm |
| 06:00.20 | bulatitoy | need help, again. i've connected my regular phone line to my asterisk box, i can call using a regular telephone |
| 06:00.57 | bulatitoy | I also have a wireless network at home, can i use those wireless ip phone to call outside? |
| 06:01.10 | j0 | bulatitoy: yes |
| 06:01.44 | bulatitoy | j0: even if i do not have a voip service? |
| 06:01.54 | bulatitoy | i am just using the regular pstn line |
| 06:02.48 | j0 | well your wifi phone will connect to the same part of asterisk your regular phone is |
| 06:03.01 | j0 | the start and end of a call dont matter if asterisk is in the middle |
| 06:03.19 | j0 | * will translate |
| 06:03.37 | bulatitoy | ok..so basically, the ip phone will just use my network to connect to asterisk |
| 06:03.41 | j0 | yes |
| 06:03.45 | bulatitoy | then from asterisk to outside line |
| 06:04.12 | j0 | well now what do u need us for, go buy a sweet highly expensive wifi phone.. whee! |
| 06:04.31 | FuriousGeorge | meetme doesnt work with video clients right |
| 06:04.40 | dlynes_laptop | nah..get one of those spread spectrum cordless voip phones |
| 06:04.53 | dlynes_laptop | those'll get you much better range |
| 06:04.54 | bulatitoy | :) no somebody just lend me the unit to test |
| 06:04.56 | j0 | spread spectrum.. not we're talking fancy |
| 06:05.31 | j0 | thats just a fancy way of saying "uses all the bandwidth killing everything else" |
| 06:05.59 | bulatitoy | i can see that these units are really expensive |
| 06:06.19 | bulatitoy | im ok with the softphones |
| 06:06.20 | *** join/#asterisk jeffgus (n=jeffgus@marlene.zimage.com) |
| 06:06.41 | j0 | bulatitoy: if u want an ip phone to play with.. a grandstream gxp2000 is a good way to get all the features on the cheap |
| 06:06.53 | j0 | probably set you back $100 |
| 06:07.03 | j0 | less on ebay |
| 06:08.51 | bulatitoy | really? wow...i'll save some more to afford that |
| 06:09.15 | bulatitoy | i really want to test and play with it |
| 06:10.48 | bulatitoy | anybody tried the linksys ip phone? |
| 06:11.45 | hads | The Linksys stuff is OK, higher quality than the GXP2000 but you can't to BLF with them. |
| 06:12.07 | hads | I'd have a Linksys over a GXP2000 anyday though. |
| 06:13.48 | j0 | yeah.. gxp2000 is buggy |
| 06:13.53 | j0 | but it works for playing |
| 06:14.24 | bulatitoy | if you are going to use one for a small business, would you recommend the linksys? |
| 06:16.25 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.10.39.revip2.asianet.co.th) |
| 06:16.25 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:17.44 | FuriousGeorge | speaking of gxp2000 |
| 06:17.50 | j0 | i'd get a polycom 501 or 601 |
| 06:17.52 | FuriousGeorge | anyone try the 3000 video phone model |
| 06:18.23 | HaMYaI | anyone ever compiled the gastman uisng MinGW? |
| 06:19.12 | j0 | FuriousGeorge: they look fun eh? |
| 06:19.31 | j0 | i'd like one of those upright cisco ones.. the ones on 24 and ncix |
| 06:19.49 | hads | Aparently they work fine, same build quality as the others. |
| 06:20.20 | hads | Cisco don't do a video phone do they? |
| 06:20.25 | j0 | i'm not impressed with my gxp2000... i accidentally hang up on plenty of calls because it wont go on hold properly, or switch between speaker/headset |
| 06:20.28 | j0 | i could be wrong |
| 06:20.57 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 06:22.48 | [TK]D-Fender | Polycom beats pretty much everyone on quality and functionality |
| 06:22.57 | j0 | mm and arn't they sexy |
| 06:23.11 | [TK]D-Fender | j0 : Cisco beats them there, but fail on usability |
| 06:23.16 | j0 | i've heard good things about snom though |
| 06:23.44 | HaMYaI | [TK]D-Fender: have you ever compiled gastman using MinGW? |
| 06:23.49 | [TK]D-Fender | Snom has had a history of flakey hardware, only passable audio quality, and seriously shitty usage of LCD |
| 06:23.55 | [TK]D-Fender | HaMYaI : nope |
| 06:24.07 | j0 | scratch that thought then |
| 06:24.33 | HaMYaI | [TK]D-Fender: man, the package is there to download but I don't have any idea how one could get it compiles |
| 06:24.50 | HaMYaI | err compiled |
| 06:25.25 | HaMYaI | it's a very useful prog though |
| 06:25.51 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 06:26.41 | [TK]D-Fender | Linksys is decent enough quality, but again fails on LCD usability, a little less on audio quality, but in North America simply "isn't worth it". |
| 06:27.04 | hads | [TK]D-Fender: You are harsh on snoms. The build quality isn't quite up with Polycom/Cisco, the audio quality is fine IMHO and the LCD, yeah, isn't that great. |
| 06:27.56 | FuriousGeorge | i like my snom but the lcd usage leaves something to be desired |
| 06:28.26 | FuriousGeorge | [TK]D-Fender: if someone asked you to install asterisk for three way video conferencing what would you say to them? |
| 06:28.41 | hads | As far as the Linksys and LCD, I agree, it will be interesting to see the SPA962 when it comes out - as long as it's price isn't ridiculous. |
| 06:28.59 | [TK]D-Fender | hads : I wouldn't like to sound harsh, its jsut too many strikes with Polycom and the Aastra 480i out there to consider. |
| 06:29.23 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com) |
| 06:29.34 | [TK]D-Fender | hads : Just tally up the points and Snom just rarely fits (ask yourself where it costs so much less than the rest to vie for a place) |
| 06:30.10 | *** join/#asterisk Op3r (n=op3r@121.97.215.225) |
| 06:30.16 | Op3r | anyone up? |
| 06:30.18 | [TK]D-Fender | hads : Problem with Linksys is no support for basic presence, and poor call control (limited to 1 call/key),a nd you have to pay to get maximal use of them. |
| 06:30.30 | Op3r | i get this very very freaky problem when installing zaptel |
| 06:30.32 | [TK]D-Fender | hads : it does LOOK nice, I just doubt it'll be worth the $ or trouble. |
| 06:30.47 | Op3r | make -C SUBDIRS=/usr/src/zaptel-1.2.9.1 clean |
| 06:30.47 | Op3r | make: *** SUBDIRS=/usr/src/zaptel-1.2.9.1: No such file or directory. Stop. |
| 06:30.47 | Op3r | make: *** [clean] Error 2 |
| 06:31.12 | Op3r | any one? |
| 06:31.38 | hads | [TK]D-Fender: Yeah, I agree on the points about the Linksys (I'm so agreeable tonite). |
| 06:33.47 | [TK]D-Fender | hads : Well I do try to be fair and everything has its place based on circumstances. Price does matter. Thats a large equalizer. Cisco just plain isn't worth it almost ANYWHERE :) |
| 06:34.10 | orlock | [TK]D-Fender: except for TV and government epartments |
| 06:34.12 | hads | I couldn't agree more on that :) |
| 06:34.14 | orlock | departments |
| 06:34.59 | FuriousGeorge | my problem is, since asterisk doesnt properly support video sip, i need to depend on a client to make the three way conference happen. grnadstream says they do 3-way conference but i dont see anywhere where it says it splits the video display |
| 06:36.37 | FuriousGeorge | so i was thinking maybe * wasnt the best solution, but ip video conferencing in a box is still VERY expensive |
| 06:38.44 | FuriousGeorge | also the dude wants to display it on a plasma, and i have a feeling the analog out on the grandstream is gonna look like crap at 50" as its gonna probably try to stretch it to take up the whole screen |
| 06:39.10 | [TK]D-Fender | FuriousGeorge : QCIF <- |
| 06:39.23 | FuriousGeorge | quails can't infiltrate france? |
| 06:39.33 | Op3r | lol |
| 06:39.34 | [TK]D-Fender | FuriousGeorge : Its all about the resolution. If you want to video conference, use eyebeam anyways |
| 06:39.58 | j0 | haha |
| 06:40.17 | [TK]D-Fender | ok, checkout time for me. |
| 06:40.25 | [TK]D-Fender | Night all, back (maybe) tomorrow |
| 06:40.28 | FuriousGeorge | [TK]D-Fender: yeah, i just thought he would be less than impressed with a softclient. how come everyone else knows what that stands for but me? |
| 06:40.37 | FuriousGeorge | besides eyebeam uses an older video codec, doesnt it? |
| 06:40.42 | FuriousGeorge | i was looking at wengo |
| 06:43.05 | [TK]D-Fender | FuriousGeorge : Its a question of resolution. Most formats are too low to blow up decent anyways. Remember that most videoconf stuff using boring TV's whose resolution if 520 by 400 or so. And VCR is HALF of that. |
| 06:43.24 | [TK]D-Fender | Do your research. And you'llwant H.264 (eyebeam supports) for video. |
| 06:43.32 | [TK]D-Fender | Anyways, I'm outta here. |
| 06:44.12 | FuriousGeorge | yeah i heard h.264 was da bomb, i thought eyebeam didnt support it |
| 06:45.47 | Op3r | beta3 was released |
| 06:51.16 | FuriousGeorge | theyre gonna do RCs after the betas right |
| 06:54.02 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:54.08 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 06:57.52 | TripleFFFF | op3 eyebeam beta3 ? |
| 06:57.54 | TripleFFFF | u have ? |
| 07:00.15 | *** join/#asterisk LoneShadow (n=duh@c-67-188-235-220.hsd1.ca.comcast.net) |
| 07:05.26 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 07:09.42 | *** join/#asterisk AJaymn (n=me@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 07:10.39 | AJaymn | So.. what is everyone doing about E911? I have users all over the US.. there an easy way to get E911 working? |
| 07:10.46 | FuriousGeorge | linux # svn ls http://svn.digium.com/svn/asterisk/tags/ |
| 07:10.46 | FuriousGeorge | svn: PROPFIND request failed on '/svn/asterisk/tags' |
| 07:10.46 | FuriousGeorge | svn: PROPFIND of '/svn/asterisk/tags': could not connect to server (http://svn.digium.com) |
| 07:10.49 | FuriousGeorge | who broke svn? |
| 07:11.10 | *** join/#asterisk appelza (n=pieter@196.40.106.74) |
| 07:11.29 | appelza | Hi, what command can I type into asterisks' console to show me all the extentions? |
| 07:11.45 | FuriousGeorge | show dialplan |
| 07:12.24 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:12.53 | appelza | ty |
| 07:14.02 | FuriousGeorge | np |
| 07:17.12 | *** join/#asterisk ltd (n=z@202-161-28-21.dyn.iinet.net.au) |
| 07:17.37 | FuriousGeorge | something is going on behind the scenes at asterisk.org |
| 07:17.47 | FuriousGeorge | i think its on purpose to mess with me |
| 07:21.40 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-753a1079e6849923) |
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| 07:27.13 | bulatitoy | need to go guys, thanks for all your help |
| 07:27.25 | *** part/#asterisk bulatitoy (n=rmn@adsl-70-231-146-111.dsl.snfc21.sbcglobal.net) |
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| 07:28.49 | infinity1 | were can i get hold music mp3s? |
| 07:34.14 | *** join/#asterisk shellshark (n=x86@get.hooked.on.voip.with.shellshark.net) |
| 07:37.51 | *** join/#asterisk Laggy_McGee (n=jchadwic@pool-71-245-122-170.cmdnnj.fios.verizon.net) |
| 07:38.16 | Laggy_McGee | I've been searching for hours -- how do I make the stuttered dialtone when messages are waiting? |
| 07:41.42 | infinity1 | for zap interfaces? |
| 07:43.34 | shellshark | Laggy_McGee: good question, when you find out, let me know :) |
| 07:43.58 | shellshark | Laggy_McGee: i think the end device has to support it (ATA, IP Phone, etc) |
| 07:44.12 | shellshark | Laggy_McGee: like my grandstream bt101 does it automatically |
| 07:48.00 | *** join/#asterisk puthre (n=puthre@80.97.12.10) |
| 07:52.21 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
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| 08:02.38 | Laggy_McGee | shellshark: ahh... makes sense somewhat. |
| 08:02.49 | Laggy_McGee | I've got a Sipura ATA286 |
| 08:04.45 | hads | Huh? |
| 08:04.53 | hads | Sipura ATA286? |
| 08:04.55 | HaMYaI | anyone got gastman compiled under MinGW successfully? |
| 08:06.06 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 08:07.02 | ontae | hi, need help with [context], someone has a little bit time? |
| 08:09.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.136) |
| 08:11.56 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 08:29.44 | *** join/#asterisk matsavhalev (n=marc@212.150.97.22) |
| 08:30.41 | matsavhalev | can anyone help me with a DID number problem? Asterisk doesn't seem to diffrentiate between calls from different incoming DID numbers instead passing them all to a single extension... |
| 08:36.38 | dlynes_laptop | Laggy_McGee: you mean a grandstream ata-286? |
| 08:37.36 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
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| 08:46.09 | *** part/#asterisk matsavhalev (n=marc@212.150.97.22) |
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| 09:03.11 | HaMYaI | dlynes_laptop: have you ever compiled gastman using MinGW? |
| 09:04.09 | dlynes_laptop | nope |
| 09:04.09 | dlynes_laptop | Never had a need to |
| 09:04.16 | dlynes_laptop | gastman compiles easily on linux |
| 09:05.51 | HaMYaI | dlynes_laptop: most of my PCs are windows XP based |
| 09:06.23 | dlynes_laptop | ic |
| 09:07.17 | HaMYaI | dlynes_laptop: but there's an app called VNC that can connect to XWindows |
| 09:07.37 | dlynes_laptop | HaMYaI: what for? |
| 09:07.39 | HaMYaI | dlynes_laptop: I will probably compile it under linux instead |
| 09:07.47 | dlynes_laptop | HaMYaI: you can get x11 clients for windows |
| 09:07.53 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:08.05 | dlynes_laptop | HaMYaI: and then use x11 forwarding through putty |
| 09:08.10 | HaMYaI | dlynes_laptop: never used that =) |
| 09:08.21 | HaMYaI | dlynes_laptop: will it be easier? |
| 09:08.22 | dlynes_laptop | HaMYaI: then you can have an encrypted tunnel |
| 09:08.28 | dlynes_laptop | HaMYaI: vnc sucks...no encryption |
| 09:08.30 | HaMYaI | dlynes_laptop: ic |
| 09:09.51 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 09:20.41 | *** join/#asterisk AJaymn (i=TJBoi@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 09:21.55 | AJaymn | im lossing inbound callerid from Broadvoice. someone suggested doing "setcallerid(CALLERID)" but that doesnt work? Am i doing that wrong? |
| 09:22.25 | AJaymn | i know the info in coming into me.. the callerid NAme is right, but the # is being replaced with my VoIP # and not the callers |
| 09:22.35 | hads | Set(CALLERID(num)=12345) |
| 09:23.24 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 09:23.59 | AJaymn | hads But I dont know what the number is.. I need it to pull it from the header from Broadvoice ??? |
| 09:25.28 | hads | No idea sorry, I know nothing about broadvoice, or your setup. |
| 09:28.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.136) |
| 09:29.13 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 09:30.06 | AJaymn | whats the command id type to see the Header of a call coming in? |
| 09:32.13 | dlynes_laptop | sip show channel channelname |
| 09:32.28 | dlynes_laptop | If it's an incoming sip call |
| 09:32.43 | dlynes_laptop | If it's something else, it'll be something similar, if the channel supports it |
| 09:37.46 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-91-240.w80-8.abo.wanadoo.fr) |
| 09:41.23 | *** join/#asterisk FTexcom (n=info@14.Red-80-26-4.staticIP.rima-tde.net) |
| 09:47.00 | FuriousGeorge | file: that weird issue ive been having where certain calls transferred into meetmes by certain clients would result in one party's audio being 'lost': solved it with upgrade |
| 09:47.08 | FuriousGeorge | thought you might be interested to know |
| 09:56.03 | *** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net) |
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| 10:07.16 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 10:08.24 | AJaymn | Anyone know of less expensive way to offer E911 to users? |
| 10:08.47 | ontae | hi, having problems with [context]: When setting for UA "A" context=A in sip.conf, the extension for an outgoing call from UA "A" is looked up in context [default] in extensions.conf ! Any suggestions what maybe I am doing wrong or what is going wrong ? |
| 10:09.07 | boddy | hii all I have problem sip clients connect to * over internet.and * connect meridian 1c via E1 When they call me or I call them(I am on meridian) they hear me but I cant hear anything |
| 10:09.13 | boddy | have you any idea ? |
| 10:09.48 | monsted | ~thebook |
| 10:09.59 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:10.00 | monsted | guess not :) |
| 10:11.33 | boddy | jbot:you say to me ? |
| 10:12.04 | ontae | @boddy: i had the same problem two times: For me it was a NAT and ROUTE issue. |
| 10:14.25 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:14.54 | MrChimpy | anyone using the 1.4 gui stuff? |
| 10:15.05 | MrChimpy | how advanced is the gui for conference management? |
| 10:16.20 | reza_ | anyone know what this error might mean : |
| 10:16.38 | reza_ | chan_sip.c:1084 __sip_xmit: sip_xmit of 0x84a6d88 (len 699) to 10.0.0.181:5060 returned -1: Bad file descriptor |
| 10:28.21 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-149-4.red.bezeqint.net) |
| 10:28.33 | ontae | Can anyone acknowledge this: UA A with context=A in sip.conf searches for extensions in section [A] in extensions.conf ! |
| 10:30.24 | kaldemar | ontae: ACK |
| 10:33.57 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 10:34.02 | ontae | kaldemar: O.K., it works for my "Cisco ATA 186 v3.2.1 atasip (050616A)" but not for my "Grandstream BT110 1.0.8.23" or "eyeBeam release 3004t stamp 16741" ; any suggestions ? |
| 10:38.22 | kaldemar | ontae: does your sip.conf have [user]-contexts that match the user names you have defined for the BT or the eyeBeam? |
| 10:39.51 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 10:45.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:46.39 | Zeeek | don't all talk at once |
| 10:47.42 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 10:47.59 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-15-130.w81-248.abo.wanadoo.fr) |
| 10:49.18 | ontae | kaldemar: no, eg.: [BT] type=friend ; username=BT ; context=A |
| 10:49.29 | *** join/#asterisk Dr-Linux|work (n=Nothing@202.125.139.198) |
| 10:49.48 | Dr-Linux|work | a question, for what reason we use 5004 port in asterisk? |
| 10:50.56 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 10:58.41 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 10:58.42 | *** join/#asterisk XIN01OZ (n=askme@c-68-63-34-189.hsd1.al.comcast.net) |
| 10:59.24 | XIN01OZ | Anyone in the southeast intrested in a partnership? |
| 11:00.22 | XIN01OZ | I need help but have good direction with great profits in return.. |
| 11:01.19 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:02.56 | dlynes_laptop | Dr-Linux: cause it's not reserved by anything else? |
| 11:12.47 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 11:14.25 | *** join/#asterisk henrique (n=henrique@200-168-128-186.dsl.telesp.net.br) |
| 11:15.39 | k31th | anyone using the Linksys SPA942 ?? care to tell me wher ei add an account :s |
| 11:16.26 | k31th | found it |
| 11:27.35 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 11:27.39 | kippi | hey |
| 11:28.10 | Dr-Linux|work | a question, for what reason we use 5004 port in asterisk? |
| 11:28.18 | dlynes_laptop | Dr-Linux: cause it's not reserved by anything else? |
| 11:28.35 | Zeeek | k31th I have a 941 |
| 11:28.43 | dlynes_laptop | hotdamn |
| 11:28.47 | Dr-Linux|work | dlynes_laptop: i didn't get ya ? |
| 11:28.59 | dlynes_laptop | I guess I shouldn't have tried doing an upgrade all under Solaris :( |
| 11:29.01 | Zeeek | k31th use the web interface |
| 11:29.13 | dlynes_laptop | upgrading like 50 million packages :( |
| 11:29.48 | dlynes_laptop | Dr-Linux: they had to pick a port number that wasn't already reserved by another daemon |
| 11:30.02 | dlynes_laptop | Dr-Linux: so that when they do get an official port number, they don't have to worry about collisions |
| 11:30.22 | kippi | I would like to know how easy it would be to have a * code that when you dial it it goes to a voicemail box and then when you dial it again it will send all alls to the que |
| 11:31.07 | dlynes_laptop | kippi: not that hard |
| 11:31.08 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 11:31.21 | dlynes_laptop | kippi: just use astdb to keep track of the state, and branch, depending on state |
| 11:33.38 | kippi | anyone know of an example of what I am going to try and do |
| 11:34.02 | dlynes_laptop | MrChimpy: nice |
| 11:35.00 | RoyK | http://karlsbakk.net/fun/tannkost.jpg |
| 11:39.17 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 11:42.55 | *** join/#asterisk ltd (n=z@202-161-18-50.dyn.iinet.net.au) |
| 11:43.34 | dlynes_laptop | cool |
| 11:43.41 | dlynes_laptop | package update's finally finished :) |
| 11:53.21 | *** join/#asterisk phpboy (n=shane@196.211.17.202) |
| 11:53.47 | phpboy | hey all... i've managed to figure out how to record calls but how do I combine the in and out file? |
| 11:53.56 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:54.53 | HarryR | phpboy, use something like sox... or even something like mixmonitor :) |
| 11:55.15 | HarryR | see: http://www.voip-info.org/wiki/view/MixMonitor |
| 11:56.00 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:58.06 | dlynes_laptop | phpboy: is this for a call center? |
| 11:58.08 | Zeeek | heynow (tm) |
| 11:58.25 | dlynes_laptop | phpboy: i.e. a high volume phone system, where call recording will get called multiple times? |
| 12:00.16 | *** join/#asterisk ontae (n=ontae@clnet-p03-090.ikbnet.co.at) |
| 12:01.58 | mosty | what am i doing wrong here? -- Executing GotoIf("SIP/101-0818e430", ""false" = "true"?5") in new stack -- Goto (office-hours,s,5) |
| 12:02.27 | mosty | it looks like it should NOT goto priority 5, but it is, i can't see why |
| 12:04.00 | HarryR | they still produce vinyl for non-dj stuff!? |
| 12:04.27 | mosty | dj's mix almost anything |
| 12:04.30 | RoyK | this one came with a good old record :) |
| 12:05.15 | dlynes_laptop | mosty: what priority is that gotoif in? |
| 12:05.48 | HarryR | mosty, yeah but you don't get the same quality with cds compared to vinyl.. it sounds flat and slightly less interesting unless you've got an uber-studio-god to master it for you :) |
| 12:06.01 | mosty | dlynes_laptop: the GotoIf is in priority 2, priority 3 is a Dial comand |
| 12:06.28 | dlynes_laptop | mosty: i would think it's probably something in your gotoif command |
| 12:06.42 | dlynes_laptop | mosty: Can you paste yoru gotoif exact code into the channel? |
| 12:06.50 | mosty | exten => s,2,GotoIf("${looped}" = "true"?5) |
| 12:07.10 | mosty | and in this case, ${looped} is false |
| 12:07.49 | dlynes_laptop | mosty: exten => s,2,GotoIf(["${looped}"="true"]?5) |
| 12:08.17 | dlynes_laptop | mosty: understand the difference? |
| 12:08.44 | mosty | nope |
| 12:09.00 | JT | dlynes_laptop: why do you ask, re volume of recording? |
| 12:09.07 | dlynes_laptop | mosty: your evaluation expression must be enclosed in square brackets |
| 12:09.13 | dlynes_laptop | JT: huh? |
| 12:09.43 | mosty | dlynes_laptop: looking at the wiki now, it says that an expression needs to be inside $[ ] to be evaluated |
| 12:09.43 | JT | you asked phpboy if it was a high volume recording scenario |
| 12:09.45 | JT | i'm curious |
| 12:10.46 | dlynes_laptop | JT: oh, because mixmonitor usually segfaults after about 3000 iterations or so |
| 12:10.51 | JT | nice |
| 12:10.55 | dlynes_laptop | JT: that's what I've heard on here from several people trying to use it in a call center |
| 12:11.02 | dlynes_laptop | JT: they've given up on it |
| 12:11.18 | JT | as in 3000 concurrent calls, or total calls in an asterisk process? |
| 12:11.30 | dlynes_laptop | mosty: yeah...i forgot about the enclosing $ |
| 12:11.44 | dlynes_laptop | mosty: erm outer $ |
| 12:12.05 | dlynes_laptop | JT: total executions of mixmonitor within the current asterisk invocation |
| 12:12.13 | JT | hrm |
| 12:12.20 | JT | do the other monitor apps have this prob? |
| 12:12.23 | JT | or only mixmonitor |
| 12:12.26 | dlynes_laptop | no...only mixmonitor |
| 12:12.34 | JT | that's goos i guess |
| 12:12.36 | JT | good |
| 12:12.53 | dlynes_laptop | they've gone back to using monitor and then using sox(?) to do a manual merge |
| 12:12.56 | JT | i'm inclined to record both seperate streams |
| 12:13.12 | JT | and then use an external app to make a third stream, but store the original too |
| 12:13.18 | JT | hdd killer i know |
| 12:13.23 | dlynes_laptop | JT: even then, sometimes mixmonitor doesn't properly merge it |
| 12:13.25 | JT | but it's good for audio forensics |
| 12:13.33 | dlynes_laptop | JT: it does about 95% of the time, though |
| 12:13.36 | JT | you can hear clearly what's happening on each side |
| 12:19.25 | *** join/#asterisk ltd (n=z@202-161-28-21.dyn.iinet.net.au) |
| 12:26.22 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:28.33 | *** join/#asterisk nafigator (n=IM@84.22.192.250) |
| 12:30.38 | bXi | how does one set the default expire time for iax clients ? |
| 12:30.48 | *** join/#asterisk VijayG (i=vijay@202.131.145.234) |
| 12:30.50 | VijayG | hello |
| 12:31.01 | VijayG | whats the command the stop asterisk |
| 12:31.09 | VijayG | stop now, from asterisk cli is not working |
| 12:31.16 | VijayG | whats the command i can use in linux |
| 12:31.17 | bXi | why isnt it working? |
| 12:31.19 | VijayG | like kill-a |
| 12:31.26 | VijayG | it has hanged it seems |
| 12:31.53 | monsted | man killall |
| 12:32.08 | bXi | killall -9 asterisk should do the trick |
| 12:32.15 | JT | killall is unnecessary |
| 12:32.20 | JT | use ps aux |
| 12:32.22 | JT | and kill |
| 12:32.38 | VijayG | what would the command |
| 12:32.41 | VijayG | ps aux |
| 12:32.44 | VijayG | and kill asterisk |
| 12:32.45 | bXi | it achieves the same result so why do it the hard way |
| 12:32.48 | VijayG | will this do? |
| 12:32.50 | JT | list current processes |
| 12:32.58 | JT | kill <pidnumber> |
| 12:33.21 | JT | bXi: it's a poor habit, killall should only be used after kill fails |
| 12:33.24 | JT | and then with care |
| 12:33.34 | JT | as you can kill more than necessary if you use it carelessly |
| 12:33.45 | VijayG | how can i find the job Id for asterisk |
| 12:33.50 | JT | ps aux |
| 12:34.37 | bXi | Nov 21 13:34:17 NOTICE[20071] chan_iax2.c: Restricting registration for peer '1437337' to 60 seconds (requested 3600) |
| 12:34.41 | VijayG | ok |
| 12:34.50 | bXi | any idea on how to get that 60 seconds up to 3600 ? |
| 12:35.22 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:35.48 | monsted | killall is especially evil because the linux one does things much different from most of the commercial Unix flavors |
| 12:36.20 | JT | don't the commercia unix ones kill even more? |
| 12:36.30 | VijayG | kill 5698 is not working |
| 12:36.31 | RoyK | monsted: what do you mean? |
| 12:36.37 | VijayG | the process is still running |
| 12:36.39 | RoyK | VijayG: kill -9 |
| 12:36.43 | JT | VijayG: kill -9 5698 |
| 12:36.50 | JT | use kill -9 is normal kill fails |
| 12:36.54 | RoyK | kill -9 -1 |
| 12:36.55 | RoyK | :) |
| 12:37.01 | JT | s/is/if/ |
| 12:37.03 | Zeeek | kill -9 RoyK |
| 12:37.15 | RoyK | hehe |
| 12:37.23 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 12:37.29 | VijayG | thanks guys |
| 12:37.32 | Zeeek | sudo rm -r RoyK |
| 12:37.33 | VijayG | kill -9 worked |
| 12:37.43 | Zeeek | UNSUBSCRIBE RoyK |
| 12:38.02 | Zeeek | s /R/RoyK/g |
| 12:38.07 | RoyK | Zeeek: have you forgotten to take your medicine today, sir? |
| 12:38.17 | Zeeek | no, I did take it,n that's why |
| 12:38.19 | monsted | RoyK: killall -9 blah on linux kills all "blah" processes, but on solaris kills everything |
| 12:38.32 | Zeeek | poor zeedo didn't do anything |
| 12:38.54 | RoyK | monsted: that's the old one, called killall5 in linux |
| 12:38.58 | RoyK | system 5 killall |
| 12:39.08 | JT | monsted: that's what i thought |
| 12:39.10 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 12:39.38 | Zeeek | there should be a killRoyK |
| 12:39.39 | *** join/#asterisk coppice (n=chatzill@213.143.17.210.dyn.pacific.net.hk) |
| 12:39.50 | Zeeek | there prolly is for debian, but it's a pkg |
| 12:39.59 | JT | * cpm feels strangely happy, seeing the violence before him |
| 12:40.00 | Zeeek | cpm? SHould be dead by now |
| 12:41.21 | Zeeek | you want folders? |
| 12:42.39 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 12:42.47 | qdk | any words on the Grandstream GXP-2000 IP-phone? good, bad, avg., a joke? |
| 12:42.58 | Zeeek | the word on the street is... |
| 12:43.01 | Zeeek | Yo! |
| 12:43.23 | JT | qdk: not that good aparently |
| 12:43.35 | JT | better than the budgetone, but not better than much else |
| 12:43.38 | Zeeek | The truth is, I've heard a lot of bad and little good |
| 12:43.58 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 12:44.02 | Zeeek | The BT100s were a good starter phone. I'd move to anoither brand after that. Sipura comes to mind |
| 12:44.28 | Zeeek | Or if it must be cheap, the chinses IAX capable phones are cool |
| 12:44.35 | qdk | JT: ok, you know of any specific problems? |
| 12:44.44 | Zeeek | we're taling like $50-$60 here |
| 12:45.05 | JT | Zeeek: got any names for cheap iax phones? |
| 12:45.06 | qdk | JT: it seems TOO feature rice to its price... |
| 12:45.24 | Zeeek | for reviews of entry-level phones, you can't beat FWD forum |
| 12:45.29 | JT | audio quality is a common one i've heard |
| 12:45.36 | bXi | Nov 21 13:34:17 NOTICE[20071] chan_iax2.c: Restricting registration for peer '1437337' to 60 seconds (requested 3600) |
| 12:45.36 | bXi | any idea on how to get that 60 seconds up to 3600 ? |
| 12:45.36 | Zeeek | JT just a s ec. What country are you in? |
| 12:45.53 | JT | au |
| 12:46.26 | qdk | any recommendation for a good (about) $100 IP-phone? |
| 12:46.30 | Zeeek | check this and other pages on the wiki: http://www.voip-info.org/wiki/view/Yuxin |
| 12:46.51 | JT | i don't think you'll find anything very close to good for $100 |
| 12:46.54 | Zeeek | I have the earlier technology phone the PA1688 |
| 12:47.09 | Zeeek | They're great, I always have one in my suitcase |
| 12:47.23 | Zeeek | they work anywhere you can get an ip ny DHCP |
| 12:47.27 | JT | as well as enriched plutonium.. |
| 12:47.47 | Zeeek | the IAX phones are way under $100 |
| 12:47.49 | qdk | JT: :-) |
| 12:47.55 | FTexcom | any good iax phone? |
| 12:47.56 | *** join/#asterisk nafigator (n=IM@84.22.192.250) |
| 12:48.28 | dlynes_laptop | FTexcom: as good as polycom, you mean? |
| 12:48.39 | Zeeek | not true |
| 12:48.53 | Zeeek | We use nufone with IAX only and it has great quality |
| 12:48.57 | Aurs | anyone using the polycom 430? |
| 12:49.12 | dlynes_laptop | FTexcom: or good as in usable, and not necessarily superb business quality? |
| 12:49.57 | cpm | qdk, I don't know why, I use iax for all my provider termination, and for a few stations, and I prefer it to sip, for those things that iax does |
| 12:50.28 | *** join/#asterisk achu (n=achu@124.125.37.223) |
| 12:51.21 | achu | I have configured astfax with qmail , no I can send tiff images as fax , but the problem is I am not gettig the actual image |
| 12:51.32 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 12:51.34 | achu | instead of the actual image its showing a blank image |
| 12:52.01 | achu | please help me if anybody have configured astfax |
| 13:00.54 | monsted | anyone using chan_skinny? I'm attempting to have a cisco 7960 register and have added stuff to skinny.conf, but "skinny list devices" doesn't show any devices and registration fails |
| 13:01.53 | jm|work | monsted: you got all your tftp stuff set up? |
| 13:04.26 | monsted | looks like it |
| 13:04.28 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 13:04.40 | monsted | the phone attemps to register, but is rejected |
| 13:04.44 | jm|work | monsted: when it's working post me a HOWTO |
| 13:04.48 | monsted | heh |
| 13:04.58 | jm|work | I installed SIP firmware on mine |
| 13:05.01 | monsted | nowai! if it's hard for me, it's supposed to be hard for you too! |
| 13:05.35 | monsted | one of my three phones has problems registering via SIP, so i thought i'd try skinny on it instead |
| 13:07.52 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-6af40b48c6a68d56) |
| 13:11.10 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 13:11.46 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.216) |
| 13:12.42 | phpboy | dlynes_laptop: it won't get called too often, no... more for the staff... |
| 13:12.48 | phpboy | it's an attorneys firm |
| 13:12.54 | phpboy | why, what do you have in mind? |
| 13:13.19 | dlynes_laptop | phpboy: nvm then...asterisk will probably crash from other stuff before you have to worry about mixmonitor, then |
| 13:13.37 | phpboy | so mix monitor is my best bet? |
| 13:13.52 | dlynes_laptop | phpboy: yeah...it'd be the easiest way to do what you want to do |
| 13:13.56 | monsted | jm|work: first caveat is the fact that chan_skinny doesn't load by default :) |
| 13:13.58 | phpboy | can I use mixmonitor and record in and out |
| 13:14.02 | phpboy | at the same time? |
| 13:14.06 | phpboy | so I have 3 files? |
| 13:14.15 | JT | that seems illogical |
| 13:14.28 | JT | just use an outside prog like soz to make the third at the ceasation of the call |
| 13:14.33 | JT | sox i mean |
| 13:14.49 | JT | i think you can spawn it from the monitor or record command |
| 13:14.56 | JT | or do it in the dialplan anyhow |
| 13:15.01 | dlynes_laptop | phpboy: you would use mixmonitor when you want both legs combined and don't care about the separate legs |
| 13:15.14 | dlynes_laptop | phpboy: if you want to keep both separate legs, you don't want mixmonitor |
| 13:15.15 | phpboy | :< |
| 13:15.26 | phpboy | dlynes_laptop: what do I want then? |
| 13:15.34 | dlynes_laptop | phpboy: you want monitor |
| 13:15.38 | JT | the non-mixing version? |
| 13:15.42 | dlynes_laptop | yeah |
| 13:15.48 | phpboy | I want separate leg's and combined legs |
| 13:15.55 | JT | yes and as i just said |
| 13:15.58 | JT | monitor |
| 13:16.01 | dlynes_laptop | phpboy: yeah, so monitor combined with what jt told you |
| 13:16.05 | JT | then make a combined one at the end of the call |
| 13:16.21 | phpboy | but I'm using monitor and it's giving me 2 separate legs |
| 13:16.22 | phpboy | :/ |
| 13:16.26 | JT | gar |
| 13:16.29 | JT | LISTEN ok |
| 13:16.35 | JT | yes make 2 seperate legs |
| 13:16.42 | JT | and use the program sox to combine them |
| 13:16.44 | JT | gar |
| 13:16.47 | phpboy | ok |
| 13:17.04 | JT | there's no reason why that approach shouldn't prove successful |
| 13:17.46 | phpboy | I agree |
| 13:17.53 | phpboy | /usr/ports/audio/sox |
| 13:17.57 | phpboy | busy installing... |
| 13:18.13 | JT | there's a bit of info about this on voip-info.org btw |
| 13:18.23 | ontae|lunch | kaldemar: you asked "does your sip.conf have [user]-contexts that match the user names you have defined for the BT or the eyeBeam?" answer: "no, eg.: [UA] type=friend ; username=UA ; context=A" |
| 13:18.26 | JT | see all the pages related to asterisk recording, monitoring, barging, etc |
| 13:18.53 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:21.13 | phpboy | I'm installing sox with mp3 encoding |
| 13:21.16 | phpboy | it's gonan help |
| 13:22.43 | JT | hmm |
| 13:22.50 | achu | Is asterfax support qmail ? |
| 13:23.13 | dlynes_laptop | achu: qmail has a symbolic link, 'sendmail', so probably |
| 13:23.33 | JT | yeah no reason in theory why it shouldn't work |
| 13:23.39 | JT | that said, eww, qmail :P |
| 13:23.48 | *** join/#asterisk ka05 (n=graeme@cpc5-asht3-0-0-cust317.manc.cable.ntl.com) |
| 13:24.04 | achu | ok |
| 13:28.38 | ka05 | Anyone seen "HFC-MULTI: Card 'HFC-4S Digium Card' found, but not given by module's options, ignoring..." When trying to load a Digium B410P? |
| 13:28.38 | achu | dlynes_laptop : Did you tried Astfax ? |
| 13:28.39 | dlynes_laptop | achu: i think i took a look at it, and said forget it |
| 13:28.39 | dlynes_laptop | achu: it was too convoluted considering how simple it should have been |
| 13:28.39 | dlynes_laptop | achu: i just wrote my own scripts instead |
| 13:28.39 | achu | dlynes_laptop : I have configured it and its almost ok but I think asterisk doesn't send fax |
| 13:29.47 | *** join/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 13:29.59 | dlynes_laptop | achu: do you have spandsp installed? |
| 13:30.14 | achu | yes |
| 13:30.15 | dlynes_laptop | achu: i.e. app_sendfx/app_recvfx? |
| 13:30.21 | achu | yeah |
| 13:30.42 | dlynes_laptop | what indicates that asterisk isn't trying to send a fax? |
| 13:32.02 | achu | I am getting 3 logs |
| 13:32.06 | achu | Channel SIP/bv-09c03a00 was an swered. |
| 13:32.20 | dlynes_laptop | eww |
| 13:32.25 | dlynes_laptop | you're trying to fax over sip? |
| 13:32.36 | achu | yes |
| 13:32.36 | *** join/#asterisk jserve (n=mail@vpngw.de.callforcenetwork.eu) |
| 13:32.37 | JT | crazy |
| 13:32.40 | jserve | Hi all |
| 13:32.41 | achu | <PROTECTED> |
| 13:32.51 | mosty | achu: fax doesnt work reliably over voip |
| 13:32.53 | JT | did you actually check to see with anyone whether that was possible? :P |
| 13:33.01 | dlynes_laptop | achu: you're a masochist |
| 13:33.08 | dlynes_laptop | achu: and you must not value your time |
| 13:34.10 | achu | hmm |
| 13:34.47 | achu | dlynes_laptop : Is this work with iax2 ? |
| 13:35.02 | dlynes_laptop | nope |
| 13:35.10 | JT | it *might* work if on a stable local network with no packet loss etc |
| 13:35.12 | dlynes_laptop | it works with iax2 and sip |
| 13:35.22 | dlynes_laptop | if you've got perfect network conditions |
| 13:35.28 | dlynes_laptop | and you're using ulaw or alaw codec |
| 13:35.38 | dlynes_laptop | but if you need to go outside your lan |
| 13:35.40 | JT | voip was designed to carry voice, not data |
| 13:35.40 | dlynes_laptop | forget it |
| 13:35.55 | mosty | achu: fax does not cope with jitter or packet loss. faxing over voip protocols is therefore at the mercy of your network quality |
| 13:36.05 | dlynes_laptop | voip has enough problems trying to carry voice, never mind fax |
| 13:36.20 | JT | you need to come up with a lateral solution |
| 13:36.25 | JT | or look at fax over ip |
| 13:36.27 | achu | can you tell me a best software for email to fax with asterisk ? |
| 13:37.04 | JT | are the faxes coming on over real physical lines? |
| 13:37.09 | JT | if so asterisk can handle that |
| 13:37.11 | *** join/#asterisk n3glv (n=N3GLV@c-71-60-125-243.hsd1.pa.comcast.net) |
| 13:37.31 | n3glv | hi guys, I have a procedure question |
| 13:37.53 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-149-4.red.bezeqint.net) |
| 13:37.55 | achu | JT : I have two connection from broadvoice and teliax to my asterisk server |
| 13:37.56 | b11d|bbl | morning lads |
| 13:38.05 | JT | achu: only sip? |
| 13:38.09 | achu | I want to use that for faxing |
| 13:38.11 | n3glv | when a call is made from asterisk, where and when does the ring signal file figure in? |
| 13:38.18 | JT | are they voip connections? |
| 13:38.21 | achu | no teliax connection is an IAX2 connection |
| 13:38.23 | achu | yes |
| 13:38.26 | mosty | achu: dont fax over voip |
| 13:38.30 | JT | they can't do it |
| 13:38.35 | JT | either get real phone lines |
| 13:38.44 | JT | or get a fax to email service from a provider |
| 13:39.09 | n3glv | in other words, I want to make distinct ringing audio files for separate trunks |
| 13:39.10 | achu | ok I will get a telephone line |
| 13:39.35 | achu | and can you please guide me which software is best for a phone line for faxing ? |
| 13:39.36 | JT | you will need to buy a TDM400P for an analogue line, if you still want to receive them with asterisk |
| 13:39.43 | JT | umm |
| 13:39.44 | mosty | achu: i like hyalafax |
| 13:39.47 | JT | name escapes me |
| 13:39.52 | JT | ah yeah, that seems popular |
| 13:40.05 | mosty | hylafax + an external serial modem, is very easy to setup |
| 13:40.06 | achu | thanks |
| 13:40.16 | JT | if it's not involving asterisk, i think yhou can use normal modems |
| 13:40.17 | JT | yeah |
| 13:40.41 | JT | do you just receive faxes |
| 13:40.43 | JT | or send too? |
| 13:40.52 | achu | but is it possible to intergrate it with qmail ? |
| 13:41.11 | achu | I need both sending and recieving it through email |
| 13:41.23 | mosty | achu: hylafax can do email to fax and fax to email. doesn't matter which mail server you use |
| 13:41.23 | JT | email integration is not usually MTA dependant on *nix |
| 13:41.33 | JT | as they all have sendmail comand interfaces |
| 13:41.45 | achu | ok |
| 13:41.55 | n3glv | standard "fax modem" should work, as long as you have real physical phone line. |
| 13:42.27 | mosty | external serial fax modems are the simplest to use |
| 13:42.33 | n3glv | yup |
| 13:42.49 | n3glv | I know where there are like 30 of them, brand new |
| 13:42.56 | n3glv | US Robotics |
| 13:43.23 | achu | so I can experiment with my 56K fax modem right ? |
| 13:43.27 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 13:43.41 | b11d | USR always made the sweetest external modems |
| 13:43.48 | n3glv | as long as it's not a WIN modem... |
| 13:43.49 | b11d | Hayes was great back in the day but they tapered off |
| 13:43.54 | n3glv | or is a LIN modem...... |
| 13:44.09 | n3glv | yup, USR was always good |
| 13:44.13 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 13:44.37 | mosty | i have a great old dlink fax modem kicking around somewhere, most reliable modem i've ever owned |
| 13:44.43 | JT | i have two PRI USR TotalControls here |
| 13:44.45 | b11d | haha.. I remember my first external USR 28.8 |
| 13:44.54 | b11d | yeah i've got a totalcontrol here too |
| 13:45.02 | b11d | its alright |
| 13:45.03 | JT | how much did they cost? |
| 13:45.09 | b11d | dunno, been here since before my time |
| 13:45.09 | JT | mine aren't in use |
| 13:45.20 | JT | they're at my house doing nothing :P |
| 13:45.22 | b11d | i'd say like $5k if i remember correctly |
| 13:45.28 | b11d | yeah mine are out of service as well |
| 13:45.28 | JT | AUD$25 ea |
| 13:45.32 | JT | so like USD$20 |
| 13:45.50 | b11d | this is a big rack-mount modem pool w/ PRI interfaces |
| 13:45.50 | n3glv | I can also get total controlls cheap |
| 13:46.03 | JT | each one has a 30 modem E1 card |
| 13:46.12 | *** join/#asterisk nafigator (n=IM@84.22.192.250) |
| 13:46.14 | n3glv | would be cool to recycle the PRI interface cards for use as t1-e1 cards |
| 13:46.20 | b11d | yeah |
| 13:46.32 | b11d | i'll go macgyuver that right now |
| 13:46.36 | n3glv | I think they ran an ISA backplane |
| 13:46.39 | n3glv | haha |
| 13:46.58 | jm|work | 3Com USR External Fax Modem 56K |
| 13:47.00 | b11d | using some common household bleach and a tack, I was able to do it |
| 13:47.06 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 13:47.10 | jm|work | Best. Modem. Ever. |
| 13:47.13 | b11d | agreed |
| 13:47.22 | b11d | Externals always rocked |
| 13:47.30 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-166-226.buff.east.verizon.net) |
| 13:47.34 | b11d | ahh |
| 13:47.34 | jm|work | there were those funky grey ones ... |
| 13:47.43 | jm|work | with a volume control :O I just had a manilla one :( |
| 13:47.55 | JT | hehe, speaking oh, have a couple of rack boxes that have 16 modems each :P |
| 13:47.57 | JT | 33.6k |
| 13:48.04 | n3glv | I think these have the vol wheel |
| 13:48.07 | JT | s/oh/of/ |
| 13:48.17 | n3glv | beige |
| 13:48.29 | b11d | let's start Global 28.8 Day Where we all switch back to 28.8's for a day.. |
| 13:48.31 | n3glv | purple or pink label |
| 13:48.32 | b11d | :| |
| 13:48.42 | JT | if dialup was back in fashion i easily have enough hardware to start up and 80 line isp |
| 13:49.08 | n3glv | hey, we used to push our Commodore 1650 modems from 300 to 485Baud, we were MOVIN man!!! |
| 13:49.17 | b11d | hell yes |
| 13:49.19 | Nivex | b11d: my parents never got out of 28.8 day :( |
| 13:49.26 | b11d | hahaha that sucks |
| 13:49.40 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:49.52 | Nivex | chat: Nov 21 08:48:52 CONNECT 26400/ARQ/V34/LAPM |
| 13:50.02 | n3glv | used to run an acustic modem on a Deckwriter into the PDP11-44 |
| 13:50.03 | n3glv | whee |
| 13:50.09 | b11d | I miss my old INIT string.. |
| 13:50.20 | b11d | hehe |
| 13:50.21 | achu | JT : Is it possible to share the company directory (Extension list) between two asterisk servers ? |
| 13:50.33 | jm|work | S0W3ATDT |
| 13:50.42 | n3glv | Drop your carrier, we have you surrounded |
| 13:50.46 | JT | achu: i'm not even sure what you're refering to with company directory |
| 13:50.55 | *** join/#asterisk Hello2007 (n=wardeh20@mail.splendor.net) |
| 13:51.00 | Nivex | +++ATS0=1 |
| 13:51.03 | b11d | probably the xml directory used in polycom's |
| 13:51.04 | monsted | it used to be great fun to do "/ctcp #channel ping +++ATH0" and see how many people had non-patched OS's :) |
| 13:51.08 | b11d | hahahah |
| 13:51.09 | n3glv | ATD |
| 13:51.17 | Hello2007 | <PROTECTED> |
| 13:51.19 | jm|work | ATZ 'ok' |
| 13:51.25 | Hello2007 | <PROTECTED> |
| 13:51.44 | b11d | AT&F2E0M0S192=0-SDR=2 |
| 13:51.50 | JT | Hello2007: which phone will ring? |
| 13:51.53 | jm|work | b11d: SYN |
| 13:51.55 | JT | the one that called originally |
| 13:52.00 | JT | or another one in the dialplan? |
| 13:52.03 | b11d | :P |
| 13:52.09 | achu | JT: I mean when you are entering into the Digital receptionist , after pressing the "#" key it will ask for the first 3 letters of a person's firstname or lastname and tell us the extension number |
| 13:52.16 | Hello2007 | my polycom phone whin i configured when time out |
| 13:52.23 | *** part/#asterisk n3glv (n=N3GLV@c-71-60-125-243.hsd1.pa.comcast.net) |
| 13:52.34 | JT | achu: still have no idea what that is, did you set it up? |
| 13:52.38 | Nivex | jm|work: SABM |
| 13:53.20 | Hello2007 | may be its a delay setting i should configure???? |
| 13:53.53 | achu | JT : I have two asterisk servers and if somebody calls from external phones such as cellphone or landline to my asterisk server it goes to Digital receptionist right |
| 13:53.53 | jm|work | ATE1Q0 |
| 13:53.58 | JT | Hello2007: is it the calling or the callED phone that rings? |
| 13:54.20 | JT | did you setup "digital receptionist"? |
| 13:54.28 | achu | JT : Yes |
| 13:54.33 | Hello2007 | yes |
| 13:54.41 | Hello2007 | like if i didn t hang up |
| 13:54.48 | JT | Hello2007: it's not a yes/no answer, which one rings??? |
| 13:54.52 | Hello2007 | it rong for 3 /4 time than stop |
| 13:55.14 | Hello2007 | like there is a long delay before it recognize that i hanged up the phone |
| 13:55.15 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:55.29 | jm|work | *TORA* *TORA* *TORA* *TORA* |
| 13:57.24 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 13:58.04 | Hello2007 | the called phone |
| 13:59.02 | JT | for the calling phone you probably need to set a timeout or something in the zap config |
| 13:59.11 | Hello2007 | how? |
| 13:59.22 | JT | i don't know off hand |
| 13:59.27 | JT | i'm sure it will be documented |
| 13:59.31 | JT | try voip-info.org |
| 13:59.33 | JT | or |
| 13:59.35 | JT | ~thebook |
| 13:59.44 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:59.44 | Hello2007 | i have to check th zaptel.conf file? |
| 13:59.58 | JT | zapata.conf more likely |
| 14:00.26 | Hello2007 | i will try |
| 14:01.26 | Hello2007 | is it the rxwinj featurE? |
| 14:01.32 | Hello2007 | rxwink |
| 14:02.03 | JT | i just said i don't know off hand |
| 14:02.12 | Hello2007 | k |
| 14:02.15 | JT | and i really couldn't be bothered to look it up, to be honest |
| 14:02.33 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 14:03.13 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 14:03.21 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:03.46 | *** join/#asterisk M_at (n=643744f8@82.152.60.82) |
| 14:03.56 | Hello2007 | ;-) |
| 14:03.58 | M_at | Anyone here admin user on voip-info.org? |
| 14:06.49 | M_at | There seems to be a bot editting pages |
| 14:08.41 | *** part/#asterisk achu (n=achu@124.125.37.223) |
| 14:08.48 | *** part/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 14:09.35 | monsted | M_at: i'm not an admin, but that sounds interesting - what do you see it doing? |
| 14:09.37 | JT | M_at: there's probably better avenues for contacting admins than this channel |
| 14:09.45 | JT | havent heard of any admins here |
| 14:09.56 | M_at | it's changed a number of links to point to searchadv.com |
| 14:10.21 | M_at | And it re-changed them after I restored them within seconds. |
| 14:11.59 | M_at | And my "beware" message has just been removed |
| 14:12.30 | JT | have you reported it via contax details on voip info? |
| 14:12.36 | JT | contact |
| 14:12.44 | M_at | Cant see them on there - it's a bit of a mess for navigation |
| 14:13.05 | *** join/#asterisk sloth101 (n=mallen@69.71.225.45) |
| 14:13.24 | JT | there's support@voip-info.org mentioned on the front page |
| 14:16.20 | SuPrSluG | wow all the news links point to searchadv |
| 14:16.25 | M_at | Yup |
| 14:16.35 | M_at | No matter how many times I edit them to say otherwise |
| 14:17.17 | monsted | i've reported it to the search site, since this seems to be an "affiliate" |
| 14:17.32 | *** join/#asterisk lowlevel (n=Stuart@bigbrother.vermeulens.com) |
| 14:17.39 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:17.46 | JT | sure it's not a spam site? |
| 14:17.48 | SuPrSluG | probably verizon or another telco |
| 14:18.10 | M_at | It's not support@voip-info.orgerizon |
| 14:18.13 | M_at | Ooops |
| 14:18.34 | *** join/#asterisk ToyMan (n=stuq@74-32-66-238.dsl1.mdl.ny.frontiernet.net) |
| 14:18.54 | monsted | JT: it's a link farm, but i don't really care if they harvest the gmail address i gave them |
| 14:19.02 | JT | hmm |
| 14:19.11 | monsted | (there's no way in hell i'm giving them my real address :)) |
| 14:21.07 | *** join/#asterisk alerios (n=alerios@201.244.173.92) |
| 14:24.36 | monsted | is there a way for a regular user to back out a change made by someone? |
| 14:24.55 | monsted | (ie. delete "Bob"'s changes without redoing it by hand) |
| 14:26.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:27.07 | M_at | oops - support@voip-info.org - Unable to deliver to destination domain |
| 14:27.25 | DarKnesS_WolF | what is wrong with this |
| 14:27.27 | DarKnesS_WolF | exten => _3XX.,1,Set(foo=${ENUMLOOKUP(${EXTEN:1})}) |
| 14:27.31 | PupenoR | For zaptel 1.4.0-betaN, do I still have to run make linux26 to build it for Linux 2.6.X ? |
| 14:27.52 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:27.58 | Juggie | hmmm |
| 14:28.05 | Juggie | yeah the mainpage has like 3 links right on the front |
| 14:28.26 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:28.26 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:28.40 | Juggie | edited by someone whos only a member since today |
| 14:28.47 | M_at | Yup |
| 14:28.58 | rob0 | Damn spammers. :( |
| 14:28.59 | Juggie | anthm, who runs voip-info |
| 14:29.28 | anthm | a guy, forgot his name |
| 14:30.30 | Juggie | theres some bot editing pages on it. |
| 14:30.38 | Qwell | PupenoR: no |
| 14:32.38 | DarKnesS_WolF | hmm this enum thing seems to be very evil ! |
| 14:32.48 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:33.13 | DarKnesS_WolF | exten => _3XX.,1,Set(foo=${ENUMLOOKUP(${EXTEN:1})}) <--- foo is empty in the logs what is wrong here? |
| 14:33.56 | Gr1ncheux | brb |
| 14:34.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 14:35.07 | rob0 | What's this "shockwave flash media" I keep getting prompted for on every page? (Linux x86_64 browser, no Flash player available.) |
| 14:37.33 | M_at | Bloody bob |
| 14:38.55 | anthm | Juggie: James H Thompson <jht@lj.net> |
| 14:45.15 | coppice | I wish he'd stop running voip-info :-) |
| 14:45.57 | sevard | I have a PBX that is groundstart that I only have the ability to feed with loopstart lines, don't ask ;/. I was looking at "ground start" to "loop start" converters and found http://www.gkinc.com/ProductPage-GS1.htm and http://www.gkinc.com/ProductPage-271.htm -- the prices on these boxes seem a little outrageous. I'm really thinking I could either build my own or find one that's cheaper. Does anyone have any experience with this? |
| 14:46.48 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 14:46.54 | klictel | morning all |
| 14:48.39 | tzafrir | sevard, Astrisk/Zaptel supports both gs and ls |
| 14:49.26 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-15-130.w81-248.abo.wanadoo.fr) |
| 14:50.10 | sevard | tzafrir: that wasn't my question at all. |
| 14:51.10 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 14:52.23 | sevard | sup hmmhesays |
| 14:52.31 | b11d | sup Sedorox |
| 14:52.43 | hmmhesays | not much you walking std |
| 14:52.57 | sevard | only because i banged your mom |
| 14:53.06 | hmmhesays | true |
| 14:53.35 | b11d | I fell off a bicycle and landed on a fire hydrant last night :( |
| 14:53.39 | cpm | ouch! |
| 14:53.47 | hmmhesays | damn |
| 14:54.00 | sevard | b11d: how are your teeth? |
| 14:54.02 | hmmhesays | why were you riding a bicycle its cold in minnesota |
| 14:54.08 | b11d | yeah well I lied too |
| 14:54.11 | b11d | sorry |
| 14:54.20 | sevard | you have issues |
| 14:54.26 | hmmhesays | went and saw casino royale last night |
| 14:54.29 | sevard | hmmhesays: it's warm in minnesota. |
| 14:54.37 | sevard | I'm in shorts and a t-shirt |
| 14:54.44 | jeremy_g | waht sevard |
| 14:54.48 | jeremy_g | koool |
| 14:54.56 | hmmhesays | sevard you are on crack |
| 14:55.09 | sevard | don't know where you're at, but the northren states have a warm front friend |
| 14:55.12 | sevard | florida is like -14 though |
| 14:55.17 | hmmhesays | there are 3 minnesotans in this channel right now |
| 14:55.25 | sevard | minnesotians |
| 14:55.30 | b11d | hell yeah |
| 14:55.34 | hmmhesays | www.blaccuweather.com |
| 14:55.34 | sevard | word to the MN bitches |
| 14:55.45 | sevard | BLACCUWEATHER NO WAY |
| 14:55.52 | sevard | b11d: what part of the state are you in?> |
| 14:55.52 | hmmhesays | haha yup |
| 14:56.11 | b11d | Northern MN.. Hibbing actually |
| 14:56.17 | sevard | hmmhesays: HAHHAHHhahahahah that website rules. |
| 14:56.21 | sloth101 | If you guys are warm up there and its cold here in Fla....something is horribly wrong |
| 14:56.34 | sevard | hmmhesays+b11d we should get together and have a drunkfest |
| 14:56.50 | hmmhesays | sevard yes, yes it does |
| 14:56.51 | b11d | fuck yeah! thats the best damned idea i've heard all week |
| 14:57.03 | hmmhesays | can't tonight, band practice |
| 14:57.06 | hmmhesays | tomorrow though |
| 14:57.29 | b11d | where are you guys at? |
| 14:57.30 | sevard | BAND PRACTICE |
| 14:57.35 | sevard | wtf are you some sort of NERD |
| 14:57.45 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:57.49 | hmmhesays | its not school band dipshit |
| 14:58.02 | sevard | bullcrap you're a bandgeek |
| 14:58.16 | awannabe | is there any other way to debug MOH, ive got debug and verbose on 10, and i just see "started music on hold for channel..." then stopped for channel |
| 14:58.18 | *** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) |
| 14:58.20 | sevard | or have your formed your emo group yet |
| 14:58.22 | PupenoR | trying to install zaptel-1.4.0betaN, I've got these errors: http://paste.lisp.org/display/30362, any ideas what's wrong and/or what am I missing ? |
| 14:58.45 | hmmhesays | yeah sevard we're all depressed and want to kill ourselves in shit |
| 14:58.51 | hmmhesays | messy |
| 14:58.58 | sevard | hmmhesays: sounds like a plan, i'll bring the razor blades |
| 14:59.08 | converx | is there a realtime dynamic table for musiconhold.conf config file? |
| 14:59.10 | hmmhesays | sevard: cool, you go first |
| 14:59.18 | sevard | hmmhesays: right after you drink the koolaid |
| 14:59.36 | *** join/#asterisk xlogik (n=Miranda@pawn.twbg.com) |
| 15:00.11 | hmmhesays | looks like the makefile can't uname for the kernel version |
| 15:00.31 | b11d | anyone here ever use a Spherical PBX? |
| 15:00.43 | hmmhesays | as for MOH make an extension that calls music on hold |
| 15:00.59 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 15:01.00 | awannabe | ok, ill try that |
| 15:01.18 | converx | anyone knows if in asterisk realtime tables one can load musiconhold classes dynamically? |
| 15:01.48 | awannabe | same thing, just start/stop it says |
| 15:02.02 | *** join/#asterisk zamba (i=marius@flage.org) |
| 15:02.23 | RoyK | zamba: #asterisk-no |
| 15:04.50 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 15:05.38 | awannabe | just dead silience, lol |
| 15:05.47 | awannabe | unless the mp3s that asterisk comes with are bad..lol |
| 15:06.00 | sevard | did you check the permissions? |
| 15:06.10 | sevard | can that user that's running asterisk read the mp3s? |
| 15:06.49 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:06.58 | awannabe | i set them to 777 for testing, heh |
| 15:09.56 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:10.07 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 15:10.34 | pingwin | i need to reverse the order of zap channel use. To start at channel 23 and work it's way to 1, how can I do that? |
| 15:11.19 | mercestes | pingwin: Why? |
| 15:12.06 | pingwin | because the PRI line I am supplied is trying to create calls on that line begining at 1 going toward 23, when we try to create a call and recieve a call there is a collision and both the line resets |
| 15:12.19 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 15:12.58 | mercestes | pingwin: Good enough. That shouldn't happen though. |
| 15:13.10 | RoyK | pingwin: scan the config files. there are options for that |
| 15:13.23 | pingwin | nope, but any idea how to make the channel hop this way? |
| 15:13.24 | RoyK | pingwin: but there won't be any collisions |
| 15:13.30 | pingwin | RoyK I've been looking around for that |
| 15:13.32 | RoyK | dial(zap/g1 ..... |
| 15:13.35 | file | use a capital G in your group |
| 15:13.39 | RoyK | the group will arrange itself |
| 15:13.42 | RoyK | oh |
| 15:13.43 | RoyK | ok |
| 15:13.49 | file | when you are referencing it in extensions.conf |
| 15:13.57 | mosty | i am having a problem with queues. when multiple callers ring in at the same time, only one of the phones in the queue rings |
| 15:14.17 | pingwin | is your ring_type ringall? |
| 15:14.59 | file | traditionally most telcos send calls starting at the top... which is why 'g' is so common |
| 15:15.02 | pingwin | c'mon is using a capital G really change the way it hunts for an available channel any differently than if I use a lower case g? |
| 15:15.03 | mosty | i am using that now, which works, but my boss wants it to ring one extension at a time |
| 15:15.10 | file | pingwin: ...yes |
| 15:15.17 | sevard | pingwin: very much so |
| 15:15.20 | pingwin | file grovey, thanks just need to check these |
| 15:16.15 | mosty | pingwin: using leastrecent and rrmemory it seems to block if an agent isn't answering their call |
| 15:16.59 | pingwin | i use ringall for all my queues, but I have a simple call center setup |
| 15:17.59 | mosty | also, using leastrecent it sometimes sends two calls to a single extension when others are free |
| 15:19.28 | *** join/#asterisk Sasch (n=Sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 15:19.32 | Sasch | hi all |
| 15:20.43 | mosty | pingwin: i guess it sends blocks until an agent answers because that stops a caller 2nd in line from being answered before the 1st in line is answered. but that's a silly policy really |
| 15:21.11 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:21.47 | pingwin | yeah, but to someone, someone uses it that way, just not me :) |
| 15:22.40 | mosty | if there are N free agents in a queue, it should let the N first callers in the queue ring to those phones |
| 15:22.47 | mercestes | I've used "Ringall" in a call center before. |
| 15:23.08 | mosty | i'm using ringall now, because nothing else seems to have a reasonable policy |
| 15:23.18 | mosty | but my boss is not happy with the solution |
| 15:24.10 | mercestes | mosty: When users start "DND" their phones and forwarding them to their neighbors (or themselves) etc. so they won't have to *work for a living* then ringall is, indeed, the best strategy. |
| 15:24.12 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:25.16 | mercestes | mosty: Hmm...I like your idea, but I think it goes against the idea of a "queue" does it not? |
| 15:25.52 | mosty | mercestes: depends if you like the idea of an entire queue being held up by a single agent |
| 15:26.36 | mosty | also, it seems that it should be smart enough not to send two callers to the same agent at the same time |
| 15:29.26 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 15:29.39 | pingwin | hehe yep |
| 15:29.48 | pingwin | the glory of ringall ;) it's simplicity |
| 15:30.36 | mosty | i'm looking at app_queue.c now :( |
| 15:31.18 | *** join/#asterisk fatpelt (n=pat@mail.feltonline.com) |
| 15:31.59 | *** join/#asterisk ztriver (n=ztriver@62.209.179.131) |
| 15:32.29 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 15:32.30 | mut | Our direct client is currently in need of an Asterisk IP PBX Voice Analyst for a permanent position in the Lowell, MA area. |
| 15:33.05 | fatpelt | good morning. i'm not seeing a function that will allow me to dial two extensions in a dial plan and connect them. does such a thing exist? |
| 15:33.21 | *** part/#asterisk zumbush (n=ztriver@62.209.179.131) |
| 15:34.08 | *** join/#asterisk e-milio (n=emilio@pmr.pmrtechnologies.com) |
| 15:35.06 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-15-130.w81-248.abo.wanadoo.fr) |
| 15:35.46 | mut | wtf |
| 15:35.52 | mut | i get an email from this IT recruiter |
| 15:36.00 | mut | says "Hello, |
| 15:36.00 | mut | I was hoping you could help me out. |
| 15:36.02 | mut | " |
| 15:36.15 | mut | and goes on about how they're looking for an asterisk person in Lowell, MA |
| 15:36.20 | hmmhesays | nice |
| 15:36.36 | mut | lazy fscker |
| 15:36.40 | mut | he can find his own ppl |
| 15:36.49 | mut | or give me a job here! |
| 15:36.52 | mut | :P |
| 15:36.54 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 15:37.30 | mosty | how do i search for bugs at bugs.digium.org? i can't find any search links anywhere |
| 15:38.07 | jserve | 'View Issues' |
| 15:38.19 | jserve | They is the search for that (Filter) |
| 15:38.31 | mosty | hmm ok |
| 15:42.12 | mercestes | mut: Hey....what's that dudes contact information?? |
| 15:42.15 | mercestes | >.> |
| 15:42.50 | mut | i can forward the email if ya want |
| 15:44.11 | mercestes | mut: lol sweet! |
| 15:44.22 | mut | address? |
| 15:44.39 | mut | ya owe me a finders fee if ya get the job |
| 15:44.40 | mut | :P |
| 15:44.53 | mercestes | mut: lol |
| 15:50.17 | hmmhesays | I like to avoid jobs where I actually have to leave the house |
| 15:50.38 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.150.Dial1.SanJose1.Level3.net) |
| 15:50.53 | nays85 | anyone here from NY/NJ/PA? |
| 15:50.54 | mut | heh |
| 15:50.59 | mut | i need one of those jobs |
| 15:51.07 | mut | i don't have to leave the house for mine |
| 15:51.12 | mut | but they prefer i come to work |
| 15:53.01 | vader-- | nays i am |
| 15:53.06 | vader-- | why whats up? |
| 15:54.00 | rob0 | nays85: No, but I play a N'Yawka on the telephone. ;) (Thanks to Stanaphone's area code 347 numbers. :) ) |
| 15:54.10 | blitzrage | is there a clear-all option in voicemail? |
| 15:54.24 | vader-- | nays im actually in the NJ/PA/DE/MD area |
| 15:54.41 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 15:54.50 | dasenjo | Hi! |
| 15:57.38 | nays85 | just wondering... heh... seems like everyone is from the west/midwest in the * community |
| 15:58.49 | blitzrage | I'm from mid-east :) |
| 15:59.05 | blitzrage | (Toronto specifically) |
| 15:59.08 | mercestes | nays85: Are you referring to west/midwest USA or west/midwest world?? |
| 15:59.23 | blitzrage | mercestes: wouldn't that mostly be the same place? :) |
| 15:59.42 | nays85 | i meant the USA, the rest of the users are from all over the globe |
| 15:59.50 | mercestes | blitzrage: west/midwest yes.....but if you start referring to "east" then it gets rather different. |
| 16:00.01 | blitzrage | mercestes: true |
| 16:00.38 | mercestes | I'm centrally located in the US but for some reason Texas is definately considered "west side." |
| 16:00.43 | mercestes | I think it's all the sand, snakes and pistols. |
| 16:03.48 | nays85 | now i wonder, where are all the ITSPs that we use located? |
| 16:04.34 | file | nays85: on Earth :D |
| 16:04.35 | pigpen | I am trying to change the default emailbody of the voicemail email messages, but even after a complete shutdown of asterisk and startup, no dice... ideas? |
| 16:04.52 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:05.08 | nays85 | file : i don't know, based on sellvoip's reply time to email, they must be at least a light-year away |
| 16:07.13 | bXi | is there anybody around who has a working asterisk/iaxmodem/hylafax setup ? |
| 16:08.53 | pigpen | bxi, what troubles are you having....? |
| 16:08.55 | awannabe | how is music on hold such a total PITA. like the default settings dont even work! lol |
| 16:09.20 | bXi | pigpen: the faxes are received now but i need to mail them |
| 16:09.28 | mosty | awannabe: works fine here |
| 16:09.39 | awannabe | monsted: you have a soundcard? |
| 16:09.41 | pigpen | you need to setup your /etc/spool/fax/etc/FaxDispatch file |
| 16:09.47 | bXi | okay |
| 16:09.58 | awannabe | mosty: sorry, do you have a soundcard, or /dev/dsp entry? |
| 16:10.05 | bXi | pigpen: any examples? |
| 16:10.38 | pigpen | google it, but if you come back dry, ask... |
| 16:10.43 | pigpen | this is how I found my setup. |
| 16:11.02 | pigpen | (and I am trying to fix an issue of my own too :) ) |
| 16:11.05 | bXi | and what daemons should be running for ths to work? |
| 16:11.26 | *** join/#asterisk brat7987 (n=Miranda@62.85.37.235) |
| 16:11.29 | pigpen | do you have the hylafax scheduler running? |
| 16:12.00 | pigpen | I can rephrase: can you receive faxes? |
| 16:12.13 | bXi | i can receive faxes as .tif files |
| 16:12.26 | pigpen | good...now you should only have to setup your FaxDispatch file |
| 16:12.51 | bXi | cool |
| 16:12.57 | pigpen | if you can rec the fax to that point, you are almost there. |
| 16:13.28 | mosty | awannabe: no |
| 16:14.04 | hmmhesays | bah I hate faxing |
| 16:14.08 | hmmhesays | it makes me want to kill people |
| 16:14.08 | bXi | crap |
| 16:14.15 | bXi | now i dont have anyone to send me a fax :/ |
| 16:14.20 | b11d | i can |
| 16:14.30 | bXi | you can? |
| 16:14.34 | b11d | sure |
| 16:14.47 | bXi | wouldnt that be like uber expensive? |
| 16:14.54 | b11d | sending you a fax? where are you? |
| 16:14.56 | awannabe | mosty: can you post your config? ive tried like 10 differnet wqays, i either get dead silence, or TONS of static noise, using the stock mp3s |
| 16:14.57 | bXi | .nl |
| 16:15.34 | pigpen | bxi, you may want to call a friend or google for a free faxback service. |
| 16:16.00 | bXi | most of my friends are too young to even know what a fax is :p |
| 16:16.15 | pigpen | lucky them. |
| 16:16.36 | hmmhesays | that would be freaky if you are like 45 |
| 16:16.43 | bXi | i'm 21 :D |
| 16:16.50 | hmmhesays | ahh a fun age |
| 16:16.50 | pigpen | dam..just a kid. |
| 16:16.58 | hmmhesays | have you passed out on a bar floor yet? |
| 16:17.04 | awannabe | LOL |
| 16:17.31 | pigpen | hmm...this really shows our age... |
| 16:17.48 | bXi | hehe |
| 16:17.53 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 16:17.55 | awannabe | im 22 and never passed out in a bar |
| 16:18.03 | awannabe | car, yes, house, possibly, but never a b ar |
| 16:18.10 | bXi | :p |
| 16:18.13 | sloth101 | Is it a misconfiguration if "sip show channels" shows (None) for User/ANR and Form unkn? |
| 16:18.16 | Corydon-w | I'm 30, and I've never passed out in a bar |
| 16:18.33 | sloth101 | for == and |
| 16:18.55 | sloth101 | Sorry that simply wasnt english |
| 16:19.56 | pigpen | bxi, you will probably want to send that file off as a pdf |
| 16:20.17 | bXi | do i need to do something extra for that? |
| 16:20.22 | bXi | i found out a basic configuration |
| 16:20.31 | bXi | which says to send as pdf |
| 16:20.35 | pigpen | Ok..I give up on changing the emailbody of the voicemail emails.... |
| 16:20.38 | pigpen | k |
| 16:20.48 | pigpen | yeah..that is probably it. |
| 16:20.51 | bXi | you need help with the email body ? |
| 16:21.01 | pigpen | yeah... |
| 16:21.19 | *** join/#asterisk Asttt (n=asterisk@193.110.8.51) |
| 16:21.21 | bXi | do you have freepbx ? |
| 16:21.28 | rob0 | <== 45 and never passed out anywhere :) |
| 16:21.30 | Asttt | Hi all |
| 16:21.41 | pigpen | I just stuck emailbody= and emailsubject= in the voicemail.conf (with a example off of voipinfo) , but it doesn't take it. |
| 16:21.54 | pigpen | bXi, no..not freepbs. |
| 16:21.59 | pigpen | freepbx that is. |
| 16:22.05 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 16:22.12 | pigpen | I run Gentoo with all custom. |
| 16:22.19 | Asttt | I have a problem with h323/sip ... One way audio. Can anybody help me please ? |
| 16:22.32 | rob0 | I'm going to do a Linux kernel upgrade today, thought I might try zaptel 1.4 beta. Would that be a problem with * 1.2, or should I upgrade that too? |
| 16:22.37 | Asttt | If i describe it |
| 16:22.47 | bXi | pigpen: i have a custom product here for customers |
| 16:23.01 | bXi | they install it and they have a fully working asterisk box in 20 minutes |
| 16:23.14 | pigpen | yeah..but we haven't gotten to the quickie install yet.... |
| 16:23.40 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 16:23.41 | awannabe | MOH requires ztdummy or zaptel?? |
| 16:24.08 | pigpen | Asttt, shoot away.... |
| 16:24.13 | file | rob0: most likely yes |
| 16:24.26 | file | rob0: I doubt zaptel 1.4 will work with 1.2 |
| 16:24.38 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 16:24.49 | *** join/#asterisk lupino3 (n=lupino3@217-133-98-121.b2b.tiscali.it) |
| 16:24.53 | lupino3 | hello guys |
| 16:24.56 | lupino3 | I've got a problem |
| 16:25.04 | bXi | pigpen: http://pastebin.ca/254190 |
| 16:25.08 | lupino3 | I'm unable to make two hard videophone to video-call |
| 16:25.09 | bXi | maybe thats any help |
| 16:25.14 | rob0 | Thanks, I guess I'll hold off on the zaptel upgrade, just recompile my zaptel 1.2 for the new kernel. |
| 16:25.28 | lupino3 | they are TelePhoSee |
| 16:25.37 | lupino3 | is there any special settings I have to use? |
| 16:25.43 | lupino3 | they use h261 |
| 16:25.49 | lupino3 | and in sip.conf, context general |
| 16:25.50 | pigpen | bxi, thanks...I will review. |
| 16:25.50 | lupino3 | I have |
| 16:25.53 | lupino3 | disallow = all |
| 16:25.55 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 16:26.00 | lupino3 | allow = h261 |
| 16:26.04 | Asttt | I make a call in the following way |
| 16:26.04 | Asttt | H323 Softphone-->Asterisk-->Sip Operator |
| 16:26.04 | Asttt | For this call flow I get one way audio (H323 client doesn't hear remote side). As debug shows (rtp debug) Asterisk doesn't receive RTP stream from Sip operator. I see only packets from H323 and to Sip Operator. |
| 16:26.04 | Asttt | For H323 client I use openphone and SJphone. Asterisk uses ooh323 which is in Asterisk-addons. I tried both 1.2 version and 1.4. |
| 16:26.05 | Asttt | As I see from sip debug Sip Operator uses SER. |
| 16:26.07 | Asttt | I tried to use different codecs ( gsm, g711 etc.). As I see from sip debug codecs are supported by remote side. |
| 16:26.10 | Asttt | if I use these schemes everything is ok: |
| 16:26.12 | Asttt | H323 Softphone-->Asterisk-->Sip Phone |
| 16:26.14 | Asttt | Sip Phone-->Asterisk-->Sip Operator |
| 16:26.16 | Asttt | I don't use firewalls. |
| 16:26.21 | lupino3 | allow = alaw |
| 16:26.24 | lupino3 | allow = ulaw |
| 16:26.36 | lupino3 | please help me! :( |
| 16:26.52 | Corydon-w | lupino3: Asterisk does not support h261, only h263 and h264 |
| 16:27.05 | lupino3 | oh, ok |
| 16:27.18 | Qwell[] | patches accepted (if it isn't patented) |
| 16:27.18 | lupino3 | but.. :http://www.voip-info.org/wiki-Asterisk+video |
| 16:27.23 | file | sure it supports h261 |
| 16:27.30 | lupino3 | it says h261 |
| 16:27.36 | lupino3 | well, I'll try h263 thanks |
| 16:27.42 | sloth101 | Is it a misconfiguration if "sip show channels" shows (None) for User/ANR and for Form unkn? I am still unsure how it can connect the call and not know if its ulaw etc. Sorry kind of got this system dumped on me. :-( |
| 16:28.16 | Asttt | Does anyone have an idea about may problem ? |
| 16:28.23 | lupino3 | file: so what could be my problem? |
| 16:28.28 | lupino3 | the phones are registered to asterisk |
| 16:29.25 | file | without seeing sip debug one can only guess |
| 16:29.38 | lupino3 | ok so |
| 16:29.45 | lupino3 | I activate sip debug |
| 16:29.47 | lupino3 | and then |
| 16:29.56 | lupino3 | I'll post more info |
| 16:29.56 | lupino3 | thanks |
| 16:31.17 | lupino3 | csdasdasdasèm |
| 16:31.29 | lupino3 | sorry, focus problems. :) |
| 16:32.32 | *** join/#asterisk apardo (n=apardo@87.217.145.57) |
| 16:32.45 | sloth101 | I will take that as "you missed a google link" ;-) |
| 16:33.02 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-d1155d2dc48f8336) |
| 16:33.37 | Asttt | So can anybody suggest what to check ? |
| 16:34.03 | pigpen | Asttt, what is the "SIP Operator" ? |
| 16:34.08 | pigpen | sip phone? |
| 16:34.14 | Asttt | voiptalk.org |
| 16:34.27 | Asttt | I have username/password to register |
| 16:34.51 | pigpen | if a sip phone works, but the voiptalk does not, then you may want to contact the. |
| 16:34.55 | pigpen | s/the/them |
| 16:37.42 | Asttt | sounds good ... :( |
| 16:38.52 | pigpen | yeah...sorry. |
| 16:39.26 | pigpen | Asttt, you are connecting to VoIPtalk over the internet right? |
| 16:39.32 | Asttt | yes |
| 16:39.33 | pigpen | Don't you have a firewall that does nat'ing? |
| 16:39.39 | Asttt | no |
| 16:39.47 | Asttt | normal IP |
| 16:40.01 | pigpen | Ok..so this is a "real" ip on the internet. |
| 16:40.07 | Asttt | yes |
| 16:41.01 | pigpen | what service are you using them for? |
| 16:41.13 | pigpen | sip trunk? |
| 16:41.23 | Asttt | to make calls PC-to-Phone |
| 16:41.27 | Asttt | yes |
| 16:41.52 | pigpen | have you tried iax trunk with them? |
| 16:42.03 | pigpen | iax is preferred with Asterisk |
| 16:42.11 | Asttt | hmmmm ... I'm not sure they can |
| 16:42.26 | Asttt | I don't configure trunk ... just exten => _00[1-9].,1,Dial(SIP/voiptalk/${EXTEN}) |
| 16:42.27 | pigpen | yep....http://www.voiptalk.org/products/voip-setup.php |
| 16:42.42 | pigpen | see the first link under "Asterisk PBX Setup" |
| 16:42.53 | pigpen | IAX would work better for you anyway... |
| 16:43.09 | Asttt | 10x |
| 16:43.12 | Asttt | I'll try |
| 16:43.30 | pigpen | and you could probably get more help from the asterisk guys...as it is not as goofy as sip. |
| 16:43.44 | pigpen | ie: it handles nat, routing, latency better than sip....normally. |
| 16:43.45 | lupino3 | solved |
| 16:43.48 | lupino3 | I missed |
| 16:43.51 | lupino3 | allow = h261 |
| 16:43.52 | lupino3 | and |
| 16:43.56 | lupino3 | videosupport = yes |
| 16:44.00 | lupino3 | in [general] |
| 16:44.01 | lupino3 | :) |
| 16:46.11 | *** join/#asterisk xnon (n=xnon@200.8.30.3) |
| 16:49.27 | zamba | how can i set up a dialplan for providing the local extension to dial? |
| 16:49.44 | zamba | i call a number and then i get prompted for the local extension that should be dialed? |
| 16:50.13 | sloth101 | Is it normal for a VoIP * system to be allowing g726,ulaw,alaw,g723,slin,and gsm? It would seem from my admited limited understanding that you should standardize. |
| 16:50.56 | e-milio | Hello all |
| 16:51.25 | e-milio | just wanted to ask everybody if someone have had any problems with newer HP/Proliant servers |
| 16:52.00 | e-milio | or if a specific one of them can be recomended for a cliente that only uses HP |
| 16:52.02 | e-milio | ?? |
| 16:53.38 | awannabe | e-milio: what are you doing with thix box? |
| 16:53.46 | e-milio | two boxes |
| 16:53.49 | awannabe | like just SIP/IAX, or what? |
| 16:54.06 | e-milio | one sip gw to a colo where the other box connects to 4 t1s |
| 16:54.40 | e-milio | the one at colo also connects to a box out the country |
| 16:54.44 | sloth101 | Any guesses on the setup I mentioned. Just trying to get a handle on this as it seems that this box was never finished. |
| 16:55.03 | sevard | zamba: you want an IVR, check voip-info.org |
| 16:55.15 | e-milio | and to many small (3/4 voip phones) branches |
| 16:55.42 | *** join/#asterisk apardo (n=apardo@87.217.147.49) |
| 16:55.49 | awannabe | well your linux distro depends more im sure, heh |
| 16:56.04 | awannabe | gotta make sure those RAID cards are supported, and the Sanngoma PRI cards work in just about anything |
| 16:56.09 | *** join/#asterisk afrosheen (n=cj@txprotoa2.august.net) |
| 16:56.39 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-133.range81-152.btcentralplus.com) |
| 16:56.40 | GaVak | is where any way to see what codec a channel is using? |
| 16:56.47 | GaVak | while active i mean. |
| 16:56.47 | e-milio | awannabe: sanngoma > digium ? |
| 16:56.49 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 16:57.02 | sloth101 | sip show channel (channel id) |
| 16:57.14 | sloth101 | for sip sorry |
| 16:59.02 | awannabe | e-milio: i like sangoma better.. |
| 16:59.31 | awannabe | i dont want to start a flame war in here, but according to everyone ive talked to, no know hardware problmes exist |
| 16:59.38 | awannabe | know what i mean? |
| 17:00.05 | tRSS | mine is a bit complicated situation and I will try my best to explain as much as I can. I have a server A with a public IP, while Server B DOES NOT have a public IP. I am trying to call server B from server A, while asking server B to register with Server A using IAX (register => statement). When Server A calls Server B, I get this error: Rejected connect attempt from 1X4.XX9.XX.1XX, who was trying to reach 's@' |
| 17:00.35 | GaVak | cool, thnx |
| 17:01.33 | tRSS | on Server B and Server A gives this error: Call rejected by 1X3.2XX.2X1.XX: No authority found |
| 17:01.33 | sloth101 | GaVak: Glad it heled |
| 17:01.33 | sloth101 | GaVak: healped even |
| 17:01.33 | sloth101 | :-( |
| 17:01.33 | b11d | gilgamec |
| 17:01.38 | b11d | WHAT WAS THAT ABOUT THE GILGAMECS??! |
| 17:01.57 | e-milio | awannabe: no prob exists with hp/sangoma you mean ? |
| 17:02.01 | hmmhesays | haha |
| 17:02.24 | b11d | so when are we having that party? |
| 17:02.38 | b11d | MinnesotAsterisKonvention2005 |
| 17:02.39 | b11d | doh |
| 17:02.53 | Asttt | I have another question ... Does anybody use some kind of global directory for contacts ? |
| 17:03.04 | awannabe | e-milio: from what ive read yes, but you never know, heh |
| 17:03.31 | b11d | what are some common default extensions? I've got 9 for outside line, 0 for operator, 8 for voicemail, and 7 for directory. |
| 17:03.31 | Asttt | So that any registered user could use global contacts |
| 17:03.32 | b11d | what am I missing |
| 17:04.17 | b11d | iaxy? is that still around? |
| 17:04.36 | b11d | :P |
| 17:05.25 | b11d | IAX? is that still around? |
| 17:05.26 | b11d | :P |
| 17:05.34 | e-milio | awannabe: thanks |
| 17:06.00 | b11d | Inter Asterisk eXchange? is that still around? |
| 17:06.00 | b11d | :P |
| 17:06.13 | b11d | ok im done |
| 17:07.06 | hmmhesays | so the new gui looks real nice |
| 17:07.41 | pigpen | anywhere we could see screenshots? |
| 17:09.42 | *** join/#asterisk af_ (n=af@ip-130-179.sn2.eutelia.it) |
| 17:09.50 | *** join/#asterisk atauser (n=someone@212.188.138.98) |
| 17:10.29 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:10.31 | mercestes | Can I use / as a line continuation? |
| 17:11.24 | awannabe | e-milio: i think you will be safe |
| 17:12.11 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-f5cabc688a015ac1) |
| 17:12.39 | b11d | yeah.. where can we see it hmmhesays? |
| 17:12.53 | hmmhesays | http://www.asterisk.org/node/111 |
| 17:12.57 | hmmhesays | i'm installing the framework atm |
| 17:13.22 | b11d | *shudder* |
| 17:13.42 | b11d | it doesnt look bad, but I doubt i'll use it |
| 17:13.57 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 17:14.00 | hmmhesays | end users like guis |
| 17:14.48 | b11d | yeah, they do. |
| 17:15.01 | rob0 | Asterisk? Is that still around? |
| 17:15.06 | b11d | :) |
| 17:15.15 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:15.18 | pigpen | I thought Cisco bought that.... |
| 17:15.31 | b11d | Cisco folded a few years back.. |
| 17:15.35 | pigpen | hmm...maybe it was Apple... |
| 17:15.41 | b11d | yeah it might have been them |
| 17:15.49 | pigpen | iPhone...yeah.... |
| 17:15.56 | Qwell[] | atauser: You're connecting an ATA186 to a CCM? |
| 17:16.23 | b11d | Appletrix |
| 17:17.01 | pigpen | Appletrix@home ? |
| 17:17.02 | atauser | Qwell[]: yes. all works fine for faxes. but for various reasons i need to have an analog phone connected too, with options to transfer calls in and out |
| 17:17.28 | Qwell[] | atauser: If you were able to get me a pcap dump of a couple of calls...you'd totally be my hero |
| 17:17.50 | atauser | Qwell[]: heheh |
| 17:18.18 | Qwell[] | atauser: It'd help me get chan_skinny working better, so you don't have to use CCM anymore, heh |
| 17:18.24 | b11d | when Qwell tells you to get a pcap dump, YOU GET IT. |
| 17:18.27 | b11d | now GO |
| 17:18.28 | b11d | :) |
| 17:18.52 | Qwell[] | I actually need a pcap dump from a phone with a display (ie; not an ATA186), but...whatever I can get, will help |
| 17:19.20 | pigpen | yeah...I am here in the "jump position" just waiting for the command... |
| 17:19.23 | Qwell[] | I have a dump of a call FROM a phone to CCM, but I really need one going the other way |
| 17:19.58 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:20.36 | atauser | well i have a complete CCME (call manager express) system here with 79xx phones and a number of ATAs as well |
| 17:20.46 | Qwell[] | atauser: That'd be perfect, if you could manage that |
| 17:20.48 | atauser | if you can get call transferring working for me from an ata-connected phone, i may be able to oblige! |
| 17:20.48 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 17:20.51 | spaghetty | hi |
| 17:21.31 | b11d | we're closed. |
| 17:21.43 | b11d | ok, we're not.. |
| 17:21.53 | b11d | whats up? |
| 17:23.05 | *** join/#asterisk riddlebox (n=riddlebo@24-207-167-28.dhcp.stls.mo.charter.com) |
| 17:23.36 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 17:23.37 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 17:24.10 | *** join/#asterisk sam33 (n=sam@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
| 17:24.10 | angryuser | anyone here use astribank? |
| 17:24.33 | angryuser | i have close/related a bit offtopic question can i ask it?:) |
| 17:25.30 | b11d | yes.. ask it |
| 17:25.30 | angryuser | :) |
| 17:25.30 | b11d | dont be so angry.. relax.. |
| 17:25.32 | spaghetty | sorry I need to keep information about number of open channel into my asterisk box ... |
| 17:25.38 | spaghetty | working with manager interface |
| 17:27.16 | tRSS | so can anyone help me with my problem, stated above? |
| 17:27.21 | angryuser | well i have bought astribank, their init script goes to /proc/bus/usb, and normally in linux sysstem when you when you plug usb devise, you got some folders under /usb, and i have nothing, suse. 10.1, under suse 9 i got it, but i need to use 10.1, any help? |
| 17:27.42 | angryuser | *device |
| 17:29.18 | angryuser | i tryed to contact xorcomm, i have recieve no answer |
| 17:30.36 | *** join/#asterisk alerios (n=alerios@201.244.173.92) |
| 17:30.36 | riddlebox | can anyone go to mozilla.com and try to download thunderbird and firefox, I cant get to it |
| 17:31.22 | *** part/#asterisk alerios (n=alerios@201.244.173.92) |
| 17:33.06 | b11d | i can get there |
| 17:33.10 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:33.20 | riddlebox | so you can actually download them? |
| 17:33.30 | b11d | yep |
| 17:33.36 | riddlebox | I wonder why I cannot |
| 17:33.38 | b11d | downloads popped right up |
| 17:33.54 | riddlebox | yeah downloads did for me to but when I click to download them it doesnt |
| 17:34.36 | b11d | I was able to dl them at 400k/sec |
| 17:34.52 | b11d | im at www.mozilla.com/en-US/ |
| 17:34.58 | riddlebox | me too |
| 17:35.20 | riddlebox | I get Internet Explorer cannot display the webpage |
| 17:35.21 | b11d | I need to write a parser for these damned CDRs |
| 17:35.26 | b11d | reboot your box |
| 17:35.29 | riddlebox | I have |
| 17:35.34 | b11d | thats strange then |
| 17:35.38 | b11d | clear your offline files and all that jazz |
| 17:35.48 | riddlebox | I wish they would just let me put linux on here |
| 17:35.58 | b11d | get a bootable distro and just use that |
| 17:37.04 | riddlebox | but I need to use windows for my job, I program phone systems and sadly most have tools that are only for windows |
| 17:37.29 | b11d | sure.. sure.. :P |
| 17:38.25 | riddlebox | my bosses dont see how asterisk could compete with the "others", I have tried to explain.... |
| 17:39.19 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-15-130.w81-248.abo.wanadoo.fr) |
| 17:43.45 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:45.10 | paolob | anyone uses wengo? I have problems with sound: asterisk connects me to wengo, but when the communication is established the sound disappears. |
| 17:46.30 | b11d | Anyone ever use a Spherical PBX? |
| 17:48.57 | angryuser | i have used wengo, quality was really bad, so i have changed |
| 17:49.05 | angryuser | happy now |
| 17:49.47 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 17:49.49 | paolob | angryuser, happy with what provider |
| 17:49.51 | paolob | ? |
| 17:50.22 | angryuser | i dont know i you consider it like advertising, but its a phonesystems |
| 17:50.46 | angryuser | its a french one |
| 17:51.12 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 17:52.24 | angryuser | another thing, if you souscribe that, the dont really tell you that you can have 2 communication's at the same time per account |
| 17:52.44 | angryuser | ;) now you know |
| 18:02.48 | *** join/#asterisk ramtha (n=ramtha@p5088B33C.dip0.t-ipconnect.de) |
| 18:02.49 | ramtha | hey |
| 18:03.34 | b11d | hey |
| 18:03.39 | ramtha | is t.38 passthrough in 1.4 also working over iax? sip client - asterisk - (iax2)asterisk(sip) - provider |
| 18:03.40 | ramtha | ? |
| 18:04.05 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 18:04.06 | ramtha | after reading i think t.38 ist only implemented in sip... |
| 18:05.53 | *** join/#asterisk p1p (i=pip@mail.comp911.com) |
| 18:07.24 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:14.33 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 18:15.05 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 18:15.27 | hoobastooba | i dont know how i did it, but somehow i have installed asterisk without SIP support it looks like. |
| 18:16.09 | hoobastooba | I did an show channeltypes and SIP is not in there... is there a way to fix this? |
| 18:16.14 | hoobastooba | I dont know how i missed it... |
| 18:17.17 | hoobastooba | actually it looks like somehow nearly everything is missing. |
| 18:17.17 | hoobastooba | i will start over. |
| 18:21.56 | *** join/#asterisk zx6r (n=_@CPE000ab89cd33e-CM0011e6beb37b.cpe.net.cable.rogers.com) |
| 18:24.23 | Dr-Linux|work | hhm.. strange .. MOH is not playing |
| 18:24.24 | Dr-Linux|work | <PROTECTED> |
| 18:24.50 | converx | anyone knows how to define musiconhold classes in realtime asterisk? |
| 18:25.52 | Dovid | converx: what version of asterisk r u using ? |
| 18:26.10 | *** join/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com) |
| 18:26.16 | Dr-Linux|work | 2.9 |
| 18:26.25 | Dr-Linux|work | Dovid: how can i start moh player? |
| 18:26.54 | Corydon-w | converx: it only supports the old static database config. There is no realtime config of musiconhold |
| 18:28.06 | *** join/#asterisk yogurt2ungue (n=yogurt2u@host96.200-117-56.telecom.net.ar) |
| 18:28.12 | k31th | whats a decent codec ? |
| 18:28.25 | Dr-Linux|work | Corydon-w: any clue? |
| 18:28.56 | Corydon-w | Nope |
| 18:29.03 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-143-6.dyn.embarqhsd.net) |
| 18:29.13 | Dr-Linux|work | ok thx |
| 18:29.26 | De_Mon | what is the equivilant to SetIf() |
| 18:29.41 | Corydon-w | There is no SetIf |
| 18:30.12 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 18:30.18 | De_Mon | I'm looking at Set(${IF(<cond>?<true>:<false>)}) |
| 18:30.42 | Corydon-w | but you could do Set(${IF(cond?varname:foo)}=value) |
| 18:31.43 | De_Mon | err Set(var=${IF()}) |
| 18:31.45 | Corydon-w | Or: Set(varname=${IF(cond?newval:${varname})}) |
| 18:32.18 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 18:32.22 | *** join/#asterisk luke-jr_work (n=luke-jr@2002:4335:4375:0:20d:60ff:fe60:756a) |
| 18:32.51 | De_Mon | I didn't really like the idea of passing the varname for value if false, but if I use an IF for the varname itself, it'll raise an error each time it tries to set a nonexistant varname woln't it? |
| 18:33.28 | Corydon-w | Nope |
| 18:33.35 | Corydon-w | You can set any variable you like |
| 18:34.07 | De_Mon | you set foo if the var didn't exist, I wasnt going to give a value for false |
| 18:34.15 | De_Mon | Set(=value) basically |
| 18:34.17 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:34.33 | Corydon-w | Yeah, that's an error |
| 18:34.46 | De_Mon | okay, foo it is. it looks a bit cleaner |
| 18:34.47 | Corydon-w | but you could set foo just fine, in the case of the false |
| 18:35.38 | EmleyMoor | If I want to support a pulsedial phone on a given FXS port, but not on others, is it correct to put "pulsedial=yes" in zapata.conf for the wanted one and "pulsedial=no" for the others? If so, what diagnostics can I run to investigate why it doesn't work? |
| 18:36.11 | EmleyMoor | (or is the word something other than pulsedial?) |
| 18:36.52 | Dr-Linux|work | how can i start mpg123 player service :S |
| 18:37.16 | Dr-Linux|work | it's not running, i assume it should be atuo start when MOH is on |
| 18:37.18 | Dr-Linux|work | but it doesn't |
| 18:38.10 | zx6r | Dr-Linux|work: it should auto-start with the asterisk startup |
| 18:38.18 | EmleyMoor | (or is pulsedial a global setting?) |
| 18:38.45 | Dr-Linux|work | zx6r: yeah, but it's not, even it process is also not running |
| 18:38.49 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 18:38.51 | hmmhesays | wtf you need zaptel to install asterisk-gui? |
| 18:39.20 | hmmhesays | death |
| 18:39.35 | zx6r | Dr-Linux|work: you have mpg123 installed, correct? (stupid question, but i have to ask:P) |
| 18:40.05 | hmmhesays | why would you want to allow pulse for some an not others |
| 18:40.08 | Dr-Linux|work | zx6r: i'm sure it's installed bcoz i'm using full package version of RHEL 4 |
| 18:40.19 | hmmhesays | pulse=yes doesn't break dtmf |
| 18:40.35 | EmleyMoor | Should be "pulse"? OK |
| 18:41.06 | Dr-Linux|work | zx6r: it's installed but moh is not picking it up |
| 18:42.14 | EmleyMoor | Just getting congestion tone back when I use it |
| 18:43.10 | hmmhesays | curious why are you using pulse? |
| 18:43.15 | ramtha | Dr-Linux|work: do you using custom command for spawning mpg123? |
| 18:43.31 | EmleyMoor | hmmhesays: I am planning to use a rotary phone as one of my extensions |
| 18:43.32 | Dr-Linux|work | ramtha: no dude it's default |
| 18:43.41 | Dr-Linux|work | ramtha: it's working fine for all of my other servers |
| 18:43.43 | *** join/#asterisk algorithmn (n=algorith@ool-45722b4c.dyn.optonline.net) |
| 18:43.46 | spaghetty | some one can help me |
| 18:43.46 | Dr-Linux|work | but not for this :S |
| 18:44.02 | spaghetty | i need to know how many calls created with Originate |
| 18:44.20 | spaghetty | are active |
| 18:44.25 | *** join/#asterisk knight111 (n=v@pat.syndesis.com) |
| 18:44.36 | spaghetty | everithing using manager api |
| 18:44.47 | knight111 | hi.. |
| 18:45.00 | *** join/#asterisk fatpelt (n=pat@titan.q90.com) |
| 18:45.17 | knight111 | my asterisk process runs at 100% CPU.. unless I restart asterisk it stays at 100% |
| 18:45.21 | fatpelt | is there a way to have a dialplan dial two extensions then bridge them together? |
| 18:45.39 | knight111 | this happens randomly.. |
| 18:46.14 | knight111 | is there a problem with FC5 and asterisk.. I dont see this happening in FC4... |
| 18:46.53 | zx6r | Dr-Linux|work: if you try to run mpg123 from a terminal, what happens? |
| 18:47.04 | *** part/#asterisk algorithmn (n=algorith@ool-45722b4c.dyn.optonline.net) |
| 18:47.10 | EmleyMoor | It's having problems interpreting the pulses |
| 18:47.37 | Dr-Linux|work | zx6r: how can i run it from the shell? :S |
| 18:48.37 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:49.14 | knight111 | i am planning on installing asterisk on centos instead of FC5.. anyone thinks that will help? |
| 18:49.24 | zx6r | just type mpg123 |
| 18:49.37 | Dr-Linux|work | zx6r: nah it doesn't |
| 18:49.49 | zx6r | there's your problem |
| 18:50.53 | zx6r | Dr-Linux|work: do a "locate mpg123" from the terminal and see if it finds it |
| 18:50.56 | vader-- | knight try debian |
| 18:51.04 | Dr-Linux|work | zx6r: it did |
| 18:51.07 | Dr-Linux|work | i got a few |
| 18:51.10 | vader-- | i have great success with it and so have alot of other people in here |
| 18:51.21 | Dr-Linux|work | zx6r: let's find a new version of mpg123 and install it |
| 18:51.25 | zx6r | Dr-Linux|work: try running one explicitly |
| 18:51.34 | zx6r | i think rhel4 comes with 0.60 |
| 18:51.35 | Dr-Linux|work | i'm intersting with native moh , but this kinda hard for now |
| 18:53.26 | Dr-Linux|work | zx6r: should i install this one? http://www.macupdate.com/download.php/6275/mpg123-0.59r.pkg.tgz |
| 18:54.24 | zx6r | Dr-Linux|work: if you already have it installed, you shouldnt have to |
| 18:54.31 | zx6r | yu might just have to explicitly call it in the musiconhold.conf |
| 18:55.42 | Dr-Linux|work | zx6r:i don't think i have it> please /msg |
| 19:01.49 | knight111 | vader.. is Debian better? These are 64 bit boxes.. can I use debian 64 bit version? |
| 19:02.19 | knight111 | the problem is I am not sure if the 100% CPU usage is beacuse of asterisk or it's because of the OS.. |
| 19:02.50 | *** join/#asterisk rene1 (n=rene1@189.140.242.8) |
| 19:03.00 | rene1 | hello |
| 19:03.19 | knight111 | hi rene |
| 19:03.30 | rene1 | i have an * box that has a PRI connected |
| 19:04.09 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 19:04.19 | rene1 | i need to |
| 19:04.35 | flujan | hi guys... is there a native postgres connector to use with asterisk? |
| 19:04.45 | rene1 | set a callerid from sip.conf |
| 19:05.08 | rene1 | otherwise the telco will reject the call (ISDN) |
| 19:05.53 | rene1 | the problem is that even if i set a callerid in iax.conf i can get the call to connect via ISDN |
| 19:06.03 | rene1 | *cant |
| 19:07.40 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 19:08.05 | *** join/#asterisk AJaymn (i=TJBoi@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 19:09.24 | rene1 | so i was wondering if i can use setcalleridpres |
| 19:09.49 | rene1 | to avoid doing some hack of having to route every iax-isdn call as a iax-sip-isdn call |
| 19:10.00 | flujan | guys? someone using asterisk with postgresql without odbc? |
| 19:10.38 | Corydon-w | What's wrong with using ODBC? |
| 19:10.50 | flujan | when I use asterisk and postgresql using unixodbc I have strange problems. |
| 19:11.09 | Corydon-w | Which would be what? |
| 19:11.58 | knight111 | I have been using unixodbc with mysql and haven't had any problems... |
| 19:12.59 | *** join/#asterisk HCISjason (n=HCISjaso@hcisfw2.hcis.net) |
| 19:13.00 | *** join/#asterisk Menace- (n=menace-@66.181.104.11) |
| 19:13.32 | *** join/#asterisk saftsack (n=saftsack@pD9E06AB5.dip.t-dialin.net) |
| 19:13.41 | flujan | Corydon-w, I dunno... If there are in the same machine I have problems. If I remove postgresql and put it in other machine it solves the problem. |
| 19:14.06 | flujan | knight111, all our system is based on postgresql. :( |
| 19:14.20 | Corydon-w | Have you tried using TCP/IP sockets with postgres on the local machine? |
| 19:14.48 | HCISjason | Hello, need some expertise on zaptel. Having problems recognizing hookflash on E&M channels. When we receive the hook flash asterisk responds back with a "hangup". Anyone know why this is? Thanks |
| 19:15.35 | *** join/#asterisk zotz (n=zotz@208.196.247.175) |
| 19:16.00 | Corydon-w | HCISjason: are you using em signalling or em_w? |
| 19:16.07 | HCISjason | em |
| 19:16.12 | Corydon-w | Try em_w |
| 19:16.40 | Corydon-w | That hook flash is considered a "wink" |
| 19:17.03 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 19:17.42 | HCISjason | IVR system we are connecting to is expecting ANI#DNS# in that format in order for us to connect to the call. This works in featd by just dialing the DNIS or in em by dialing ani#dnis#, however we cannot get it to work with E&M wink signal. |
| 19:19.40 | knight111 | has anyone experienced 100% cpu usage of asterisk process? |
| 19:20.04 | HCISjason | Corydon-w but I read in a bounty that the hookflash doesn't work in featd |
| 19:20.38 | knight111 | just need to know how to debug before I make drastic change such as changing the OS that I use... |
| 19:20.47 | Corydon-w | I haven't used featd in a while |
| 19:20.56 | HCISjason | k |
| 19:21.15 | HCISjason | I know when we were trying to use the E&M wink signal he was seeing too many digits |
| 19:21.36 | Corydon-w | How many digits too many? |
| 19:24.35 | hmmhesays | nine gajillion |
| 19:24.53 | hmmhesays | i'm thinking about converting a honda to an EV |
| 19:25.10 | hmmhesays | how badass would that be |
| 19:26.48 | *** join/#asterisk Assid (i=assid@59.183.42.129) |
| 19:27.06 | HCISjason | Corydon-W: It sent Last 5 of ANI#DNIS#ANI#DNIS# |
| 19:27.34 | flujan | Corydon-w, I am already using tcp/ip sockets... |
| 19:27.41 | flujan | Corydon-w, :( |
| 19:28.08 | *** join/#asterisk asdx (n=diego@200.61.236.54) |
| 19:28.20 | rene1 | why would IAX2 wouldnt pass callerid to a ISDN PRI where SIP does |
| 19:28.21 | Corydon-w | flujan: so you tried both TCP/IP sockets as well as local Unix socket? |
| 19:28.40 | rene1 | and can i use setcallerpres would do any difference |
| 19:28.40 | Assid | damn.. i want a free DID |
| 19:28.46 | Corydon-w | HCISjason: I don't understand |
| 19:28.59 | flujan | Corydon-w, but I need the TCP/IP sockets enabled... :) |
| 19:29.03 | flujan | this is the matter... :P |
| 19:29.16 | Corydon-w | flujan: you can have both enabled |
| 19:29.35 | flujan | Corydon-w, thanks so I will check it. :) |
| 19:30.57 | rene1 | flujan: i wouldnt put too much login in the database nor i would bound to an specific one but that is just me |
| 19:31.08 | rene1 | too much -> any |
| 19:31.33 | HCISjason | Corydon-w: We have the E&M_w working with the proper digits but we are still getting a hangup |
| 19:33.00 | *** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net) |
| 19:33.25 | Corydon-w | HCISjason: have you tried all the others available to you? I see that you've tried featd, have you tried featdmf? |
| 19:33.42 | HCISjason | Corydon-w: this is DTMF not MF |
| 19:34.20 | HCISjason | Corydon-w: Does Asterisk support hook flash? |
| 19:34.30 | Corydon-w | HCISjason: yes |
| 19:34.41 | Corydon-w | HCISjason: but the wink may be longer than Asterisk is expecting |
| 19:35.07 | Corydon-w | HCISjason: see the timing parameters in zapata.conf |
| 19:35.44 | flujan | rene1 me too... but my boss doesn't want another machine... :( |
| 19:37.06 | saftsack | if i have 2 lines on a device registered to asterisk which initiate calls on different ports, is asterisk then able to differ the two accounts? |
| 19:37.22 | *** join/#asterisk DaveCanoe (n=Dave@host14.dxstorm.com) |
| 19:40.44 | *** join/#asterisk alerios (n=alerios@201.244.173.92) |
| 19:44.24 | *** join/#asterisk Gr1ncheux (n=devine@AStDenis-105-1-15-130.w81-248.abo.wanadoo.fr) |
| 19:47.58 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 19:48.56 | hoobastooba | when changing an extension to type=peer i have an issue where periodically where when i try to dial, i get: Nov 21 12:47:33 NOTICE[4265]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default' |
| 19:49.27 | hoobastooba | but when i change it back to type=friend it dials correctly |
| 19:49.36 | hoobastooba | then when i change it back to type=peer it does it again. |
| 19:49.53 | hoobastooba | if i reboot the phone it works correctly for a while set as type=peer, but then after a while breaks again. |
| 19:50.31 | hoobastooba | here is my sip.conf entry for that phone. |
| 19:50.32 | hoobastooba | http://pastebin.ca/254344 |
| 19:50.50 | hoobastooba | i have tried with and without nat=yes. |
| 19:50.52 | hoobastooba | no affect. |
| 19:51.09 | AJaymn | Anyone use DIDx.org for DIDs? |
| 19:51.39 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 19:52.05 | hoobastooba | what could possibly be causing this issue? |
| 19:53.09 | hoobastooba | i have restarted asterisk and the issue persists. |
| 19:54.42 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 19:54.50 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-235-141.red.bezeqint.net) |
| 19:56.42 | *** join/#asterisk backblue (n=moo@87-196-64-183.net.novis.pt) |
| 19:58.04 | *** join/#asterisk strippe (n=strippe@ncmx3.tecinfo.com) |
| 19:59.29 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 20:03.12 | blitzrage | is RetryTime in callfiles in seconds, or minutes? |
| 20:04.44 | HCISjason | Corydon-w: do you know what the debounce timing is for? |
| 20:05.17 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 20:06.47 | *** join/#asterisk hyperthread (n=janusmis@c906f745.virtua.com.br) |
| 20:07.05 | Corydon-w | HCISjason: not really, no |
| 20:07.25 | hyperthread | hello all....can I have a TDM400P (with 4 FXO) and a TE110P in the same system ? |
| 20:09.18 | *** join/#asterisk Dr-Linux (n=nono@202.59.73.131) |
| 20:09.21 | Corydon-w | hyperthread: sometimes, yes |
| 20:09.49 | hyperthread | Corydon-w, sometimes ? rsrsrs |
| 20:10.01 | Corydon-w | The motherboard matters |
| 20:10.23 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool1-11.mtco.com) |
| 20:10.29 | Corydon-w | Sometimes the motherboard will object strenuously |
| 20:10.40 | hyperthread | ok....its just fine....the problem its in zaptel.conf.... |
| 20:11.08 | Corydon-w | What's the problem? |
| 20:11.17 | hyperthread | there is a order to configure the cards ? |
| 20:11.34 | Corydon-w | Yes, they configure in the same order as the drivers are probed |
| 20:11.57 | Corydon-w | so if you probe wctdm first, then the TDM400 card is first |
| 20:12.20 | Corydon-w | otherwise the TE110P is first |
| 20:12.22 | strippe | I am expecting a call from either Laura clark, Dr. Karen Fox, or Dr. Brook w/ Delta Health Alliance or Miss. State University |
| 20:12.37 | *** part/#asterisk strippe (n=strippe@ncmx3.tecinfo.com) |
| 20:12.40 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 20:13.00 | hyperthread | hmmm.....good....so my TE110P (E1 configuration) its the first to load, so in zaptel.conf the channels are: 1-15,17-31 and 32-36 (to TDM400) right ? |
| 20:13.27 | *** join/#asterisk Luzer (n=Air_idea@c-24-12-5-97.hsd1.il.comcast.net) |
| 20:13.28 | Corydon-w | 32-35 |
| 20:13.48 | Corydon-w | 32-36 is 5 channels |
| 20:13.55 | hyperthread | ok...right...sorry... |
| 20:14.17 | Corydon-w | Also, do you have the jumper set? |
| 20:14.33 | hyperthread | E1/T1 ? yes |
| 20:14.53 | Corydon-w | sounds about right |
| 20:14.53 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 20:15.44 | hyperthread | let me try it here... |
| 20:20.38 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:20.52 | fatpelt | anyone know if there is a way to have a dialplan dial two extensions, then bridge them together? |
| 20:21.08 | brodiem | does anyone know if * 1.4 has any wideband codecs or has the support to add any? |
| 20:21.35 | hyperthread | seens that when I load wcte11xp, its try to configure to TDM400 |
| 20:21.57 | hyperthread | ZT_CHANCONFIG failed on channel 32: No such device or address (6) |
| 20:22.40 | ramtha | is t.38 passthrough in 1.4 also working over iax? sip client - asterisk - (iax2) - asterisk - (sip) - provider |
| 20:24.52 | *** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net) |
| 20:25.03 | *** join/#asterisk merbanan (n=banan@c80-216-155-218.bredband.comhem.se) |
| 20:25.50 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 20:27.28 | brodiem | does anyone know if * 1.4 has any wideband codecs or has the support to add any? |
| 20:29.07 | AJaymn | anyone use DIDx.org? |
| 20:29.14 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:29.33 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 20:34.56 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-114-42-210.satx.res.rr.com) |
| 20:35.07 | nortex | brodiem, Support was added to 1.4 for Polycom's HD phone, which is G.722 |
| 20:35.20 | AJaymn | has anyone implemented E911 without a POTS line? |
| 20:35.33 | blitzrage | I don't think wideband codec support is quite ready in 1.4 |
| 20:35.38 | blitzrage | last I heard anyways |
| 20:35.41 | blitzrage | AJaymn: yah, I have |
| 20:35.52 | Juggie | no wideband codec that i know of |
| 20:36.02 | Juggie | blitzrage, go leafs! |
| 20:36.05 | blitzrage | w00t! |
| 20:36.14 | Juggie | are you getting me tickets for xmas :P |
| 20:36.22 | blitzrage | Juggie: ummm... yaaaaaaaaaaaaahhhhhhhhhhhh.................... |
| 20:36.32 | Juggie | sundin will be back saturday |
| 20:36.36 | nortex | I understood G.722 to be wideband, and it was working at Astricon. |
| 20:36.37 | blitzrage | meh |
| 20:36.54 | Juggie | nortex, i'm afraid you have misunderstood |
| 20:36.58 | blitzrage | nortex: I think the passthrough might be working, but not sure the prompts got fixed yet... but I could be wrong. It would be all fixed now. |
| 20:37.12 | blitzrage | It might be all fixed now* |
| 20:37.26 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 20:37.41 | Juggie | so far as i know, * understands nothing past 8khz |
| 20:37.58 | nortex | Ok, I see what you mean, I saw the working passthrough. |
| 20:38.00 | Juggie | there was talk @ *-con about it, but i dont think anything has happened since. |
| 20:38.11 | blitzrage | Juggie: kpfleming implemented G.722 support at AstriCon |
| 20:38.21 | Juggie | blitzrage, for passthrough only? |
| 20:38.26 | blitzrage | yah |
| 20:38.29 | blitzrage | and prompts |
| 20:38.40 | AJaymn | blitzrage: what/who are you using for E911? |
| 20:38.42 | Juggie | so then it transcodes it? |
| 20:39.01 | blitzrage | AJaymn: sorry, that's priviledged |
| 20:39.06 | blitzrage | Juggie: no... it does passthrough :) |
| 20:39.13 | blitzrage | prompts is native |
| 20:39.18 | Juggie | ah. |
| 20:39.27 | Juggie | must only work within sip channel though? |
| 20:39.36 | blitzrage | I think it works with all channels afaik |
| 20:39.45 | *** join/#asterisk sremington (n=sremingt@rrcs-24-123-243-44.central.biz.rr.com) |
| 20:39.45 | blitzrage | I'd assume at least IAX2 and SIP |
| 20:39.53 | *** join/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net) |
| 20:40.00 | Juggie | hmmmmm |
| 20:43.00 | Juggie | hmmm |
| 20:43.02 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 20:43.06 | Juggie | i have a box thats been up for 5 weeks |
| 20:43.09 | Juggie | with no leaked channels |
| 20:43.20 | b11d | is that good? |
| 20:43.31 | Juggie | is probally because i dont use sip on that box. |
| 20:43.35 | EmleyMoor | I need to recompile wctdm for a different debounce |
| 20:43.45 | b11d | I was just going to say that you needed to do that |
| 20:43.58 | EmleyMoor | Unfortunately, it is proving difficult to work out what to do as I'm using Debian packages |
| 20:44.15 | b11d | screw packages.. |
| 20:44.19 | b11d | source all the way! |
| 20:44.40 | b11d | I dont even use Makefile's.. thats how hardcore I am |
| 20:44.46 | b11d | :P |
| 20:45.40 | justinu|laptop | either that or you have waaay too much free time |
| 20:45.41 | justinu|laptop | :P |
| 20:45.46 | b11d | haha |
| 20:45.57 | b11d | I do remember the pre-make days though. |
| 20:46.15 | justinu|laptop | make is ancient |
| 20:46.18 | justinu|laptop | how old are you?? |
| 20:46.35 | b11d | im 26.. been into computers since 1983 |
| 20:46.59 | justinu|laptop | i think make has been around since the early 70s |
| 20:47.01 | florz | b11d: What digits do those dots stand for? |
| 20:47.07 | dlynes_laptop | make is outdated now, anyways, with the advent of ant |
| 20:47.08 | b11d | dots? |
| 20:47.12 | justinu|laptop | even ant is outdated |
| 20:47.15 | justinu|laptop | check out maven |
| 20:47.20 | b11d | no way make is THAT old. |
| 20:47.39 | florz | b11d: Yes, the two "." characters after those "2" and "6" characters ... |
| 20:47.48 | dlynes_laptop | yeah...make's been around since the early to mid 70's |
| 20:47.53 | b11d | oh hahaha.. :P |
| 20:47.58 | florz | =:-) |
| 20:48.06 | b11d | ok. then I remember the "what I thought was pre-make days" |
| 20:48.14 | b11d | I was thinking of 90/91 |
| 20:48.22 | b11d | maybe i'm thinking of Configure. |
| 20:48.27 | justinu|laptop | b11d: my bad, first make was released in 77 |
| 20:48.35 | b11d | still.. the 70's |
| 20:48.37 | justinu|laptop | ok, so you mean autoconf, autotools |
| 20:48.46 | b11d | yeah |
| 20:48.48 | dlynes_laptop | I think gnu autoconfigure came out around 95 or 96 or something |
| 20:49.05 | X-Rob | nah |
| 20:49.07 | dlynes_laptop | Linux came onto the scene in 1993 |
| 20:49.07 | X-Rob | before then |
| 20:49.15 | b11d_pc | yeah |
| 20:49.28 | X-Rob | I was using autoconf on IRIX machines in 1992 |
| 20:49.36 | X-Rob | and sequent machines |
| 20:49.41 | b11d | ahh IRIX.. |
| 20:49.44 | dlynes_laptop | ok...thought it didn't exist until after linux came onto the scene |
| 20:50.17 | justinu|laptop | it was C that pretty much debuted in 72 |
| 20:50.23 | b11d | :) |
| 20:52.08 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 20:52.20 | b11d | are there any good utils out for parsing CDR's? |
| 20:52.41 | dlynes_laptop | Here's a cool tool for the next time you're returning your videos to the movie rental place: http://img.7chan.org/b/src/116414280456348765.jpg |
| 20:53.18 | b11d | heh.. looks like one of those mini-helicopter launching pads painted up |
| 20:53.25 | b11d | someone would be dumb enough :) |
| 20:53.46 | justinu|laptop | asterisk-stat is kinda handy |
| 20:53.49 | justinu|laptop | but basic |
| 20:53.57 | b11d | i'll take a look at that |
| 20:54.00 | justinu|laptop | needs mysql or postgres tho |
| 20:54.00 | b11d | tks |
| 20:54.03 | justinu|laptop | don't think it parses the csv |
| 20:54.06 | b11d | ah. yeah.. |
| 20:54.19 | b11d | maybe I should just bite it and go with mysql |
| 20:54.34 | justinu|laptop | not a bad idea, a lot of the billing packages out there want it |
| 20:54.43 | b11d | i just didnt want another peice of software in the chain to fail.. |
| 20:54.49 | b11d | especially when flat text will do.. |
| 20:54.51 | b11d | but.. |
| 20:54.54 | justinu|laptop | yeah, but you can log to both |
| 20:54.57 | b11d | now's the time to do this.. |
| 20:55.01 | dlynes_laptop | justinu|laptop: mcc uses postgresql |
| 20:55.10 | b11d | I'd imagine if I go with mysql for cdr, i'll put my extensions & vm shit in there too |
| 20:55.15 | justinu|laptop | postgres module is smart enough to store CDRs, then postthem if the db connection goes down |
| 20:55.17 | dlynes_laptop | justinu|laptop: it's actually pretty good, after a few code mods and some database hacks |
| 20:55.20 | justinu|laptop | not sure how well that works tho |
| 20:55.35 | b11d | there is a Parse::CSV module for perl.. i'll take a look at that too |
| 20:55.37 | justinu|laptop | i prefer postgres, because i'm an ex-oracle guy |
| 20:55.46 | justinu|laptop | but lots of ppl use mysql, and i'm cool with that |
| 20:55.49 | b11d | fair enough.. |
| 20:55.56 | dlynes_laptop | justinu|laptop: mcc is written for the european market, so it doesn't work well without some changes for north america |
| 20:56.32 | b11d | http://www.areski.net/areski/index.php?option=com_content&task=view&id=21&Itemid=52 |
| 20:56.36 | b11d | this guy's grandmother rules |
| 20:56.37 | *** part/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 20:56.46 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 20:57.15 | justinu|laptop | lol |
| 20:57.28 | justinu|laptop | areski is a good guy |
| 20:59.13 | blitzrage | wow... that's a load of beer |
| 21:02.07 | *** join/#asterisk A-Tuin (n=a-tuin@213.237.228.225) |
| 21:03.02 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 21:05.18 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:07.30 | b11d | yeah |
| 21:07.32 | b11d | its great |
| 21:07.37 | dlynes_laptop | damn..i feel sorry for his cat |
| 21:07.52 | b11d | although his grandmother seems to like head |
| 21:08.01 | b11d | yeah, that poor kitty |
| 21:09.51 | *** join/#asterisk conico (n=chatzill@85.107.214.105) |
| 21:11.28 | e-milio | helloa ll |
| 21:11.48 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 21:11.49 | e-milio | is this correct to dial with a long distance code: _1NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}${PBXCODE} ) |
| 21:12.06 | *** join/#asterisk Mportnoy (n=test@200.122.158.88) |
| 21:12.42 | e-milio | fello asteriskians ??? |
| 21:15.00 | sahafeez | any good voip phone that is wireless? i guess i mean cordless. can have a base station. |
| 21:15.52 | justinu|laptop | not really... get a POTS cordless and an ATA |
| 21:16.11 | blitzrage | e-milio: looks fine |
| 21:16.20 | CunningPike | sahafeez: Aastra are bringing out a DECT cordless with a SIP base station in the Spring |
| 21:16.29 | sahafeez | hum. no. i have a pure voip network with all voip phomes |
| 21:16.29 | blitzrage | the stuff in the Dial() will depend on your dialplan and such |
| 21:16.31 | sahafeez | phones |
| 21:16.52 | blitzrage | sahafeez: that's what the ATA is for |
| 21:16.59 | synthetiq | meetme gurus: how come when accessing the confernce room it does not ask if you are an adminstrator enter such and such password? |
| 21:17.05 | CunningPike | Phomes - Smon phones? :) |
| 21:17.08 | sahafeez | ok. good ata? |
| 21:17.14 | CunningPike | s/Smon/Snom/ |
| 21:17.32 | synthetiq | meetme gurus: or do i need to creat a seperate extension with the same channel using aA mose? |
| 21:17.44 | synthetiq | s/mose/mode |
| 21:17.52 | justinu|laptop | dect looks cool |
| 21:18.15 | e-milio | blitzrage: The idea is that for long distance dialing |
| 21:18.31 | sahafeez | also if using an ATA hows does that fit in to transfers, voicemail, etc.. |
| 21:18.36 | e-milio | blitzrage: an "access code" from the pstn needs to be dialed |
| 21:18.53 | e-milio | blitzrage: is to control ld dial |
| 21:19.13 | blitzrage | e-milio: where are you getting the code from? |
| 21:19.26 | blitzrage | e-milio: you're not showing enough based on what you just said -- use pastebin and someone might be able to help |
| 21:19.46 | blitzrage | you're setting some variables, but I have no idea what the trunk expects, or what is set in those variables |
| 21:19.54 | e-milio | blitzrage: [globals] |
| 21:20.00 | e-milio | PBXCODE=12345 |
| 21:20.26 | e-milio | TRUNKMSD=1 |
| 21:20.26 | e-milio | TRUNK=Zap/G1 |
| 21:20.26 | e-milio | PBXCODE=78358 |
| 21:20.31 | blitzrage | ok... so the softswitch or whatever you're dialing needs to handle that... and I still have no idea if that would work |
| 21:21.34 | e-milio | blitzrage: basically my question is if it is on the correct syntax, I need to dial after the EXTEN a pause and the code |
| 21:22.54 | blitzrage | e-milio: the 'D' option might do it -- what you're doing is just going to dial the whole string to the trunk |
| 21:23.32 | blitzrage | so if EXTEN is 18005551212 and PBXCODE is 12345, you're dialing: 800555121212345 (assuming TRUNKMSD=1) |
| 21:23.42 | dlynes_laptop | CunningPike: what's wrong with their sip base station with the spread spectrum handsets? |
| 21:24.54 | CunningPike | dlynes_laptop: Range, and you're tied to a handset. Their DECT solution will purportedly work like mesh WiFi - you can transfer from one base station to another mid-call |
| 21:24.54 | dlynes_laptop | sahafeez: The Aastra 480i CT can have up to four spread spectrum cordless handsets...they cover up to a 30K square foot warehouse |
| 21:25.05 | dlynes_laptop | CunningPike: ah |
| 21:25.08 | sahafeez | k thanks |
| 21:25.34 | CunningPike | Have anyone else's hints stopped working on 1.4beta3? |
| 21:25.48 | sahafeez | i am just looking for something for a home users that has vpn to my network that they can walk with |
| 21:26.04 | dlynes_laptop | CunningPike: my hints don't work properly on 1.2 |
| 21:26.19 | CunningPike | dlynes_laptop: Mine do^h^h did |
| 21:26.27 | dlynes_laptop | CunningPike: damned aastra phones keep 500/487'ing on hints |
| 21:26.45 | e-milio | blitzrage: When I dial out what happens next is that the call is transfered to a queue cause a agi script will be executed. I use background detect to check for ans mach |
| 21:26.51 | CunningPike | dlynes_laptop: There isn't even a SIP message sent |
| 21:26.58 | blitzrage | e-milio: ok |
| 21:27.05 | dlynes_laptop | CunningPike: cool |
| 21:27.29 | e-milio | blitzrage: Do you know if D will interfere with Background detect after answered ? |
| 21:27.34 | blitzrage | no idea |
| 21:27.55 | e-milio | blitzrage: thanks, will play with it. |
| 21:28.47 | *** join/#asterisk jmacz (n=jmacz@201.244.173.92) |
| 21:28.53 | jmacz | asterisk-dev |
| 21:28.57 | CunningPike | 'show hints' shows the hints, and the correct number of watchers. but the status doesn't change from 'Idle" |
| 21:34.59 | blitzrage | Juggie: can the Mitel 5220's register to more than one line? |
| 21:37.40 | *** join/#asterisk ltd (n=z@202-161-1-61.dyn.iinet.net.au) |
| 21:40.18 | *** join/#asterisk jax0m (n=iam@65.106.29.154.ptr.us.xo.net) |
| 21:40.20 | jax0m | hi guys |
| 21:40.44 | jax0m | do you know if i would be able to put two Wildcard TDM04B (http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=RTDM04B) in the same machine side by side with no issues? |
| 21:43.12 | EmleyMoor | jax0m: Some motherboards don't like it |
| 21:43.35 | EmleyMoor | Anyone on FWD prepared to accept a call, or to call me? |
| 21:45.01 | X-Rob | 47846 should work |
| 21:45.23 | b11d | you got that fixed then EmleyMoor? |
| 21:45.27 | b11d | what was the issue? |
| 21:45.42 | EmleyMoor | X-Rob: failed |
| 21:45.47 | EmleyMoor | I'm on 794933 |
| 21:45.50 | *** join/#asterisk RoyK (n=roy@216-99-29.0506.adsl.tele2.no) |
| 21:46.26 | X-Rob | sorry, 47876 |
| 21:47.48 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 21:48.58 | X-Rob | callerid came through as 'asterisk' |
| 21:49.34 | EmleyMoor | Yes - having trouble setting caller ID for FWD at present |
| 21:49.47 | EmleyMoor | Phone is a 1971 GPO Yeoman BTW |
| 21:50.42 | jax0m | EmleyMoor: thanks for the advice |
| 21:51.59 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:52.28 | EmleyMoor | It will be better when my replacement FXO module arrives |
| 21:55.27 | EmleyMoor | Still, at least I got my rotary phone working |
| 21:58.56 | EmleyMoor | Presumably, if I wish to do things that normally need * or # from the rotary phone, I need to program rotary-dialable alternatives? |
| 22:00.14 | EmleyMoor | (partner's comment: "Oh, you're not dialing, are you?" |
| 22:00.17 | EmleyMoor | ) |
| 22:00.17 | [TK]D-Fender | EmleyMoor : Why exactly is ti again that your using a rotary phone at all? |
| 22:01.14 | EmleyMoor | [TK]D-Fender: I like the ring and it's the only working table phone I've got - need it for by my bed in case of emergencies |
| 22:01.55 | [TK]D-Fender | EmleyMoor....... um... thats just SAD... |
| 22:02.26 | EmleyMoor | The Yeoman is a bit of a classic, though - besides, this one's amplified so I could pass it to David if I need to |
| 22:02.38 | [TK]D-Fender | EmleyMoor : Your time has got to be worth more thant he 10$ it'd cost to buy a basic DTMF phone.... Heck at that price I'd likely offer CID while you're at it ;) |
| 22:03.01 | EmleyMoor | I'll usually be on softphones anyway |
| 22:04.01 | [TK]D-Fender | EmleyMoor : Ick.. you're all shades of masochist it seems. |
| 22:04.22 | EmleyMoor | I actually bought this phone for the amplified handset, but couldn't get the other parts I needed to fit it to the DTMF wall version of same model I have |
| 22:04.43 | EmleyMoor | (the DTMF version needs the amplifier to be battery powered) |
| 22:05.08 | [TK]D-Fender | EmleyMoor : This is a good place for analog zaptel channels since you can rape the gains direct. That'd do it on any ophone.. |
| 22:06.49 | EmleyMoor | I could turn my two versions of this phone into a goofy wall amplified rotary and a plain table DTMF if I wanted to |
| 22:07.37 | justinu|laptop | i want a candlestick phone |
| 22:08.02 | EmleyMoor | justinu|laptop: I toyed with that idea - the bellset put me off |
| 22:08.14 | brodiem | does anyone know if * 1.4 has any wideband codecs or has the support to add any? |
| 22:08.21 | EmleyMoor | Besides, GPO 100s are expensive |
| 22:09.56 | [TK]D-Fender | justinu|laptop : It was Mr. Green in the lobby! |
| 22:10.47 | *** join/#asterisk jm|work (n=jamiem@zen.jamiem.com) |
| 22:12.25 | EmleyMoor | In zap show channel x, Pulse phone only shows yes when there's a call - perhaps not surprising really but slightly odd |
| 22:12.27 | file | yay circles |
| 22:12.43 | *** join/#asterisk alamantia (i=alamanti@nat/digium/x-0c350c5e04391915) |
| 22:14.28 | [TK]D-Fender | file : They're the new "square" :) |
| 22:15.33 | EmleyMoor | "Please dial the number you require, followed by circle." to paraphrase BT |
| 22:19.28 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 22:21.13 | *** join/#asterisk d4rkstar (n=d4rkstar@ip-232-6.sn2.eutelia.it) |
| 22:21.18 | d4rkstar | hi to all |
| 22:21.29 | *** join/#asterisk xnon (i=xnon@200.8.30.3) |
| 22:21.41 | d4rkstar | can i post here a question? |
| 22:21.58 | [TK]D-Fender | d4rkstar : Sure |
| 22:22.14 | d4rkstar | thank you |
| 22:22.33 | d4rkstar | i've a samsung wip-5000 ip phone |
| 22:22.55 | d4rkstar | and would like to know if it can operate in a wireless environment with an asterisk pbx |
| 22:23.10 | d4rkstar | or better.. if someone have been able to make it work |
| 22:23.29 | *** join/#asterisk asdx (n=diego@200.61.236.54) |
| 22:24.05 | [TK]D-Fender | d4rkstar : Don't know about samsung, but I have use a Hitachi with the same model #. |
| 22:24.32 | d4rkstar | yes, i know, but they're two different products |
| 22:24.43 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:26.51 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 22:26.51 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 22:29.50 | *** join/#asterisk designdream (n=felipe@rrcs-71-40-49-30.sw.biz.rr.com) |
| 22:31.49 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:32.40 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 22:34.32 | *** join/#asterisk xnon_ (i=xnon@200.8.30.3) |
| 22:34.44 | d4rkstar | ok bye all |
| 22:34.48 | d4rkstar | it's late for me |
| 22:34.49 | d4rkstar | :) |
| 22:35.16 | hoobastooba | i am trying to dial via iax to another server....Dial(IAX2/REC_SERVER_Local_External_Dialer:spherous456@10.0.1.189/${EXTEN}) I get the following error: |
| 22:35.29 | hoobastooba | ahh... lost it... back in a sec with it. |
| 22:35.55 | hoobastooba | Rejected connect attempt from 10.0.1.180, who was trying to reach '8018285230@' |
| 22:36.17 | hoobastooba | i have added the context into the iax.conf |
| 22:36.25 | *** join/#asterisk asdx (n=diego@200.61.236.54) |
| 22:38.05 | b11d | i love you guys |
| 22:41.40 | asdx | i love you guys too |
| 22:42.25 | b11d | Dammit.. I'm going to have to write my own CDR parser.. |
| 22:42.30 | b11d | I knew it was going to come to this.. |
| 22:42.50 | brodiem | b11d it's already in a db/csv :).. |
| 22:42.52 | b11d | time to MASTER PERL.. |
| 22:42.53 | b11d | ..again |
| 22:42.57 | b11d | what is? |
| 22:43.04 | brodiem | cdrs |
| 22:43.07 | sevard | I have a couple of DIDs I'm sending to a "disconnected" context off of my PRI, I can't find a "dialed number" variable in Asterisk (i'd like to SayDigits(number)) |
| 22:43.12 | sevard | does anyone konw of one? |
| 22:43.15 | b11d | I need something that will break up a CDR into a bill per-number. |
| 22:43.16 | sevard | or am I going to have to create another macro |
| 22:43.33 | b11d | there is one.. |
| 22:43.34 | b11d | hang on |
| 22:43.43 | sevard | there is ${DIALEDPEERNUMBER} |
| 22:43.46 | sevard | but that's busted |
| 22:43.54 | b11d | oh |
| 22:44.00 | b11d | that was it |
| 22:44.02 | b11d | :) |
| 22:44.27 | sevard | doesn't do a single thing if I do exten = s,n,SayDigits(${DIALEDPEERNUMBER}) |
| 22:45.08 | brodiem | sevard just to test do a NoOp(${DIALEDPEERNUMBER}) |
| 22:45.10 | b11d | i had a workaround for that, but I dont remember wtf I did.. |
| 22:45.10 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 22:45.22 | sevard | just passes "" |
| 22:45.23 | brodiem | see if anything is contained in there |
| 22:45.24 | brodiem | oh |
| 22:45.36 | sevard | :) |
| 22:46.15 | *** join/#asterisk lordcad (n=lordcad@20151177184.user.veloxzone.com.br) |
| 22:46.26 | bluregard | [TK]D-Fender: you around? |
| 22:47.49 | brodiem | sevard, so in your dial plan you'll know which DID was dialed right? |
| 22:47.59 | lordcad | Hi all, |
| 22:47.59 | lordcad | <PROTECTED> |
| 22:47.59 | lordcad | I have one interface zaptel TE110P with 10 channels enabled for my service provider. Alone the first channel is enable to receive call from this service provider and when a make call using other channel, like Zap/2, the channel is enable from the PBX Siemens EWSD version 10 to receive every calls. |
| 22:47.59 | lordcad | <PROTECTED> |
| 22:48.00 | lordcad | The support off my service provider, send to me the follow message that receive in their PABX: First DBAC e the last BPRN. |
| 22:48.05 | lordcad | <PROTECTED> |
| 22:48.07 | lordcad | somebody already had this problem and could help me? |
| 22:48.09 | lordcad | <PROTECTED> |
| 22:48.11 | lordcad | Tks, |
| 22:48.16 | lordcad | lordcad |
| 22:48.26 | sevard | brodiem: I do |
| 22:48.32 | brodiem | lordcad this isn't an e-mail client lol |
| 22:48.44 | sevard | brodiem: I guess I could make a macro and pass the variable |
| 22:48.47 | sevard | hahahaha |
| 22:48.48 | asdx | what distro do you use with asterisk? |
| 22:48.55 | sevard | asdx: Slackwaer. |
| 22:48.58 | brodiem | sevard yeah |
| 22:48.59 | sevard | *Slackware. |
| 22:49.03 | brodiem | slack here also |
| 22:49.08 | sevard | brodiem: niiice, |
| 22:49.14 | sevard | it seems really lame that i'd have to make a marco |
| 22:49.20 | brodiem | lol |
| 22:49.21 | sevard | asterisk knows the extension |
| 22:49.26 | sevard | it has to |
| 22:49.28 | lordcad | ok |
| 22:49.34 | b11d | i give props to slackware.. their website hasnt changed in like 10 years. |
| 22:49.39 | sevard | word. |
| 22:49.42 | lordcad | but i need a help |
| 22:49.49 | sevard | how many helps did you need again? |
| 22:49.52 | brodiem | sevard ${EXTEN} would correspond with whatever is being passed as a DID |
| 22:49.57 | lordcad | you can help-me ? |
| 22:49.57 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:50.59 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 22:51.12 | asdx | i want to try out asterisk, but it will work also if i don't have a fxo card? |
| 22:51.15 | *** join/#asterisk jax0m (n=iam@65.106.29.154.ptr.us.xo.net) |
| 22:51.17 | asdx | i just want to use software atm |
| 22:51.33 | asdx | cos i don't have a fxo card |
| 22:51.47 | b11d | sure |
| 22:51.47 | b11d | pure voip.. no problem |
| 22:51.58 | asdx | alright |
| 22:52.02 | sevard | brodiem: but if the DID says "go to disconnected,s,1" the ${EXTEN} is 's' |
| 22:52.07 | fatpelt | grr.... my tdm400 isn't hanging up an analog line with the remote end disconnects. what am i doing wrong here? |
| 22:52.16 | b11d | how long is the delay? |
| 22:52.30 | b11d | sure you have the correct signalling type in zaptel? |
| 22:52.32 | brodiem | sevard oh ok I was thinking you would just put it in the same context |
| 22:52.35 | hmmhesays | fun disconnect supervision problems |
| 22:52.49 | brodiem | sevard just do a SetVar and macro then :) |
| 22:52.54 | sevard | Bah! |
| 22:52.59 | b11d | Bosh Flimshaw! |
| 22:53.02 | brodiem | i don't know why you're so against that lol |
| 22:53.11 | sevard | brodiem: it's unwanted bulk |
| 22:53.14 | fatpelt | i'm using fxs_ks. works for everything but the disconnect, and i can't seem to phrase a google right |
| 22:53.17 | sevard | i like trim and neat configuration files |
| 22:53.23 | brodiem | you should see my dial plans then, hahah |
| 22:53.24 | b11d | try gs? |
| 22:53.47 | fatpelt | b11d: i need to reload the zaptel modules right? ( i have been in the past ) |
| 22:53.51 | b11d | yeah im pretty sure |
| 22:53.52 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 22:54.16 | b11d | Is there no free log parser for Asterisk CDRs? |
| 22:54.17 | hmmhesays | anyone set up squid before? |
| 22:54.22 | b11d | without using the DB |
| 22:54.26 | b11d | I run a few squid proxies |
| 22:54.39 | sevard | hmmhesays: almost |
| 22:55.21 | sevard | brodiem: this has to be set somewhere by default, it seems silly it's not |
| 22:55.39 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 22:55.50 | brodiem | sevard guess you'll just have to take the plunge and set it yourself:) |
| 22:56.12 | sevard | that the dialed number is not setup in a variable in the channel's varspace |
| 22:56.15 | brodiem | but yea i agree it should somewhere |
| 22:56.22 | hmmhesays | b11d: you ever use digest auth? |
| 22:56.28 | hmmhesays | I need different acl's per user |
| 22:56.28 | b11d | yeah, with samba |
| 22:56.35 | hmmhesays | with a total of 3 users |
| 22:56.45 | b11d | :| |
| 22:56.45 | b11d | good luck :) |
| 22:56.47 | hmmhesays | can't do it? |
| 22:56.54 | b11d | i just dont know if you can or not |
| 22:57.02 | hmmhesays | that would be the whole point of digest auth |
| 22:57.13 | b11d | i just used it to auth against a PDC. |
| 22:57.15 | ucfMethod | anyone want to buy (3) Cisco IP 7960 phones? |
| 22:57.29 | ucfMethod | I moved everyone to Polycom 501s |
| 22:57.40 | b11d | that's satisfying |
| 22:57.52 | ucfMethod | figured I would let you guys have first dibs before craigslist/ebay gets them |
| 22:58.37 | hmmhesays | ucfMethod: how much? |
| 22:59.01 | fatpelt | b11d: hmm. ztcfg now complains that i've got it configured wrong. i've got 4 fxo modules zaptel.conf says fxsgs=1-4 and zapata.conf says signalling=fxs_gs |
| 22:59.52 | b11d | hahahahha fatpelt.. thats the best nick ever |
| 23:00.08 | fatpelt | :) |
| 23:00.11 | b11d | i dunno what to say other than that. |
| 23:00.30 | fatpelt | b11d: i thought it had it correct... grrr |
| 23:00.40 | b11d | hahah fatpelt.. im going to name my Asterisk CDR parser fatpelt |
| 23:01.00 | b11d | i cant stop chuckling due to that name |
| 23:01.07 | fatpelt | :D |
| 23:01.18 | b11d | I wish I had some advice though.. |
| 23:01.31 | fatpelt | so, zaptel.conf is used by the modules and zapata.conf is used by asterisk proper right? |
| 23:01.44 | hmmhesays | i always thought furious george was a good name too |
| 23:01.54 | b11d | yeah that is a good one.. |
| 23:02.05 | b11d | justinu is also good |
| 23:02.22 | hmmhesays | thats the dudes name |
| 23:02.37 | b11d | oh.. i thought it was "just in u" |
| 23:02.40 | b11d | like.. sexually |
| 23:02.58 | hmmhesays | I think it is justin U |
| 23:03.00 | hmmhesays | lol |
| 23:03.05 | hmmhesays | sick bastard |
| 23:03.13 | b11d | :) |
| 23:03.20 | b11d | im from Northern Ontario.. what did you expect? |
| 23:03.23 | *** join/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 23:03.32 | hmmhesays | you're from canadia huh? |
| 23:03.35 | b11d | yep |
| 23:03.40 | *** part/#asterisk perlmonk (n=pierre@201.38.238.66) |
| 23:03.56 | fatpelt | http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf doesn't seem to think that there is a fxsgs option |
| 23:04.50 | b11d | it also doenst show loopstart... but its there |
| 23:05.36 | fatpelt | hmm. no complaints |
| 23:05.38 | fatpelt | crud |
| 23:06.11 | fatpelt | "ZT_CHANCONFIG failed on channel 1: Invalid argument (22)" is what ztcfg says |
| 23:06.32 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 23:06.39 | *** part/#asterisk sloth101 (n=mallen@69.71.225.45) |
| 23:07.17 | b11d | it wont be long.. im going to be working hard on fatpelt all night.. |
| 23:07.22 | b11d | then fatpelt will be released to the world.. |
| 23:07.58 | fatpelt | b11d: ?? |
| 23:07.58 | b11d | and we shall hear the collective sigh of relief from the 10 or so remaining Asterisk operators that want a CDR parser. |
| 23:07.58 | AJaymn | Anyone using DIDx.org ?? or have used there API for pulling numbers? |
| 23:07.58 | hmmhesays | what the hell |
| 23:08.02 | b11d | im going to name this bastasrd fatpelt.. |
| 23:08.09 | fatpelt | hehe |
| 23:08.26 | b11d | (root@jachin) % ./fatpelt -v |
| 23:08.29 | b11d | its coming.. |
| 23:08.30 | b11d | oh yeah |
| 23:09.50 | hmmhesays | wtf |
| 23:09.55 | hmmhesays | i'm to lazy to scroll up |
| 23:10.02 | b11d | well |
| 23:10.08 | *** join/#asterisk Zipper_32 (n=None@142.232.142.135) |
| 23:10.14 | b11d | since there is no Asterisk CDR CSV analyzer, im going to write one.. |
| 23:10.16 | b11d | in perl.. |
| 23:10.20 | b11d | and im going to name it "fatpelt" |
| 23:10.29 | b11d | because the nick made me chuckle over a 48 hour period |
| 23:10.35 | asdx | well, if i plan to use asterisk right now without a fxo card, only for voip, do i need softphones to use with it right? |
| 23:10.44 | b11d | they would help.. |
| 23:11.13 | GaVak | They haven't completed the RFC for telephony over telapathy yet? |
| 23:11.22 | EmleyMoor | Softphones or hard IP phones |
| 23:11.23 | GaVak | the ToT protocol. |
| 23:11.26 | Zipper_32 | I have an unusual problem where just recently, all SIP to SIP calls on my asterisk server have dropped dead with no voice travelling through. The phones ring, but no sound is present once either party speaks. IAX2 to SIP seems to work, but SIP to SIP does not. Does anybody have any recommendation where to begin troubleshooting? |
| 23:11.30 | b11d | its in RFC1925 |
| 23:11.51 | EmleyMoor | SIP or IAX for preference but H323, Skinny and UNISTIM are also supported |
| 23:12.12 | Zipper_32 | asdx: Or use an FXS and an analog phone. |
| 23:12.27 | GaVak | I hear the test models had a lot of echo problems with some users... Reports stated it sounded like an echo one would normally hear inside of a cavern. |
| 23:12.56 | asdx | Zipper_32: i don't have a FXS, just my computer and some other computers in the same LAN network |
| 23:13.33 | Zipper_32 | Then you'll be using softphones or hard IP phones. Just as EmleyMoor said, |
| 23:13.41 | EmleyMoor | asdx: If you don't want to spend money, use softphones. Hard IP phones would be a good move if you want to go further with no FXS card |
| 23:14.25 | asdx | is not that i don't want to spend money, i don't have money right now, but i'll buy some fxs/fxo and some phones in the future when i get some money ;p i just want to try out asterisk now :) |
| 23:14.37 | asdx | alright |
| 23:14.43 | EmleyMoor | Softphones will do fine for testing |
| 23:14.59 | asdx | ok thanks |
| 23:15.17 | Zipper_32 | asdx: FXO is for receiving a POTS line to your Asterisk Box. FXS is to provide your analog phones with a dialtone; be sure to know the difference if you order something. =) |
| 23:16.16 | asdx | Zipper_32: ok |
| 23:16.35 | asdx | i think i want a TDM400P |
| 23:16.42 | blitzrage | softphones: I recommend X-Lite and idefisk |
| 23:16.56 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:16.59 | EmleyMoor | asdx: That's the bare version with no modules |
| 23:17.01 | Zipper_32 | asdx: That's what I bought, with 4fxo, and lots of hard IP phones |
| 23:17.02 | blitzrage | actually, idefisk is now called zoiper I think |
| 23:17.07 | fatpelt | b11d: i assume that gs can't be turned off? eg. by a compile time option |
| 23:17.07 | EmleyMoor | I like moziax and ekiga |
| 23:17.43 | fatpelt | blitzrage: i downloaded idefisk earlier today |
| 23:17.43 | asdx | ekiga is a softphone? |
| 23:17.44 | blitzrage | wonder if the ekiga folks fixed a bug reported a while back where you can't used # in the username |
| 23:17.44 | EmleyMoor | Yes - a SIP or H323 video-capable one |
| 23:17.51 | blitzrage | I also like SJphone |
| 23:17.54 | sevard | Does anyone know if Asterisk comes with a sample of that Tri-Tone error sound usually found on a regular pstn error message |
| 23:18.04 | EmleyMoor | SJphone is nice - use that in 'doze! |
| 23:18.10 | blitzrage | fatpelt: use the built in transfer of asterisk to do it (look in features.conf) |
| 23:18.11 | sevard | DOOT DOOT DOOT - the number you have dialed has been disconnected, please hang up and try your call again. |
| 23:18.17 | sevard | that doot doot doot, the tri tone |
| 23:18.18 | EmleyMoor | sevard: Use the Zapateller() app |
| 23:18.23 | sevard | do we have a sample of that? |
| 23:18.27 | blitzrage | actually, Asterisk is itself a softphone :) |
| 23:18.29 | sevard | EmleyMoor: oo, Zapateller? |
| 23:18.37 | blitzrage | just dial from the CLI if you install chan_also or chan_oss |
| 23:18.44 | blitzrage | chan_alsa* |
| 23:18.49 | sevard | show application zapateller |
| 23:18.54 | EmleyMoor | It plays a bit fast for me but it appears to be the tri-tone |
| 23:18.58 | *** join/#asterisk unfo_ (n=QSECADM@CPE0004e2d9f884-CM0013718690da.cpe.net.cable.rogers.com) |
| 23:19.14 | unfo_ | hi all. wouldn't it be cool if asterisk came with some default configuration e.g. automatically pick up all calls on all FXO/FXS cards after 7 rings and play a default voicemail greeting and send the incoming messages as .OGG attachments to the email address of your choice? |
| 23:19.18 | sevard | EmleyMoor: i don't think that's what I'm looking for.. |
| 23:19.26 | sevard | hmm |
| 23:19.29 | EmleyMoor | sevard: Hmmm... fair enough |
| 23:20.19 | sevard | EmleyMoor: I think that might work :D |
| 23:20.27 | b11d | goodnight lads |
| 23:20.34 | Corydon-w | ogg takes up too much CPU to code in realtime at the same time as other channels |
| 23:20.37 | blitzrage | unfo_: sure would! let me know when you build it :) |
| 23:20.41 | EmleyMoor | I tried it |
| 23:20.59 | unfo_ | blitzrage: :-) |
| 23:21.16 | sevard | EmleyMoor: I wonder if using it in that method is approved |
| 23:21.31 | unfo_ | Corydon-w: so you could code later on, when the machine is idle. |
| 23:21.50 | unfo_ | blitzrage: is it possible to get such a config just by editing config files? |
| 23:22.03 | Corydon-w | unfo_: many Asterisk installations are never idle |
| 23:22.09 | fatpelt | ok, so since i can't seem to get my kernel to use fxs_gs what other possible things could be wrong with asterisk not detecting disconnect on my analog line? |
| 23:22.10 | ucfMethod | anyone know how to get ODBCquery() to work in 1.2.13? |
| 23:22.23 | Zipper_32 | unfo_: Yes it is possible. |
| 23:22.45 | Corydon-w | ucfMethod: what's that? |
| 23:23.08 | ucfMethod | this is the error im getting... "pbx.c:1700 pbx_extension_helper: No application 'ODBCquery' for extension (altnum, 3052, 1)" I am trying to migrate my extensions.conf from 1.0 to 1.2.13 |
| 23:23.08 | unfo_ | Zipper_32: so if I file a Debian bug requesting a useful default config, do you think my wish would be implemented? |
| 23:23.28 | Corydon-w | ucfMethod: where'd you find that? |
| 23:23.33 | Zipper_32 | How is what you are requesting 'A Debian Bug'? |
| 23:24.19 | unfo_ | Zipper_32: a wishlist request in the Debian bug tracker |
| 23:24.20 | Corydon-w | ucfMethod: that's not a standard Asterisk application |
| 23:24.20 | Zipper_32 | ucfMethod: That's a huge jump man. You may want to take the migration in small peices and please check to see if some of your commands are still functioning in 1.2.13 |
| 23:24.27 | ucfMethod | Corydon-w: in my extensions.conf, and it works just fine in Asterisk CVS-v1-0 but Asterisk 1.2.13 doesnt know what it is |
| 23:24.37 | unfo_ | Zipper_32: perhaps I would forward it to asterisk bugs too. |
| 23:25.02 | ucfMethod | Zipper_32: most things are migrating fine, we don't have a massive complicated extensions.conf |
| 23:25.04 | Zipper_32 | ucfMethod: The jump from 1.0 to 1.2.13 is huge. Take it in small peices and it you should figure out which command is broken. |
| 23:25.25 | Zipper_32 | unfo_: How much asterisk experience do you have? |
| 23:25.33 | unfo_ | Zipper_32: none whatsoever. |
| 23:25.37 | Corydon-w | ucfMethod: someone must have written that as a third party addon... |
| 23:25.45 | ucfMethod | hmm |
| 23:26.05 | unfo_ | I haven't even bought an FXO/FXS card. I do have a builtin internal modem in my Dell laptop though. |
| 23:26.05 | Corydon-w | ucfMethod: there is, however, func_odbc, which will be standard in 1.4 |
| 23:26.09 | Zipper_32 | unfo_: How much linux experience? |
| 23:26.24 | *** join/#asterisk SGM (n=stoyan@home.marinov.us) |
| 23:26.28 | unfo_ | Zipper_32: A fair bit. I am learning Debian packaging currently. |
| 23:26.47 | Zipper_32 | unfo_: Have you got a running copy of *? |
| 23:27.00 | ucfMethod | i need it for our dialing plan, where someone dials an extension, and it auto calls up to 4 numbers based how the employee sets it up |
| 23:27.00 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:27.00 | SGM | hi guys |
| 23:27.09 | SGM | i have a question about channel variables |
| 23:27.15 | unfo_ | I first started with Caldera maybe 5 years ago. I have used Linux all day most days for a year or two now. |
| 23:27.17 | ucfMethod | Corydon-w: thanks ill look into that |
| 23:27.41 | unfo_ | Zipper_32: no, will apt-get it now. |
| 23:28.00 | Corydon-w | http://asterisk.org/func_odbc |
| 23:28.59 | EmleyMoor | Is there an Asterisk app that will read back pulse-dialed digits? |
| 23:29.20 | Corydon-w | Pulse digits work identically to DTMF digits in Asterisk |
| 23:29.47 | EmleyMoor | Corydon-w: Sure? My IVR and the voicemail don't accept them |
| 23:30.03 | Corydon-w | EmleyMoor: yes, I'm sure |
| 23:30.13 | Corydon-w | EmleyMoor: are you using Digium hardware? |
| 23:30.19 | EmleyMoor | Asterisk itself does, I'll grant you that |
| 23:30.20 | EmleyMoor | Yes |
| 23:31.04 | Corydon-w | Then it should work just fine |
| 23:31.14 | sevard | Don't you have to enable pluse dialing in zaptel? |
| 23:31.30 | Corydon-w | The detection for pulse dialled digits is builtin to the DTMF detection routines |
| 23:31.37 | unfo_ | Zipper_32: success. |
| 23:31.55 | Corydon-w | sevard: that's for outbound only |
| 23:32.14 | Corydon-w | sevard: and no, you just need the digit p to switch to pulse dialling |
| 23:32.31 | hoobastooba | i am still having trouble making an iax call from one asterisk server to another. |
| 23:32.31 | sevard | http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing |
| 23:32.32 | Corydon-w | i.e. Dial(Zap/g1/p2345678) |
| 23:32.40 | sevard | Ah, I see. |
| 23:32.48 | Zipper_32 | unfo_: Alright, what I suggest you do is work a bit with a standard dialplan (extensions.conf) |
| 23:32.55 | sevard | I'm not insane enough to find a phone that still does puse dialing by default |
| 23:32.57 | hoobastooba | Nov 21 16:26:38 NOTICE[3714]: chan_iax2.c:6882 socket_read: Rejected connect attempt from 10.0.1.180, who was trying to reach '8018285230@Progrexion_O' |
| 23:33.14 | Zipper_32 | unfo_: If you can get a feel for how to navigate calls, then you're part way to getting your job done. |
| 23:33.39 | sevard | Getting your Asterisk-Legs is all part of the problem. |
| 23:33.41 | ucfMethod | holy cow, func_odbc is exactly what i needed |
| 23:33.47 | Corydon-w | sevard: you can switch back to dtmf with digit 'd' |
| 23:33.48 | ucfMethod | perfecto |
| 23:33.54 | hoobastooba | here is the iax.conf entry on the server i am tyring to dial to: http://pastebin.ca/254531 |
| 23:33.57 | Zipper_32 | unfo_: After that, it's as easy as following the voicemail.conf, then manipulating the .gsm file that is saved when somebody leaves you a message. |
| 23:34.02 | Corydon-w | sevard: so you can actually dial with a combination of both dtmf and pulse |
| 23:34.07 | ucfMethod | Corydon-w: you rule |
| 23:34.28 | sevard | e/xec date |
| 23:34.32 | Corydon-w | ucfMethod: yep, pretty cool stuff |
| 23:34.33 | sevard | bah |
| 23:34.50 | hoobastooba | here is the line I am using for dialing: http://pastebin.ca/254533 |
| 23:35.41 | unfo_ | Zipper_32: but wouldn't it be cool if what I suggested (voicemail autopickup after 7 rings on all lines) were the default? |
| 23:36.03 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool1-11.mtco.com) |
| 23:36.22 | Corydon-w | unfo_: there is no default, because Asterisk is a blank slate... on which you can do pretty much anything |
| 23:36.45 | Corydon-w | Asterisk is not a key system |
| 23:37.12 | hoobastooba | there must be something dumb that I am missing.... but i for the life of me cannot find it |
| 23:37.16 | unfo_ | Corydon-w: therefore it should be harder to set up than a KTS? :) |
| 23:37.30 | hoobastooba | i am doing this on another server... but this one just will not work for me/ |
| 23:37.36 | sevard | Corydon-w: It would be neet to provide key-like system functionality within Asterisk. |
| 23:37.42 | Corydon-w | It's not harder to set up... The process is just different |
| 23:37.42 | unfo_ | IMO, intelligent defaults are always nice. |
| 23:37.49 | SGM | so, as i was saying im originating a call using the ami originate action and i set a few variables. when the caller hangs up the phone it goes to the h priority as expected, but the variables are gone :/ |
| 23:38.02 | fatpelt | hmm. seems like an rxgain of 30 is way too high, but i can hardly hear a thing off incoming calls without it |
| 23:38.39 | sevard | Corydon-w: most key systems have line presence, eg just by looking at your phone you can tell "line 1,3,4,5, and 6 are active, i choose line 2" |
| 23:38.58 | Corydon-w | sevard: Asterisk does that automatically for you |
| 23:39.03 | sevard | Exactly |
| 23:39.09 | sevard | that's what a PBX does |
| 23:39.17 | Zipper_32 | unfo_: It's just as easy to configure the number 7 as the numbe of desired rings. Why make it default? Everybody has their own opinion of what should be default. It's best to leave that commented out. |
| 23:39.22 | sevard | I said: It would be neay to provide key-like system functionality within Asterisk. |
| 23:39.57 | *** join/#asterisk grexk (n=destroye@124.107.72.45) |
| 23:40.50 | sevard | Zipper_32 and unfo_: it's best to avoid saying "rings" when you want to set desired length. how many different ring tones have you heard in the past 10 days? "seconds" is more appropriate. |
| 23:41.00 | Corydon-w | sevard: yeah, all you'd have to do is restrict Asterisk's features down to a small set of functionality, and you'd have a key system |
| 23:41.19 | Zipper_32 | sevard: I'm just trying to explain it to him in his words. But yes, I agree. |
| 23:41.23 | sevard | Corydon-w: you woudln't have lin presence |
| 23:41.33 | sevard | you still wouldn't have line presence* |
| 23:41.42 | Corydon-w | sevard: it can be done with a Polycom phone |
| 23:41.56 | sevard | Corydon-w: Neat, I've never had the pleasure of playing with a polycom |
| 23:42.21 | Corydon-w | sevard: you'd just subscribe to watch those associated channels |
| 23:42.23 | bluregard | I just got my IP 501, it kicks ass |
| 23:42.56 | unfo_ | sevard: I'm just a home user, so only four kinds of ring: Bell Canada and three cellphones. :) |
| 23:43.27 | sevard | Corydon-w: how many lines can you watch? or wait, with those polycoms you can get the sidecars, right |
| 23:44.25 | hoobastooba | no one? :'( there must be someone who knows what I messed up. |
| 23:44.25 | Corydon-w | You can even watch lines from the 501, if you sacrifice a line key |
| 23:44.32 | bluregard | sevard:only the 60x can use the addon modules |
| 23:45.06 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:46.17 | hoobastooba | have i burned everyone out with my stupid issues? is that it :-[ |
| 23:46.17 | sevard | hoobastooba: did you pastebin your configs? |
| 23:46.23 | hoobastooba | sevard: |
| 23:46.26 | hoobastooba | i did.. hang on |
| 23:46.38 | hoobastooba | here is the line I am using for dialing: http://pastebin.ca/254533 |
| 23:46.58 | hoobastooba | here is the iax.conf entry on the server i am tyring to dial to: http://pastebin.ca/254531 |
| 23:47.45 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:48.00 | unfo_ | theoretical question: as soon as asterisk is running on a PC, does it automatically listen for voicemails on all line cards? |
| 23:48.13 | unfo_ | (e.g. autopickup after some default number of rings) |
| 23:48.28 | Corydon-w | listen for, what? |
| 23:48.28 | monsted | unfo_: no |
| 23:48.46 | hoobastooba | unfo_: the cards have nothing to do with voicemail |
| 23:49.15 | Corydon-w | The cards aren't like Dialogic cards... they are dumb channels... |
| 23:49.31 | monsted | fortunately, asterisk doesn't do anything you haven't asked it to do :) |
| 23:49.38 | unfo_ | monsted: fortunately? |
| 23:49.42 | monsted | yes |
| 23:49.50 | unfo_ | why? |
| 23:50.07 | Corydon-w | in other words, Asterisk does exactly what you tell it to do, no more, no less. |
| 23:50.14 | fatpelt | does anyone find they have to play with txgain and rxgain ? |
| 23:50.15 | sevard | hoobastooba: are you registered via IAX to the other server? |
| 23:50.21 | monsted | because it'd drive people nuts if it did lots of weird stuff it wasn't configured to do |
| 23:50.34 | hoobastooba | sevard: must not be... where is that at? |
| 23:50.38 | sevard | hoobastooba: iax2 show peers |
| 23:50.50 | *** join/#asterisk awannabe (n=brad@mail.azprm.com) |
| 23:50.55 | sevard | you must be registered to the server you want to push calls to |
| 23:51.00 | awannabe | hey guys, does the 601 side module have to be configured via XML? |
| 23:51.10 | monsted | unfo_: if you want crap that tries to think for you (and usually fails), install a microsoft product |
| 23:51.15 | hoobastooba | sevard: apparently not... only the demo one. |
| 23:51.21 | sevard | you'll want to do that |
| 23:51.40 | sevard | and I don't know what you mean by 'demo one' |
| 23:51.50 | sevard | *earth shakes* |
| 23:52.11 | hoobastooba | demo/asterisk 216.207.245.47 (S) 255.255.255.255 4569 Unmonitored |
| 23:52.26 | hoobastooba | besides the host= line in iax.conf, where do i make it register? |
| 23:52.50 | sevard | in the client's iax.conf add a register => line |
| 23:52.59 | SGM | hohum: register=>user:pass@host |
| 23:53.02 | SGM | oops |
| 23:53.09 | SGM | that was 4 hoobastooba |
| 23:53.12 | sevard | register =>, oh sgm already has it |
| 23:53.19 | hoobastooba | cool... will do it. |
| 23:53.20 | hoobastooba | thanks |
| 23:53.28 | Corydon-w | hoobastooba: why are you using trunking? |
| 23:53.38 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-114-42-210.satx.res.rr.com) |
| 23:53.39 | SGM | any toughts on my question? |
| 23:53.39 | sevard | that's why it's rejecting your calls, you aren't authorized to make calls on that server. |
| 23:53.42 | hoobastooba | playing with vicidial and not getting all the way through it |
| 23:53.50 | hoobastooba | thus the trunking. |
| 23:53.58 | sevard | SGM: I missed your question |
| 23:54.01 | hoobastooba | you guys have experience there? |
| 23:54.02 | unfo_ | monsted: I insist that intelligent defaults are an advantage, not a disadvantage. If someone does not want voicemail, they can disable it, but it is a good default for home users. |
| 23:54.06 | SGM | about channel variables |
| 23:54.12 | SGM | im originating a channel using the ami |
| 23:54.13 | sevard | what about them |
| 23:54.14 | Corydon-w | hoobastooba: btw, if both hosts are configured statically, there's no need to use authentication |
| 23:54.15 | SGM | and then |
| 23:54.28 | SGM | im dialing some number |
| 23:54.33 | hoobastooba | Corydon-w: new concept for me |
| 23:54.58 | SGM | if the callee (originated with the ami) picks up the phone |
| 23:55.00 | SGM | and then hangs up |
| 23:55.07 | Corydon-w | Authentication only really matters when one of the hosts is dynamic |
| 23:55.09 | awannabe | does anyone have the polycom XML files?? Im in a total crunch and do anything for them please! |
| 23:55.10 | SGM | it jums to the h exten |
| 23:55.19 | sevard | the h exten = hangup |
| 23:55.23 | SGM | but the vars, set with the originate action r missing |
| 23:55.27 | Corydon-w | sevard: and he can't register if the destination doesn't have host=dynamic |
| 23:55.31 | sevard | and what the crap is ami |
| 23:55.42 | sevard | Corydon-w: missed that, need food in stomach. |
| 23:55.42 | SGM | asterisk manager interface |
| 23:55.45 | Corydon-w | ami=asterisk manager interface |
| 23:55.47 | sevard | oh |
| 23:55.49 | SGM | :) |
| 23:55.56 | sevard | i know nothing of the manager interface, sorry |
| 23:56.03 | unfo_ | Can ami or some other config tool automatically set up a simple voicemail setup for home users with just a few clicks? |
| 23:56.22 | Corydon-w | unfo_: AMI is not a config tool |
| 23:56.33 | unfo_ | i stand corrected |
| 23:56.33 | SGM | unfo_: u can do it with a few lines |
| 23:56.42 | sevard | unfo_: perhaps you should start off with trixbox, or whatever it's called these days |
| 23:56.44 | SGM | how much lines do u have to type 2 make this "tool" work |
| 23:56.45 | SGM | ;) |
| 23:56.58 | Corydon-w | unfo_: it is a management interface... |
| 23:57.13 | unfo_ | sevard: i wish apt-get/debconf in debian/ubuntu could do it for you. |
| 23:57.30 | sevard | unfo_: maybe one of these days you'll understand that in linux, not everything comes prepackaged. |
| 23:57.53 | unfo_ | sevard: I know that not everything comes prepackaged, and would like things to change. :) |
| 23:58.15 | hoobastooba | Corydon-w: how do i configure these statically? that must be how I have the other one, I just dont know what I have done to do it. my other one does not have any register=> entries on either server and it works. |
| 23:58.19 | sevard | unfo_: that kind of attitude will get you _nowhere_ |
| 23:58.26 | Corydon-w | unfo_: things won't change like that. Prepackaging would mean we'd lose all the power |
| 23:58.43 | hoobastooba | if you want prepackaged... go to windows :-D |
| 23:58.47 | Corydon-w | hoobastooba: you need a type=user on the local machine |
| 23:58.50 | sevard | unfo_: if you want that sort of solution, I know Microsoft is developing this great product for idiots, they call it Windows. |
| 23:58.59 | unfo_ | sevard: is it bad that debian offers you a variety of premade config options for exim4 as soon as you install it? |
| 23:59.02 | hoobastooba | ok, that i have on the one that is not working. |
| 23:59.09 | hoobastooba | as well as the one I am trying to connect to. |
| 23:59.30 | Corydon-w | hoobastooba: so, type=user, host=<ip address>, right? |
| 23:59.34 | sevard | unfo_: From my standpoint, yes.. but I have a different philosophical standpoint on things. that's all opinion. |
| 23:59.39 | SGM | unfo_: some guy had written a nice-looking windows tool to configure the * for you |
| 23:59.46 | SGM | i guess its a click-install |
| 23:59.46 | Corydon-w | hoobastooba: username=<remote config name> |
| 23:59.53 | hoobastooba | http://pastebin.ca/254531 |
| 23:59.54 | SGM | and it produces some configs |
