| 00:01.33 | *** join/#asterisk JoseBravo-Home (n=jbravo@190.159.19.180) |
| 00:02.14 | JoseBravo-Home | I need to identify this tone, http://www.colombiahosting.com.co/busy.wav any one know it? |
| 00:02.27 | *** join/#asterisk CoffeeKid (n=kirk@dsl093-224-026.slc1.dsl.speakeasy.net) |
| 00:03.06 | JoseBravo-Home | Its a common busy tone? |
| 00:03.45 | CoffeeKid | Hi everyone... I'm trying to get a ${DIALSTATUS} variable back with call files.. is this possible? it keeps coming back blank. |
| 00:10.36 | znoG_ | anyone happen to know why Asterisk is not transcoding between alaw and ulaw? |
| 00:10.48 | znoG_ | the call works 100% fine when the 2 codecs are the same, but not when transcoding needs to happen |
| 00:11.16 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:11.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:13.04 | *** join/#asterisk xenonex (n=xenonex@88.204.243.227) |
| 00:13.05 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 00:13.51 | SwK | hmmmmm |
| 00:16.39 | disposable | is there a way to give * a file of commands i need executed in * console? i basically need to reboot 150 phones with 'sip notify polycom-check-cfg phonename' and it will have to be done regularly. |
| 00:20.30 | CoffeeKid | disposable: i think your best bet would be to make a script to execute the notify on each polycom config. |
| 00:20.54 | CoffeeKid | loop through each phone then do a notifiy on each one individualy in the script. |
| 00:26.22 | fujin | disposable: cronjob, perl, done |
| 00:26.52 | billytwowilly | hey anyone know any asterisk consultants in Calgary? |
| 00:26.55 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 00:26.58 | fujin | What's calgary? |
| 00:27.24 | billytwowilly | A city in Alberta, Canada |
| 00:31.40 | *** join/#asterisk [gquit]bombadil (n=dana@70.94.44.157) |
| 00:32.06 | coppice | we all know canada is just a frozen wasteland, populated by mouse. Calgary was in the Bible, and the Bible wasn't set in the ice and snow |
| 00:33.18 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 00:33.27 | billytwowilly | Nice. |
| 00:33.34 | billytwowilly | It's also moose dummy. |
| 00:33.58 | coppice | you don't have any desserts up there? |
| 00:34.13 | billytwowilly | we don't eat dessert. It's bad for you. |
| 00:39.56 | *** part/#asterisk _trine (n=trine@38.100.90.3) |
| 00:40.51 | *** join/#asterisk Katty (n=Angela@adsl-75-59-138-34.dsl.stlsmo.sbcglobal.net) |
| 00:42.02 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 00:42.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:43.56 | *** part/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 00:48.41 | Katty | herro. |
| 00:49.05 | Katty | how do i run asterisk plzzzz????!?! |
| 00:49.23 | Qwell | ~nowwhat |
| 00:49.24 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 00:49.33 | Katty | oh boy! it's Qwell! |
| 00:49.37 | outtolunc | you puts it in one of those little round wheel things |
| 00:49.54 | Qwell | ~cry |
| 00:49.55 | jbot | ACTION cries and sobs until she nearly drowns in her own tears. |
| 00:50.07 | Qwell | look what you've done |
| 00:50.13 | Katty | Qwell: hug Qwell |
| 00:50.16 | Katty | oh. |
| 00:50.19 | Katty | jbot: hug Qwell |
| 00:50.20 | jbot | ACTION hugs Qwell |
| 00:50.42 | Katty | Qwell: aww. you two made up. how sweet. |
| 00:54.26 | Katty | file: ping? |
| 01:04.19 | drmessano | Hmmm |
| 01:05.08 | *** join/#asterisk Vco (i=Vco@S0106001b38367a26.cg.shawcable.net) |
| 01:06.59 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 01:07.50 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 01:08.07 | *** join/#asterisk nvrpunk (n=root@80.78.22.88) |
| 01:09.30 | adeel | in the beginning of inbound calls, there's a period of 2-3 seconds where no audio is being passed, anyway to remove that delay? |
| 01:13.32 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:14.12 | *** join/#asterisk BeeBuu (n=beebuu@219.135.42.79) |
| 01:15.07 | _ShrikE | adeel: you using echotraining? |
| 01:15.15 | Katty | drmessano: allo! |
| 01:15.25 | Katty | hey _ShrikE (= |
| 01:15.48 | _ShrikE | Katty: howdy |
| 01:16.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:18.29 | drmessano | Hi Katty |
| 01:18.37 | drmessano | Know anything about BIND? lol |
| 01:19.06 | Katty | i don't think so |
| 01:19.08 | Katty | sorry ;) |
| 01:19.16 | Katty | drmessano: what's it do? |
| 01:19.20 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
| 01:19.38 | ZPertee | what is the minimum for .call file? |
| 01:20.04 | Katty | mine's pretty small. |
| 01:20.10 | TJNII | drmessano: whatcha' workin' on? |
| 01:20.17 | outtolunc | channel + context/extension/priority OR app/data |
| 01:20.40 | Katty | ZPertee: i can pastebin mine if you need a working example. |
| 01:20.50 | draygon | !seen cmantito |
| 01:25.34 | Katty | i...think i might have figured out that call barge thing. |
| 01:25.36 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:26.05 | Katty | would zap channelredirect to meetme(whatever) and create a second call calling the original user into the conference...and then calling a 3rd party doing the barge............ actually work? |
| 01:26.14 | *** part/#asterisk stegbth (n=stegbth@p4FD0B6A0.dip0.t-ipconnect.de) |
| 01:28.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 01:29.01 | CCFL_Man2 | is old 2 pair 1970s Ma'Bell wire fine for adsl? |
| 01:29.15 | drmessano | lol |
| 01:29.21 | drmessano | It should be |
| 01:31.31 | Katty | hrm |
| 01:31.58 | Katty | i don't know how to to read(ext) and then tell the server to tell me what ZAP or SIP or whatever is connected to SIP/${EXTEN} |
| 01:32.13 | Katty | sip/'ext' i mean |
| 01:33.52 | Katty | i need a sort of... BRIDGEPEER variable |
| 01:34.37 | Katty | so it does channelredirect bridgedpeer to conferece, then me, then add sipheader auto answer to whatever the original SIP person was |
| 01:34.41 | Katty | via a .call file |
| 01:35.11 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 01:37.04 | ZPertee | Katty: sorry r u still on? i wasn't paying attention. |
| 01:38.06 | Katty | ZPertee: yes, i am still here. |
| 01:38.53 | ZPertee | Katty: ok. I want to go over zap/7 but I need to flash the channel first. should I still include the channel in the .call |
| 01:39.17 | Katty | flash the channel? |
| 01:39.50 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 01:39.57 | ZPertee | Katty: yeah Flash(). I am going over my loudspeaker to page someone |
| 01:40.08 | Katty | no idea. |
| 01:40.12 | Katty | what i did... |
| 01:40.17 | Katty | because we use it for intercom too |
| 01:40.28 | Katty | Channel: Local/1@intercomincomingcall |
| 01:41.00 | Katty | context: intercomincomingcall ... extension: s ... priority: 1 |
| 01:41.26 | ZPertee | Katty: ok what do you mean by Local/1? sorry relative newb |
| 01:41.59 | ZPertee | Katty connect to Fxo or fxs card or what? |
| 01:42.37 | Katty | well we were using 3 cards to handle analog lines |
| 01:42.40 | Katty | but we're using a pri now |
| 01:43.04 | Katty | ZPertee: the .call file just says NEW CALL GO HERE PLS |
| 01:43.18 | Katty | ZPertee: specifically, context, s, and priority |
| 01:43.32 | Katty | ZPertee: and that context defines what the call does. |
| 01:44.15 | Katty | ZPertee: http://pastebin.ca/939127 <- the .call file is told to go to intercomincomingcall,s,1 |
| 01:44.25 | CCFL_Man2 | http://cgi.ebay.com/WESTERN-ELECTRIC-CANDLESTICK-TELEPHONE-phone-and-SUBSET_W0QQitemZ280207774067QQihZ018QQcategoryZ38037QQssPageNameZWDVWQQrdZ1QQcmdZViewItem <--what i want |
| 01:44.46 | ZPertee | Katty: ok makes sense and thanks for your help. I'll test it out tomorrow. should be relatively simple to tweak until I get it to work. |
| 01:45.07 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 01:45.54 | Katty | ZPertee: it's really not too complicated at all (= |
| 01:46.02 | Katty | ZPertee: but it was real confusing the first time around too. |
| 01:46.22 | *** join/#asterisk PepOSX (n=angeldav@190.72.146.124) |
| 01:47.01 | ZPertee | Katty: yeah. I was looking for a good excuse to get it .call files figured out anyhow so it looks like I get me chance now :) |
| 01:49.09 | Katty | ZPertee: sec, i'll pastebin the whole lot |
| 01:49.20 | ZPertee | Katty: sweet |
| 01:50.41 | outtolunc | now that was funny |
| 01:50.50 | *** join/#asterisk Katty (n=Angela@adsl-75-59-138-34.dsl.stlsmo.sbcglobal.net) |
| 01:51.17 | Katty | ZPertee: http://pastebin.ca/939133 |
| 01:51.52 | Katty | ZPertee: so in that silly senario, the caller rings an operator. if the operator doesn't pick up, the call is answered and told to hold, then it dials a whole bunch of people and ends up in an auto attendant if no one answers the secondtime |
| 01:52.19 | Katty | ZPertee: MEANWHILE, the system copies the call file into /var/spool/asterisk which initiates an internal call on auto answer, telling the friggin people to answer their damn phones :P |
| 01:53.26 | Katty | ZPertee: the channel is what it's playing, and the rest is... how to get the channel open, as it were. |
| 01:53.41 | znoG_ | anyone happen to know why Asterisk is not transcoding between alaw and ulaw? |
| 01:53.43 | znoG_ | the call works 100% fine when the 2 codecs are the same, but not when transcoding needs to happen |
| 01:53.52 | droops | katty, you should move (mv) not copy (cp) |
| 01:54.09 | Katty | droops: i mv from /etc/asterisk to /var/spool (= |
| 01:54.15 | ZPertee | Katty: You're awesome! It's people like you that keeps Linux and open source going! |
| 01:55.14 | Katty | droops: REF lines 5&6, particularly 6 |
| 01:55.45 | Katty | ZPertee: you can do a bunch of variations. i'm not sure what Flash() does or how it works... |
| 01:55.53 | droops | sorry, i didnt see the pastbin |
| 01:55.55 | Katty | ZPertee: i've been thinking about using it for 911 |
| 01:56.20 | Katty | ZPertee: 911 automatically put a person into meetme, and transfer the 911 in there too |
| 01:56.45 | Katty | ZPertee: then a call would come from /var/spool/asterisk to auto answer the rest of the building, or at least supervisors, into that 911 call too |
| 01:56.50 | *** join/#asterisk saftsack (n=saftsack@p4FC76564.dip.t-dialin.net) |
| 01:57.05 | ZPertee | Katty: neat idea. might have to do that sometime. this http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash is what flash does |
| 01:57.57 | Katty | ah... |
| 01:58.11 | Katty | handy. |
| 01:58.18 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 01:58.24 | Katty | we just use some findme follow me stuff here |
| 01:59.26 | ZPertee | Katty: yeah this is the first time that I had to use it. I am trying to integrate asterisk with Avaya ACS pbx and with it you can't do anything without using flash |
| 01:59.40 | Katty | fun times. |
| 01:59.50 | Katty | may the force be with you (= |
| 02:00.07 | ZPertee | oh yeah. now tht I figured it out and the constant headache is starting to go away |
| 02:00.58 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 02:01.07 | Katty | ZPertee: do i ever know that feeling. |
| 02:01.16 | Katty | ZPertee: i've been fighting a sangoma card talking to a pri card all day |
| 02:01.38 | CoffeeKid | Hey everyone, I'm looking for a way to get ${DIALSTATUS} (or some similar status variable) from a channel started with call files.. can anyone help me? |
| 02:01.40 | Katty | ZPertee: despite it sounding easy, it's driving me nuts :< |
| 02:02.12 | ZPertee | yeah but the high that I get from finally figuring out a problem is what keeps me going! |
| 02:02.41 | *** part/#asterisk pcrack (n=pcrack@202.164.176.106) |
| 02:03.08 | Katty | CoffeeKid: where do you want the dialstatus dumped to? |
| 02:03.57 | *** part/#asterisk Merlin (i=merlin@hick.org) |
| 02:04.36 | CoffeeKid | Katty: i'm trying to just do a NoOp(${DIALSTATUS}).. but with call files its always blank |
| 02:04.46 | obnauticus | how do i have an IVR menu where it waits for the extension while it playsback a file |
| 02:04.50 | obnauticus | like say you are listening to a menu |
| 02:04.53 | obnauticus | and the user presses 2 |
| 02:05.21 | Katty | CoffeeKid: what do you actually want the dialstatus for tho |
| 02:05.25 | Katty | CoffeeKid: go to a particular context? |
| 02:05.45 | CoffeeKid | Katty: right, i need to execute different commands in the dialplan based on what the status was |
| 02:05.59 | Katty | oh, okay. |
| 02:06.01 | CoffeeKid | Katty: if it were busy, i need to notify my database with an AGI that the number was busy, etc.. |
| 02:06.28 | CoffeeKid | Katty: if its a disconnect, tell the database that the number isn't valid.. etc |
| 02:06.35 | Katty | so, exten => s,1,Dial(whoever) then s,2,Goto(s-${Dialstatus},1) |
| 02:07.06 | Katty | exten => s-NOANSWER,1,Goto(somecontextwhichcreatesnoanswer.call) |
| 02:07.21 | Katty | exten => s-chanunavail,1,Voicemailmain |
| 02:07.30 | CoffeeKid | Katty: that won't work because the call is already on the line when it reaches the context and priority |
| 02:07.34 | Katty | exten +> s-BUSY,1,do some toher stuff |
| 02:08.00 | Katty | CoffeeKid: pastebin? |
| 02:08.10 | Katty | obnauticus: WaitExten(seconds) |
| 02:08.33 | CoffeeKid | Katty: one moment, let me setup a pastbin file to show you what i'm trying to do |
| 02:08.40 | Katty | obnauticus: so you'd have s,1,Answer s,2,Background(audiofile) s,3,WaitExten(10) s,4,Dosomethingelse |
| 02:09.02 | Katty | obnauticus: then lower down, you'd have exten => 1,1,dosomething |
| 02:09.19 | Katty | obnauticus: exten => 2,1,Dial(someone) |
| 02:09.33 | Katty | obnauticus: the choices that link what they dialed to what it does |
| 02:09.51 | obnauticus | wgat di yiy neab |
| 02:10.03 | obnauticus | oop |
| 02:10.03 | obnauticus | s |
| 02:10.03 | obnauticus | lol |
| 02:10.06 | obnauticus | i will pastebin what i have |
| 02:10.09 | Katty | k |
| 02:11.03 | obnauticus | Line 7's just festival so ignore the cutoff |
| 02:11.15 | obnauticus | http://rafb.net/p/HgoWM495.html |
| 02:12.05 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:12.05 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:12.32 | LeddyHM | can the timeout on qualify=yes be modified? |
| 02:12.39 | Katty | obnauticus: move line 9 u to 7 |
| 02:13.01 | Katty | i would recommend s,7,goto(festivalmenu,s,1) |
| 02:13.34 | obnauticus | i have another one |
| 02:13.36 | obnauticus | to goto a hangup menu |
| 02:13.43 | obnauticus | that's just the relevant piece... |
| 02:13.46 | Katty | k |
| 02:13.54 | Katty | i've always kept s,whatever together. |
| 02:15.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 02:16.38 | obnauticus | Katty how would i err |
| 02:16.49 | obnauticus | make it so that it waits for the extension while executing festival? |
| 02:17.11 | Katty | dunno. |
| 02:17.14 | Katty | never worked with festival before. |
| 02:18.08 | Katty | obnauticus: pretend s is a real number. |
| 02:18.16 | obnauticus | k |
| 02:18.23 | Katty | obnauticus: don't break it apart... |
| 02:18.32 | obnauticus | i don't know how i did that |
| 02:18.35 | obnauticus | it just ... ended up that way |
| 02:18.35 | obnauticus | lol |
| 02:19.02 | russellb | WHAT |
| 02:19.32 | russellb | <3 |
| 02:19.37 | russellb | yay |
| 02:19.40 | obnauticus | 3-way |
| 02:19.48 | draygon | <- man |
| 02:19.49 | obnauticus | NO! |
| 02:20.15 | Katty | russellb: i've heard you've been collaborating shamelessly with file |
| 02:20.35 | russellb | oh really? |
| 02:20.38 | russellb | on what? |
| 02:20.46 | russellb | we collaborate all the time |
| 02:20.48 | draygon | russellb, did you get my hug........ |
| 02:20.50 | russellb | we're just like a mini-mall |
| 02:20.56 | russellb | draygon: sure did! |
| 02:21.06 | draygon | ARE YOU RICK ROLLING? |
| 02:21.12 | russellb | ~nowwhat |
| 02:21.13 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 02:21.17 | Katty | russellb: exactly. |
| 02:21.37 | obnauticus | ~thenwhat |
| 02:21.41 | obnauticus | :( |
| 02:22.31 | *** part/#asterisk Jocko (i=daemon@75.146.55.225) |
| 02:23.35 | obnauticus | err |
| 02:23.38 | obnauticus | ~asterisk18 |
| 02:23.39 | jbot | Some people are never happy |
| 02:23.40 | obnauticus | ~asterisk16 |
| 02:23.41 | jbot | new features in Asterisk 1.6 are listed at http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup |
| 02:23.44 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 02:24.55 | andrewy | from what I've read sipura devices connect with asterisk using sip. can a sipura spa-3000 (the model with a fxo) still connect to a pots? |
| 02:25.25 | riddlebox | andrewy, if thats what it was built to do, then yes |
| 02:25.27 | obnauticus | through asterisk yes. |
| 02:25.36 | obnauticus | and what riddlebox said |
| 02:25.36 | obnauticus | lol |
| 02:25.57 | russellb | ~asterisk10 |
| 02:25.59 | riddlebox | ;) |
| 02:26.04 | russellb | sad. |
| 02:26.15 | obnauticus | !asterisk69 |
| 02:26.18 | obnauticus | ~asterisk69 |
| 02:26.24 | obnauticus | someoen needs to add an entry for that. |
| 02:27.28 | russellb | jbot: asterisk10 was the first maintained Asterisk release branch. It has been completely unmaintained since the Fall of 2006. R.I.P. Asterisk 1.0 |
| 02:27.56 | russellb | jbot: ANSWER ME |
| 02:28.13 | obnauticus | what about Asterisk 69 |
| 02:28.15 | obnauticus | c'mon |
| 02:28.25 | [TK]D-Fender | andrewy, Yes, the SPA-3000 (and SPA-3102) have 1 FXS port, and 1 FXO port. Both operate fully independantly |
| 02:28.32 | file | russellb: you are just like a mini-mall |
| 02:28.58 | russellb | file: no, YOU ARE |
| 02:29.00 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6a07eae0fc0770be) |
| 02:29.09 | russellb | file: but seriously, i think asterisk is _just_ like a mini-mall |
| 02:29.11 | andrewy | [TK]D-Fender: thanks, just making sure before I buy anything |
| 02:29.27 | russellb | riddlebox: he bites my head off daily |
| 02:29.48 | riddlebox | russellb, want to start a club |
| 02:29.51 | russellb | sure! |
| 02:29.53 | file | russellb: are you talkin' 'bout flea market? |
| 02:29.55 | russellb | [TK]D-Fender: j/k <3 |
| 02:30.00 | russellb | file: yea! |
| 02:30.10 | russellb | we talkin' 'bout flea market ... |
| 02:30.27 | russellb | we got living rooms ... bedrooms ... dinettes ... |
| 02:30.39 | russellb | omg, make it stop |
| 02:32.02 | Katty | file: HI |
| 02:32.04 | Katty | [TK]D-Fender: hai! |
| 02:33.53 | file | ohai |
| 02:34.01 | [TK]D-Fender | Katty, Mew. |
| 02:38.42 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 02:44.20 | *** join/#asterisk putnopvut (n=putnopvu@76.29.245.9) |
| 02:44.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 02:46.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 02:51.26 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:52.39 | Katty | [TK]D-Fender: mew! |
| 02:52.50 | [TK]D-Fender | russellb, I've got a newfound appreciation of Rick Astley.... an inverse proportion to Mongomery. |
| 02:53.17 | russellb | heh |
| 02:54.42 | Katty | [TK]D-Fender: i like authenticate. |
| 02:54.52 | tengulre | anybody here from china? |
| 02:54.53 | Katty | [TK]D-Fender: it gives me warm fuzzy feelings. |
| 02:55.04 | Katty | tengulre: my pants are. |
| 02:55.34 | Qwell | Katty: your pants are what? |
| 02:55.44 | Katty | Qwell: from china |
| 02:55.48 | Qwell | I see |
| 02:55.58 | Katty | Qwell: really. |
| 02:56.03 | Katty | Qwell: i have my doubts. |
| 02:56.20 | Qwell | I don't buy it |
| 02:56.35 | Katty | foiled again! |
| 02:57.29 | Katty | foiled like a bloody pair of binomials!!! |
| 02:57.44 | [TK]D-Fender | Katty, you forgot to factor those in! |
| 02:57.46 | [TK]D-Fender | ;) |
| 02:57.49 | [TK]D-Fender | </pun> |
| 02:57.51 | Katty | [TK]D-Fender: i'll factor you in a minute |
| 02:58.14 | [TK]D-Fender | Katty, sounds kinky ;) |
| 02:58.53 | Katty | [TK]D-Fender: sorry, i'm strictly monomial. |
| 03:00.30 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:00.30 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:01.12 | Katty | [TK]D-Fender: go lookup terry fator on youtube |
| 03:02.10 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 03:02.11 | Katty | jbot: terry fator |
| 03:02.21 | Katty | jbot: you make me sad. |
| 03:03.00 | [TK]D-Fender | Katty, ummm.... ok,fine,sure.... |
| 03:04.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b4587094b835dd75) |
| 03:05.05 | Katty | [TK]D-Fender: it makes me happy. |
| 03:07.22 | Katty | [TK]D-Fender: maybe i just have an odd definition of 'talent' |
| 03:07.37 | Qwell | ~define talent |
| 03:07.37 | jbot | Use ~dict for definitions. |
| 03:07.41 | Qwell | ~dict talent |
| 03:07.55 | Qwell | recursive entry. fun |
| 03:08.03 | [TK]D-Fender | Katty, mine involves trampolines, and swings in the bedroom ;) |
| 03:08.10 | Katty | ... |
| 03:08.26 | russellb | WHAT |
| 03:09.06 | Corydon76-dig | russellb: Why, are those your talents, too? :-P |
| 03:09.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:09.13 | Katty | </cheesy> |
| 03:09.31 | Corydon76-dig | If your banana is cheesy, there's something wrong |
| 03:09.36 | Katty | ... |
| 03:09.38 | [TK]D-Fender | Victory is mine! |
| 03:09.42 | Katty | <bkw> NEXT!!!! |
| 03:10.10 | russellb | O.o |
| 03:10.16 | russellb | no ... my talents are ... um ... |
| 03:10.26 | Katty | golf cart driving. |
| 03:10.43 | russellb | yes. |
| 03:10.50 | russellb | i actually did that for a job for a few years in high school ... sort of |
| 03:10.54 | russellb | worked at a driving range |
| 03:11.10 | Katty | russellb: i know everything about you. |
| 03:11.16 | Katty | *1 |
| 03:11.22 | Katty | 1 - above statements are not true. |
| 03:11.23 | russellb | lies |
| 03:11.24 | Corydon76-dig | Stalker!!! |
| 03:11.31 | russellb | where was i born?! |
| 03:11.49 | Corydon76-dig | You were born on Planet Earth |
| 03:12.07 | Katty | russellb: alabama? |
| 03:12.32 | Katty | russellb: south carolina? |
| 03:13.00 | Katty | russellb: sea of tranquility! |
| 03:13.03 | Katty | russellb: north shore. |
| 03:14.27 | russellb | um ... |
| 03:14.31 | russellb | one of those was correct |
| 03:14.34 | russellb | and that is ... |
| 03:14.43 | russellb | south carolina |
| 03:14.45 | Corydon76-dig | Planet Earth |
| 03:14.48 | russellb | but not the SC on earth |
| 03:14.55 | Corydon76-dig | Rats |
| 03:15.05 | Qwell | The one on Xenu? |
| 03:15.10 | russellb | Qwell: bingo |
| 03:15.13 | Katty | russellb: i face booked your university real quick |
| 03:15.20 | russellb | heh, i figured |
| 03:15.31 | russellb | stalkernet |
| 03:15.36 | Katty | resourcefulnet. |
| 03:15.48 | CCFL_Man2 | tis a shame michael jackson went off the deep end |
| 03:16.21 | CCFL_Man2 | always likes his 80s stuff |
| 03:16.27 | Qwell | hmm. Xenu isn't a planet. I fail. |
| 03:16.28 | Qwell | ~xenu |
| 03:16.29 | jbot | His name was Xenu, he used renegades. Various misleading data by means of circuits was placed in the implants. the Arch enemy of Scientology, or xenu.net (cool site! aka Operation Clambake) |
| 03:16.42 | [TK]D-Fender | CCFL_Man2, The Jackson's dysfunction? NO.... they jsut give each other new heads for Christmas. |
| 03:17.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 03:17.48 | Katty | he has a scary nose. |
| 03:17.58 | Katty | straight from anime nose. |
| 03:18.06 | tengulre | I can not open the website http://www.markocam.com, what's happen? |
| 03:19.41 | Corydon76-dig | tengulre: you know, he does occasionally take down his machine for maintenance... |
| 03:22.25 | russellb | Corydon76-dig: lies! the markocam runs on l33tness |
| 03:22.27 | Katty | i'm thinking naptime sounds good. |
| 03:22.56 | *** join/#asterisk xenonex (n=xenonex@88.204.243.227) |
| 03:23.06 | Katty | later |
| 03:26.27 | CCFL_Man2 | [TK]D-Fender: i think he does have talent, or atleast did |
| 03:29.27 | CCFL_Man2 | i remember the 80s |
| 03:29.36 | SwK | OH NO!!@#! NOT XENU!!#@ |
| 03:29.44 | CCFL_Man2 | i liked the 80s |
| 03:30.15 | SwK | looks like he gotta new DHCP lease |
| 03:31.26 | CCFL_Man2 | SwK: in your experience, have rotary phones ever been connected to the fxs of a adtran TA850? |
| 03:32.49 | SwK | CCFL_Man2, i dont see why it would not work... i have used pulse dialing on TA750s |
| 03:33.12 | *** part/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 03:33.38 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 03:33.45 | SwK | hah |
| 03:33.49 | SwK | what was my exit message? |
| 03:34.06 | CCFL_Man2 | SwK: you remember what kind of phones? |
| 03:34.13 | CCFL_Man2 | 23:33 -!- SwK [n=SwK@user-24-214-55-149.knology.net] has left #asterisk ["Leaving"] |
| 03:34.32 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
| 03:38.21 | *** part/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 03:40.48 | *** join/#asterisk Telamon (n=telamon@blk-89-240-175.eastlink.ca) |
| 03:41.05 | SwK | CCFL_Man2, it was just a pulse dial phone |
| 03:41.26 | SwK | it was like an old 500 set |
| 03:41.30 | Telamon | Does anyone know what version of the Polycom 501 added microbrowser support? I'm trying to find it, but when I press the Applications button nothing happens... |
| 03:41.32 | SwK | or 250 set |
| 03:42.05 | CCFL_Man2 | SwK: ahh, nice |
| 03:42.17 | CCFL_Man2 | G3 handsets are nice |
| 03:43.00 | [TK]D-Fender | Telamon, its not what version of the phone, its what version of the SIP Firmware. |
| 03:43.10 | [TK]D-Fender | Telamon, and that'd be SIP 2.1.0 |
| 03:43.36 | [TK]D-Fender | Telamon, They come stock with 1.5.3 or so IIRC. |
| 03:44.27 | Telamon | [TK]D-Fender: Thanks. :) Is the 2.1 stable with Asterisk 1.4 now? I remember when I was looking at the firmwares that there were warnings not to go past 1.6.7 due to RFC compliance issues between Asterisk and the newer firmwares. I think the postings were from back in 2006 though. |
| 03:45.09 | [TK]D-Fender | Telamon, the only issue with 2.0 (not above) was a XPDF SIP Presence messaging SNAFU that was rather quickly corrected. |
| 03:45.22 | [TK]D-Fender | Telamon, and that was like 2 years ago |
| 03:45.55 | Telamon | Okay, so running the latest BootROM and SIP version I can find will not cause problems? |
| 03:46.45 | [TK]D-Fender | Telamon, I've run through 2.2.0 personally just fine. 2.2.2 is the latest of that stream, and I have 3.0.0 but have not put into service yet |
| 03:48.31 | Telamon | So SIP 2.2.0 with what bootrom? |
| 03:48.46 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 03:50.11 | [TK]D-Fender | Telamon, I use the base one. Its compatible and I see no reason to change it. |
| 03:50.13 | *** join/#asterisk shido6 (n=shido6@74-130-50-233.dhcp.insightbb.com) |
| 03:50.52 | Telamon | [TK]D-Fender: Cool. Thanks. You saved me hours of Googling. :) |
| 03:51.52 | russellb | [TK]D-Fender == Google |
| 03:51.59 | Nugget | http://i29.tinypic.com/2j4vgaa.jpg |
| 03:51.59 | *** join/#asterisk _pepo_ (n=Pepo@201.238.190.153) |
| 03:52.02 | [TK]D-Fender | ~[TK]D-Fender |
| 03:52.03 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 03:52.10 | russellb | nice |
| 03:52.19 | [TK]D-Fender | :D |
| 03:52.39 | Telamon | On a somewhat unrelated note, did your phones come with the screws to mount the base to the phone? I just got two of them to test, and neither of them did... I think I got some refurbs... |
| 03:53.44 | [TK]D-Fender | Telamon, Screws? No, its just plastic tabs to click into place... |
| 03:55.09 | Telamon | Hmm, do you have 501's or 6xx's? |
| 03:55.27 | [TK]D-Fender | Telamon, both. |
| 03:55.34 | [TK]D-Fender | Telamon, I have my IP 501 next to me now. |
| 03:55.53 | [TK]D-Fender | Telamon, 301 bedside, 600's & 430's at the office. |
| 03:56.02 | Telamon | Hmm, guess I'll have to actually read the assembly guide. Mine looks like it requires screws... |
| 03:56.14 | [TK]D-Fender | Telamon, :/ |
| 03:56.56 | *** part/#asterisk ph0ne (n=ph0ne@76-10-156-106.dsl.teksavvy.com) |
| 04:02.00 | *** join/#asterisk ccvp (n=ccvp@24.214.126.81) |
| 04:02.51 | *** join/#asterisk xanderp (n=pphillip@98.220.191.223) |
| 04:04.53 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 04:06.24 | xanderp | Having problems getting sip trunks to register. I've configured a stanaphone trunk but can't get it to register. (shows Request Sent for ages, then Unregistered). My * machine is in my DMZ, I have also pointed 5060, 10001-20000 hard coded at it in the forwarding tables of my router. I think I correctly configured my internal and external ip's as well. I've got my extensions set with nat=yes and also the trunk. Help please! |
| 04:06.38 | *** join/#asterisk CpuID2 (n=nathan@gentoo/contributor/cpuid) |
| 04:06.51 | cmantito | xanderp: stanaphone hasn't worked for me at all for months. |
| 04:06.58 | xanderp | hmmm |
| 04:07.02 | CpuID2 | hey ppls anyone know how i can get the Monitor() command to accept some form of placeholder so i end upw ith unique recording filenames for each call? |
| 04:07.02 | xanderp | how about gizmo? |
| 04:07.09 | cmantito | haven't used. |
| 04:07.16 | CpuID2 | ive tried %d as just a test, even an incrementing number will do, just so they dont overwrite eachother |
| 04:07.29 | xanderp | trying to setup a free inbound trunk to test. |
| 04:07.36 | cmantito | xanderp: try FWD |
| 04:07.38 | xanderp | (first * box) |
| 04:07.52 | xanderp | do they give you a pstn #? |
| 04:07.55 | [TK]D-Fender | xanderp : pastebin > thinking. |
| 04:08.06 | cmantito | xanderp: not that I'm aware of. |
| 04:08.07 | [TK]D-Fender | ~pb |
| 04:08.07 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:08.09 | [TK]D-Fender | ^^^^^^^^^^ |
| 04:08.14 | ccvp | Tha KillaZ |
| 04:08.23 | Telamon | [TK]D-Fender: Hmm, pressing Applications still doesn't do anything... Any ideas as to why? |
| 04:08.43 | xanderp | [TK]D-Fender: sorry.. |
| 04:08.47 | cmantito | CpuID2: * The base filename to use when saving the sound files. If not supplied, the default basename is constructed on the channel name plus a number, for example, IAX2[foo@bar]-3. The channel's input voice packets will be saved to basename-in.ext and the output voice packets will be saved to basename-out.ext. The default location for saved files is the /var/spool/asterisk/monitor directory. -- from http://www.voip-in |
| 04:08.47 | cmantito | fo.org/wiki-Asterisk+cmd+monitor |
| 04:08.51 | cmantito | gahh |
| 04:08.57 | [TK]D-Fender | Telamon, you'll need to check your provisioning to ensure you set a home page, etc |
| 04:10.20 | Telamon | Yep, got that (checked the web interface) I just can't seem to start the microwbroser on the phone. I thought I'd lust have to hit the Applications button and it would come up in a menu, but nothing. |
| 04:17.34 | *** join/#asterisk putnopvut (n=putnopvu@76.29.245.9) |
| 04:17.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 04:18.08 | xanderp | Any thoughts about vbuzzer? From what I just read it appears to have free pstn to sip. |
| 04:20.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 04:21.21 | drmessano | Looks like 2.26 a month to me |
| 04:21.50 | xanderp | drmessano: vbuzzer? |
| 04:21.55 | drmessano | yes |
| 04:22.33 | xanderp | didn't see that. |
| 04:22.50 | drmessano | Its on the homepage |
| 04:23.13 | xanderp | crap... one of these days I will learn to read. |
| 04:23.47 | xanderp | oh well, i can still test getting a trunk configured. ( I think my system isn't wired up right) |
| 04:24.04 | drmessano | ok |
| 04:28.39 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 04:29.03 | Robba | Hi Guys, is anyone using SPA 941/2's? |
| 04:33.18 | xanderp | hmm well off to find another "free" trunk provider. can't even test with vbuzzer, as their inbound settings appear to be tied to the paid pstn number. |
| 04:33.55 | JT | no such thing as a free pstn provider |
| 04:33.59 | JT | there's always a catch |
| 04:34.25 | xanderp | gizmo? |
| 04:34.40 | xanderp | stanaphone? |
| 04:36.11 | xanderp | if only grandcentral would offer up sip connections... they don't seem to mind paying for both ends of the call :) |
| 04:36.32 | JT | as i said |
| 04:36.37 | JT | there's always a catch |
| 04:36.40 | drmessano | IPKALL is sweet |
| 04:36.47 | JT | if you want real pstn connectivity, you have to pay |
| 04:36.47 | drmessano | You can always use Grandcentral |
| 04:36.59 | drmessano | and tie it to a Gizmo account |
| 04:37.11 | xanderp | did that... |
| 04:37.31 | cmantito | I'll sell you a DID ;) |
| 04:37.46 | xanderp | Come to think of it... I reinstalled after I tried gizmo. I'll try to setup a gizmo trunk again. |
| 04:38.08 | xanderp | I just need the rest of the world to give up pstn! ;) |
| 04:38.14 | cmantito | hear hear |
| 04:38.21 | drmessano | I think someone found out you cant use an IPKALL number with Grandcentral |
| 04:38.25 | drmessano | which sux |
| 04:38.52 | xanderp | you also can't use some others as well.. (doesn't do dtmf correctly for verification calls. |
| 04:39.29 | drmessano | Has nothing to do with DTMF |
| 04:39.55 | *** join/#asterisk dmz (n=dmz@64.253.5.180) |
| 04:40.36 | xanderp | ? i thought the keys didn't work because they were compressing the touch tones and the other end couldn't recognize what was being sent. |
| 04:41.21 | drmessano | IPKALL doesn't work becuase of the way the calls are routed.. I'm not discussing the DTMF, i'm discussing why IPKALL doesn't work with Grandcentral |
| 04:41.37 | xanderp | ahh |
| 04:41.50 | CCFL_Man2 | can you change the sip port asterisk uses? and use that new port with ipkall? |
| 04:42.03 | drmessano | huh? |
| 04:46.52 | *** join/#asterisk jdg (n=jdg@203.185.176.20) |
| 04:50.55 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 04:58.00 | xanderp | I think I had tried wiring up a phonegnome account and that's the one that wouldn't verify by DTMF |
| 05:00.48 | xanderp | so is ipkall pstn to sip? |
| 05:14.13 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 05:19.04 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 05:20.17 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 05:21.56 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 05:22.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-223c3968fa82b535) |
| 05:23.57 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 05:27.16 | *** join/#asterisk hohum (n=dcorbe@70-0-155-95.area1.spcsdns.net) |
| 05:28.51 | xanderp | hmmm i signed up for an ipkall, but made it an iax account, now can't get their website to change it to sip. There's not much of a website to work with from thepage i accessed. |
| 05:49.11 | xanderp | help please... i got an incoming call to go to my * box, but instead of ringing the extension, it's going straight to VM. the line i have is http://pastebin.com/d39aa166a |
| 05:51.17 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 05:52.04 | *** join/#asterisk xenonex (n=xenonex@88.204.243.227) |
| 05:53.01 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 05:54.41 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 05:55.21 | Mavvie | xanderp: are you sure that SIP/201 is correct? |
| 05:57.51 | *** join/#asterisk chendy (n=chendy@59.40.223.127) |
| 06:06.04 | xanderp | that's the extension i want to ring... |
| 06:06.18 | xanderp | (sorry for the delay... was on another window) |
| 06:07.00 | xanderp | I just called the number and it has my 'temporary greeting' being played, and then prompt to leave vm. |
| 06:10.21 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 06:10.34 | s0lid | Hi |
| 06:10.41 | s0lid | i need to do this on my dialplan |
| 06:10.45 | s0lid | here's the scenario |
| 06:11.21 | s0lid | the company needs add numbers at the end of the number for authorization codes |
| 06:11.22 | s0lid | like |
| 06:11.24 | s0lid | dialling |
| 06:11.30 | s0lid | 1800555121212345 |
| 06:11.38 | s0lid | the 12345 will be the authentication number |
| 06:11.47 | s0lid | if the last 5 digits is correct the call will go through |
| 06:12.16 | s0lid | i wanted to catch the last 5 digits and authenticate it through checking the dbase |
| 06:24.01 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:25.14 | *** join/#asterisk kalevra (n=dallas@76.185.52.163) |
| 06:26.07 | kalevra | I have a client with an existing analog PBX which has failed. I would like to replace it with an Asterisk server but am curious if I can use it on an analog system? |
| 06:26.08 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 06:29.38 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 06:40.10 | Mavvie | pbx_spool.c |
| 06:40.22 | obnauticus | lulz: http://digg.com/software/Microsoft_Firefox_2007_Professional_Edition_2 |
| 06:41.45 | CoffeeKid | Does anyone know if you can start a call using a call file on a local extension, and then dial out to an external extension (So i can get the DIALSTATUS of the outgoing call)? |
| 06:42.30 | CoffeeKid | Or is there some easy way to get the dialstatus of an outgoing call file? |
| 06:43.05 | Mavvie | CoffeeKid: check the http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#Thefailedextension |
| 06:44.00 | CoffeeKid | Mavvie: will that show the status of busy, noanswer, etc... like ${DIALSTATUS} does? |
| 06:44.43 | Mavvie | CoffeeKid: only one way to find out. |
| 06:45.27 | CoffeeKid | yeah, testing now |
| 06:47.18 | CoffeeKid | Mavvie: the problem is.. how do i know WHY it failed? |
| 06:47.47 | CoffeeKid | Mavvie: the docs aren't real specific to anything other then you can just do stuff when it failed.. what if i need the reason for it failing? |
| 06:47.50 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 06:47.56 | joelsolanki | Hi room |
| 06:47.58 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 06:48.18 | *** join/#asterisk marthag (n=martha@202.138.170.146) |
| 06:48.31 | joelsolanki | codec_g729a_v34_c3.tar.gz for which machine this codec is ? |
| 06:48.32 | joelsolanki | celeron ? |
| 06:48.40 | l0verb0y | hey does anyone if asterisk can detect answering machine beeps? |
| 06:48.42 | joelsolanki | i need g729 codec for celeron. |
| 06:49.01 | Mavvie | CoffeeKid: did you check if the DIALSTATUS was available? |
| 06:49.15 | hi365 | anyone ggod with spech grammers? For some reson my grammer fails with the error: Failed to load grammar 'company_directory': Syntax is not understandable |
| 06:49.27 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 06:49.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f339fd39660594eb) |
| 06:49.46 | joelsolanki | any hints plz ? |
| 06:50.01 | joelsolanki | codec_g729a_v34_c3.tar.gz is this for celeron ? |
| 06:50.05 | CoffeeKid | Mavvie: I did.. its not available.. because the Dial() command is never called.. by the time you reach the context/priority the call is answered |
| 06:52.58 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 06:53.21 | CoffeeKid | Mavvie: check this out.. i got: Got SIP response 503 "Service Unavailable from X.X.X.X" |
| 06:53.40 | CoffeeKid | Mavvie: how do i get that from the channel? :) |
| 06:55.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 06:56.15 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 07:01.32 | *** join/#asterisk ManxPower (n=manxpowe@127.sub-75-202-166.myvzw.com) |
| 07:01.35 | *** join/#asterisk esaym (n=user@72.183.198.134) |
| 07:01.48 | marthag | so, I upgraded a server from 1.2 to 1.4.18. However, asterisk is hanging about once an hour. What is the recommended version to downgrade to? |
| 07:02.24 | Mavvie | I would go to 1.4.10 and see if the problem is still there, and then go up one version at the time. |
| 07:03.01 | CoffeeKid | marthag: we've been using 1.4.15 on our servers, and its been running well |
| 07:03.29 | marthag | This is a production server, so I can't play around |
| 07:03.43 | Mavvie | but can you afford the hangings? |
| 07:04.01 | CoffeeKid | we use 1.4.15 on production |
| 07:05.24 | marthag | I use 1.4.9 on other server, but I see occasion crashes there :-( |
| 07:06.08 | Mavvie | I use 1.4.>10 for a long time without too much problems. |
| 07:06.18 | Mavvie | and the only problem we had was already in 1.4.0 |
| 07:06.22 | Mavvie | which is now resolved |
| 07:06.24 | Mavvie | *yay* |
| 07:13.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:18.20 | hi365 | anyone have any link for vreating srgs grammers? |
| 07:20.45 | *** join/#asterisk drcode (i=55410c9c@gateway/web/ajax/mibbit.com/x-b6461db674184d9d) |
| 07:20.52 | drcode | hi all |
| 07:21.20 | drcode | Is there testing tools for VOIP network? |
| 07:26.00 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:26.07 | drcode | hi |
| 07:26.52 | tzafrir | marthag, anything in the logs? |
| 07:27.16 | drcode | Is there testing tools for VOIP network? I want to run voip somthly on our network |
| 07:37.24 | marthag | I don't see anything in the logs |
| 07:37.37 | drcode | hi marthag |
| 07:37.41 | marthag | hi |
| 07:37.59 | drcode | I have some network in my office |
| 07:38.09 | drcode | I want to check that voip will work smothly |
| 07:38.19 | drcode | and what routers have voip problem |
| 07:38.23 | drcode | is there such tools? |
| 07:39.06 | marthag | drcode: I don't know of such a tool |
| 07:40.23 | *** join/#asterisk shital (n=shital@122.167.78.123) |
| 07:41.02 | shital | Good Afternoon AAll |
| 07:41.30 | shital | m newbie to this asterisk channel |
| 07:41.48 | shital | nice to see u all online |
| 07:43.11 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 07:43.29 | drcode | so how can I know where there are voip network problems? |
| 07:43.36 | drcode | I mean like jitters and so |
| 07:43.58 | tzafrir | drcode, another Asterisk server |
| 07:44.14 | tzafrir | or a bunch of them |
| 07:44.23 | tzafrir | to generate / terminate test calls |
| 07:44.45 | drcode | hi tzafrir |
| 07:44.52 | drcode | I think we talked before |
| 07:44.54 | drcode | whats up |
| 07:45.06 | drcode | yes |
| 07:45.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:45.19 | drcode | but still its hard to find where spcifice the problem |
| 07:45.22 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 07:45.31 | tzafrir | What problem? |
| 07:45.34 | drcode | also if its big privat lan + wan |
| 07:46.36 | *** join/#asterisk lesouvage (n=lesouvag@82.74.22.59) |
| 07:48.40 | DarKnesS_WolF | anyone did fax over IP with asterisk ? |
| 07:48.46 | DarKnesS_WolF | tzafrir: helllo ! long time didn't see u :-) |
| 07:49.04 | tzafrir | DarKnesS_WolF, hi :-) |
| 07:53.45 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 08:00.26 | yang | DarKnesS_WolF: yes, its working to some extent |
| 08:00.44 | yang | DarKnesS_WolF: we use that daily |
| 08:00.57 | *** join/#asterisk gego (n=ubuntuus@b238085.customer.hansenet.de) |
| 08:01.01 | yang | hey gego |
| 08:01.19 | gego | hi, that was fast! |
| 08:02.22 | gego | yang> are you any further reg the pickups? |
| 08:02.27 | DarKnesS_WolF | yang: can u tell me mor about ur setup ? |
| 08:03.03 | yang | DarKnesS_WolF: we have it connected to an asterisk over ATA device (you know the little box where you put in analogue FXS) |
| 08:03.44 | yang | gego: I haven't tested since yesterday, and I cannot do it now since people are using the phones |
| 08:03.55 | *** join/#asterisk oej (n=olle@62.97.243.70) |
| 08:04.06 | yang | gego: I haven't thinked up of any solution |
| 08:04.09 | DarKnesS_WolF | yang: nop that is no IP :-) |
| 08:04.27 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:04.34 | DarKnesS_WolF | i want to have like PSTN reciveds the fax to a termination over IP and that termination send using TDM |
| 08:05.14 | yang | DarKnesS_WolF: asterisk<->IP<->ATA |
| 08:05.52 | yang | DarKnesS_WolF: well tell me what you figure out, i am also interested how to enable FAX over SIP |
| 08:07.01 | yang | DarKnesS_WolF: are you planing to use a FAX software or a fax machine at your end? |
| 08:09.17 | DarKnesS_WolF | yang: fax machine for now |
| 08:09.21 | DarKnesS_WolF | yang: later can be anything :-) |
| 08:10.25 | shital | hello all |
| 08:10.40 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:11.18 | shital | please any one can tell me, wat bare minimum packages an Asterisk needs |
| 08:11.55 | shital | so that seperate distro can be created |
| 08:16.38 | tzafrir | shital, for what linux distribution? |
| 08:17.18 | tzafrir | If you want to "build your own", look into the build system of astlinux |
| 08:17.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:18.06 | tzafrir | but if you want to work one something alike of Debian or Centos, there are better alternatives |
| 08:21.53 | obnauticus | If I have a2billing defining sip friends and peers, how do i have another asterisk node in another portion of the world use that configuration? |
| 08:21.59 | obnauticus | and how can i do that with asterisk realtime |
| 08:22.06 | obnauticus | (this would be the same as using a remote extensions.conf) |
| 08:23.55 | WilliamK | morning tzafrir |
| 08:32.03 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 08:34.07 | shital | yaa tzafrir it may be any distro but just i want to create my own distro for asterisk |
| 08:34.44 | shital | this may not have all the features that now asterisk is having |
| 08:35.10 | tzafrir | shital, what Linux variant are you most comfortable with? |
| 08:35.58 | shital | basically i worked on debian and redhat based systems |
| 08:36.07 | shital | any one it can be |
| 08:37.03 | shital | my aim is just selecting backend programs from asterisk, for calls to happen |
| 08:37.34 | shital | ie make and receive calls with analog telephones |
| 08:38.11 | shital | and club these with minimum packages so that it can be a distro |
| 08:38.27 | shital | i dont need GUI part and all |
| 08:38.46 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 08:39.01 | shital | i mean just i want in runlevel 3 ie console based |
| 08:39.20 | shital | can u please give me ideas? |
| 08:39.34 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 08:40.24 | mvanbaak | start with a very basic debian install |
| 08:40.31 | mvanbaak | and do: apt-get install asterisk |
| 08:40.37 | mvanbaak | that will tell you what packages you need |
| 08:41.08 | shital | yaa correct but it has GUI parts no, its not necessary |
| 08:41.15 | mvanbaak | no, it has not |
| 08:42.06 | shital | i mean, say for ex ubuntu if i install, it has definitly X server |
| 08:42.17 | shital | wich i wont need |
| 08:42.34 | *** join/#asterisk ktwilight[m] (n=ktwiligh@171.235-241-81.adsl-static.isp.belgacom.be) |
| 08:42.49 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 08:42.52 | mvanbaak | not if you install ubuntu server |
| 08:42.53 | ktwilight[m] | hi :) |
| 08:43.03 | mvanbaak | if you use the desktop iso it will have X indeed |
| 08:43.12 | mort_gib | shital: same goes for CentOS server |
| 08:43.12 | mvanbaak | that's the whole point of the _desktop_ |
| 08:43.22 | shital | then? |
| 08:43.24 | ktwilight[m] | in our office, we have a really old siemens HiCom OfficePro PBX, am wondering if it's possible to use Asterisk to connect and configure the PBX? |
| 08:43.52 | mvanbaak | shital: download the Debian netinst cd image, install only the base system, and run apt-get install asterisk |
| 08:43.56 | mvanbaak | there you go |
| 08:44.03 | mvanbaak | clean install, no gui, only the needed packages |
| 08:44.17 | shital | ok ok |
| 08:44.45 | mvanbaak | that will give you an idea of what packages are needed |
| 08:44.55 | mvanbaak | and you can start building your own distro based on that |
| 08:44.56 | shital | it dont have any unecessary packages or drivers rt?, except wat i need for asterisk function? |
| 08:48.12 | shital | mvanbaak, have u seen Aastlinux distro? |
| 08:48.21 | mvanbaak | shital: nope |
| 08:48.37 | shital | its size is only 95 mb |
| 08:48.46 | Mavvie | ${CALLINGPRES} maybe. |
| 08:49.02 | shital | it will boot well to console mode |
| 08:50.11 | Mavvie | that might be the trick. |
| 08:51.49 | Mavvie | hmmm... that is incoming calls. |
| 08:53.05 | *** join/#asterisk tsearle (n=torrey@98.110-246-81.adsl-static.isp.belgacom.be) |
| 08:53.40 | gego | is there a way to observe CLI output periodically - like watch in shell ? |
| 08:53.57 | *** join/#asterisk noneo (n=artur@82.152.82.149) |
| 08:56.34 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-36f66134fc3ea562) |
| 08:59.29 | *** join/#asterisk s0lid (n=s0lid@202.179.107.82) |
| 09:01.42 | obnauticus | is ulaw < g729? |
| 09:03.00 | *** join/#asterisk xenonex (n=xenonex@89.218.235.65) |
| 09:05.54 | *** join/#asterisk xenonex (n=xenonex@89.218.235.65) |
| 09:07.42 | gego | obnauticus> what is a < relation with regard to codecs? in lexicographical order ulaw > g729 since u > g :-) |
| 09:10.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 09:11.22 | *** join/#asterisk xenonex (n=xenonex@89.218.235.65) |
| 09:15.17 | obnauticus | what is a good E1 card? |
| 09:16.00 | obnauticus | that can handle al ot of calls |
| 09:16.30 | *** join/#asterisk tsearle (n=torrey@98.110-246-81.adsl-static.isp.belgacom.be) |
| 09:16.36 | Mw3 | anything not from digium at least if you would like to deal with faxes |
| 09:16.44 | obnauticus | can you suggest one |
| 09:16.48 | obnauticus | i don't have experience picking these out |
| 09:16.52 | mvanbaak | obnauticus: digium TE121/TE122 or Sangoma A101 |
| 09:16.56 | obnauticus | it should be able to handle 23 concurrent cards |
| 09:16.56 | obnauticus | lol |
| 09:16.57 | obnauticus | calls* |
| 09:20.16 | *** join/#asterisk xenonex (n=xenonex@89.218.235.65) |
| 09:20.30 | obnauticus | also mvanbaak do you think a datacenter will route to your cage a T1.... |
| 09:20.32 | obnauticus | i dunno |
| 09:21.31 | *** join/#asterisk antxoneti (n=antonio@98.110-246-81.adsl-static.isp.belgacom.be) |
| 09:21.47 | shital | which Digium cards i should use for setting up calls to make and receive |
| 09:21.55 | obnauticus | FXO and FXS cards. |
| 09:22.12 | shital | MODEL No? |
| 09:22.14 | obnauticus | FXS is to get the signal from your providor (station) FXS is to provide the signal. |
| 09:22.25 | obnauticus | MODEL Digium-Go-ogl-e |
| 09:22.29 | obnauticus | sorry |
| 09:22.35 | obnauticus | digium-Go-ogl-eit |
| 09:23.10 | *** join/#asterisk xenonex (n=xenonex@89.218.235.65) |
| 09:23.33 | shital | simply i just want one FXO for incoming and 2 to 3 extensions i want to use |
| 09:32.47 | obnauticus | shital fxo's arne't generally used for incomming... |
| 09:33.13 | obnauticus | think of it this way you receive on an FXS port, and you send out on an FXO |
| 09:33.24 | shital | ok obnautics |
| 09:33.52 | shital | srry in reverse |
| 09:33.52 | obnauticus | my last statment was wrong |
| 09:33.56 | obnauticus | FXO is to provide the signal* |
| 09:34.08 | shital | yes |
| 09:35.13 | *** join/#asterisk RoyK (n=roy@80.239.107.100) |
| 09:35.15 | awk | ok great, we have the queue patch now... so its just iax and we should have a stable version of asterisk.. will get lock info when it happens again |
| 09:35.30 | obnauticus | 1.6? |
| 09:35.44 | obnauticus | i can't wait for SIP over TCP |
| 09:35.58 | shital | obnautics : to support FXO and FXS to make and receive calls want have all bare minimum packages |
| 09:35.58 | obnauticus | and shital this may be what you want |
| 09:36.01 | obnauticus | http://store.digium.com/productview.php?product_code=1TDM422EF <-- |
| 09:36.21 | obnauticus | it has 2 FXO ports, and 2 FXS ports. |
| 09:36.28 | shital | yes |
| 09:36.37 | shital | <PROTECTED> |
| 09:37.13 | shital | so that after compiling i want to port that onto seperate board |
| 09:37.38 | shital | a product like, wat distro i have to take now |
| 09:37.57 | shital | i dont want any GUI part and all |
| 09:41.02 | tzafrir | shital, the problem with your approach is that you don't exactly know what else you need |
| 09:41.23 | tzafrir | For instance, what do you want to do with CDRs? |
| 09:42.04 | tzafrir | Also, do you want to provide any voip capability? |
| 09:42.35 | tzafrir | voip means much configurability |
| 09:46.29 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:48.31 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 09:51.49 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:51.59 | dandre | Hello, |
| 09:55.05 | dandre | I have tested to call with my sip accounts and this works fine, but I can't receive incomming calls. |
| 09:55.06 | dandre | I have nothing with sip debug on and the sip show registry show these accounts as registred. What can I do? |
| 09:55.06 | *** join/#asterisk Zhad (n=tom@cpc4-sout6-0-0-cust165.sotn.cable.ntl.com) |
| 10:05.20 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.77.32) |
| 10:07.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:13.07 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
| 10:14.37 | *** join/#asterisk potsboy (n=chrisg@196.212.121.210) |
| 10:20.27 | zeeesh | i m trying to compile/install "asterisk-addons-1.4.6" by this way ... "./configure" but it does't work.. how to do it? |
| 10:20.38 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 10:23.29 | potsboy | zeeesh: it doesnt "work"??? |
| 10:24.14 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 10:25.05 | zeeesh | no sorry .. it works |
| 10:26.11 | *** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au) |
| 10:33.00 | ManxPower | zeeesh: I think he was pointing out your total lack of information |
| 10:34.32 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-206-182.bltmmd.east.verizon.net) |
| 10:34.33 | zeeesh | trying to install asterisk-addons-1.4.6 i m on step "make menuselect" where i m looking the 2nd option "2. Call Detail Recording" wherer i m looking "XXX 1. cdr_addon_mysql"... which shows some dependencies r missng or module missing .. how to check it ??/ |
| 10:35.22 | SteveTotaro | do you have mysql and mysql-devel installed? |
| 10:37.11 | SteveTotaro | mysql-libs? |
| 10:38.39 | zeeesh | yeah i hv installed mysql-5 .. and i got the missing depends..... which r "libncurses" "libmxml" am i right ? |
| 10:44.25 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 10:45.09 | *** join/#asterisk Chris-NB (n=chris@213.162.66.152) |
| 10:48.13 | michael-i | I'm trying to create a pattern which will match extensions containing the digits 0-9 as well as * and #. My pattern so far is _[*#X]! which is matching "*1234" but not "00001234" |
| 10:48.51 | michael-i | Am I missing something or are there special considerations for the * and # characters? |
| 10:49.20 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 10:54.06 | *** join/#asterisk kinom (n=jonathan@host182-44-static.48-88-b.business.telecomitalia.it) |
| 10:57.20 | michael-i | ...or the X isn't interpreted inside brackets :) got it |
| 11:00.04 | dandre | where can I find chan_misdn ? |
| 11:01.04 | tzafrir | channels/chan_misdn.c |
| 11:01.26 | marthag | dammit. I can't figure out why this new server is rejecting its IAX connections half the time |
| 11:01.55 | marthag | and if debug mode is telling me anything useful, I can't see it |
| 11:02.22 | tzafrir | marthag, does it get to a dialplan? |
| 11:02.29 | marthag | nope |
| 11:03.40 | dandre | ok tzafrir, sorry. |
| 11:03.53 | tzafrir | pastebin the debug and hope someone here will have a clue |
| 11:04.35 | dandre | tzafrir, I have tested to call with my sip accounts and this works fine, but I can't receive incomming calls. |
| 11:04.35 | dandre | I have nothing with sip debug on and the sip show registry show these accounts as registred. What can I do? |
| 11:05.20 | tzafrir | dandre, tcpdump |
| 11:06.02 | dandre | ok, I'll try |
| 11:13.08 | marthag | http://pastie.caboo.se/164892 |
| 11:15.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:17.38 | *** join/#asterisk d00gster (n=doughant@bas1-toronto44-1279461151.dsl.bell.ca) |
| 11:22.33 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:24.07 | *** join/#asterisk ToTo (n=ToTo@209.8.41.75) |
| 11:27.51 | *** join/#asterisk KillerQueen_86 (n=alan@201-212-100-188.cab.prima.net.ar) |
| 11:28.16 | KillerQueen_86 | ~thwack KillerQueen |
| 11:28.17 | jbot | ACTION hits KillerQueen on the groin with a Holy Bible |
| 11:31.15 | *** join/#asterisk ToTo (n=ToTo@209.8.41.74) |
| 11:34.15 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 11:34.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:34.54 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 11:36.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:36.24 | *** join/#asterisk nighty^ (n=nighty@p2071-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 11:37.34 | potsboy | ~thwack bill gates |
| 11:37.34 | jbot | ACTION smacks bill gates on the eblow with a AS/400 |
| 11:37.44 | potsboy | sorry, always wanted to try that :) |
| 11:38.03 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 11:38.06 | slavon_net | hello all |
| 11:38.22 | *** join/#asterisk ToTo (n=ToTo@207.176.6.66) |
| 11:38.45 | slavon_net | have some problem... after press * from sip to zap asterisk wait 1-2 sec before translate DTMF to zap.... |
| 11:38.49 | slavon_net | how to fix it? |
| 11:39.21 | potsboy | try *# |
| 11:40.31 | slavon_net | no... need fix without pressing any keys |
| 11:41.19 | awk | have you tried using relaxed dtmf |
| 11:41.26 | *** join/#asterisk xenonex (n=xenonex@89.218.235.65) |
| 11:41.40 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 11:41.42 | potsboy | relaxes dtmf, oftern is triggered by voice |
| 11:45.26 | *** join/#asterisk ToTo (n=ToTo@209.8.41.64) |
| 11:48.16 | *** part/#asterisk kinom (n=jonathan@host182-44-static.48-88-b.business.telecomitalia.it) |
| 11:48.35 | KillerQueen_86 | is there a way of changing MONITOR_FILENAME into a queue?? I wanna put the agent_id in the MONITOR_FILENAME but this can be setted before entering the queue (when i currently don't know which agent is going to pick up the phone) |
| 11:54.09 | *** join/#asterisk ToTo (n=ToTo@209.8.41.78) |
| 11:54.52 | KillerQueen_86 | ~McCain |
| 11:55.28 | slavon_net | hmmm... if i press "*1" - ZAP translate "1"... if i press *... pause...1 - zap translate *1... how turnoff.. all *xxxx detections at ZAP driver? |
| 11:57.12 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 11:58.57 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:59.54 | slavon_net | hello.... why *1 DTMF code may is cut? |
| 12:00.13 | potsboy | slavon_net: have you played with the toneduration= setting i assume the phone is analogue |
| 12:00.50 | slavon_net | potsboy nope... DTMF code normal detect by system |
| 12:00.59 | slavon_net | ZAP channel cut only if press *1 |
| 12:01.07 | slavon_net | *[2-0] normal work |
| 12:01.13 | slavon_net | *21111 normal work |
| 12:01.25 | slavon_net | *123 - return 23 |
| 12:01.58 | *** join/#asterisk ToTo (n=ToTo@209.8.41.75) |
| 12:03.35 | sysreq | slavon_net: do you have the 'one touch record' feature enabled in features.conf? it uses *1.. |
| 12:03.48 | slavon_net | nope |
| 12:03.53 | slavon_net | all configs is default |
| 12:04.08 | slavon_net | i dont use DYNAMIC_FEATURES |
| 12:04.19 | slavon_net | also if i call sip-sip all work normal |
| 12:04.32 | slavon_net | truuble only if call go to ZAP channel |
| 12:07.45 | *** join/#asterisk snappy (i=naveen@armakuni.lastninja.net) |
| 12:08.37 | *** part/#asterisk marthag (n=martha@202.138.170.146) |
| 12:11.07 | *** join/#asterisk robmac67 (n=robmacle@82-35-181-109.cable.ubr03.croy.blueyonder.co.uk) |
| 12:11.09 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 12:11.20 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.113) |
| 12:12.05 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 12:12.05 | slavon_net | not have ideas? =( some user have pincode that use *1xxxxxx and its not work =( |
| 12:12.15 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:13.47 | *** join/#asterisk Chris-NB (n=chris@213162066170.public.t-mobile.at) |
| 12:14.05 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:15.12 | *** join/#asterisk Grash (n=grash@244.47.217.87.dynamic.jazztel.es) |
| 12:18.56 | sysreq | slavon_net: it happens with a *1 exten, or when evaluating an input given by a user while on a call? |
| 12:19.13 | slavon_net | on call |
| 12:19.24 | slavon_net | to city |
| 12:20.36 | Grash | what is the simplest way to check a voip provider are alive, and if not jump to the next provider? |
| 12:21.03 | Grash | "IsAvail" ? |
| 12:22.38 | slavon_net | call and check DIALStatus |
| 12:23.12 | slavon_net | or if you use SIP - you can check qualify |
| 12:24.04 | *** join/#asterisk ndlovu (n=rudivs@wbs-41-208-242-62.wbs.co.za) |
| 12:24.13 | slavon_net | SIPPEER(<peername>[|item]) status - Status (if qualify=yes). |
| 12:25.21 | ndlovu | anyone know of a free linux softphone that supports g.729? |
| 12:25.59 | potsboy | ndlovu: g729 is proprietry, if you find one... it wont be legal |
| 12:26.35 | ndlovu | potsboy, ah, that explains it then |
| 12:26.51 | KillerQueen_86 | ~G729 |
| 12:26.51 | jbot | i heard g729 is an ITU-standard voice codec which operates at 8kbps and offers quality very similar to GSM. G.729 is patent-encumbered; those wishing to use it with Asterisk must buy a license from Digium. |
| 12:27.07 | sysreq | slavon_net: so a simple Read(input||4) and NoOp(${input}) where a user would enter '*123' will output '23' if given over a Zap channel while in a call? |
| 12:27.26 | slavon_net | sysreq i use DTMF debug |
| 12:27.53 | slavon_net | and see that if press *1 - its cut |
| 12:28.00 | slavon_net | if *2 its translae |
| 12:28.03 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 12:28.15 | slavon_net | if * pause 1 sec and 1 - translate |
| 12:28.41 | slavon_net | if i call sip to sip - *1 translate |
| 12:29.04 | slavon_net | i think its some function in ZAP (like transfer or another) |
| 12:29.12 | slavon_net | question - what? |
| 12:29.59 | sysreq | i've tried something similar here, and it seems to go through.. |
| 12:30.40 | sysreq | though i have no idea what could cause the problem.. 'just trying to help out ;p |
| 12:31.12 | slavon_net | now may calls... can't debug... =( |
| 12:31.16 | slavon_net | *many |
| 12:32.09 | *** join/#asterisk _ShrikE (n=_ShrikE@74.185.215.60) |
| 12:32.46 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
| 12:35.09 | *** join/#asterisk shinao1 (n=shinao1@35-230.rv.ipnxtelecoms.com) |
| 12:35.30 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582836.dsl.bell.ca) |
| 12:36.29 | ndlovu | can asterisk do codec translation? for example, if my voip provider only supports g.729, can I use a different codec to route calls internally? |
| 12:37.05 | ndlovu | so that I don't have to buy licences for each extension... |
| 12:37.11 | potsboy | ndlovu: you will need license if asterisk is doing codec translation |
| 12:37.17 | potsboy | ndlovu: no per channel |
| 12:37.40 | potsboy | ndlovu: speak to connection-telecom they are in CT, they can advise and price |
| 12:38.31 | potsboy | i believe there is a new office in JHB too |
| 12:38.32 | ndlovu | potsboy, thanks |
| 12:39.28 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 12:39.48 | *** join/#asterisk duckz (n=duckz@81-180-102-217.etth.opensys.ro) |
| 12:40.44 | *** part/#asterisk vap0rtranz (n=jpittman@75.110.17.157) |
| 12:42.11 | yang | Has anyone used Portech GSM gateways ? |
| 12:47.30 | *** join/#asterisk Wi_Fi (n=OUT@76.169.21.21) |
| 12:47.50 | *** part/#asterisk Wi_Fi (n=OUT@76.169.21.21) |
| 12:48.12 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:50.15 | Grash | slavon_net: ChanIsAvail() is specific to check channel availibility, is not? |
| 12:50.52 | Grash | to check a sip provider, is the best option? |
| 12:51.28 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 12:52.55 | KillerQueen_86 | ~slap Grash |
| 12:52.56 | jbot | ACTION slaps Grash, keep your grubby fingers to yourself! |
| 12:55.21 | Grash | what? |
| 12:56.52 | slavon_net | Grash man to SIPPEER(<peername>[|item]) |
| 12:57.04 | slavon_net | see to STATUS item |
| 12:57.29 | Grash | ok |
| 12:57.36 | Grash | thanks |
| 12:57.39 | slavon_net | np |
| 12:57.42 | [TK]D-Fender | slavon_net: Nope.... |
| 12:57.42 | Grash | and sorry for my suggestion |
| 12:57.53 | [TK]D-Fender | slavon_net: Thats only Qualify and is virtually meaningless. |
| 12:57.54 | slavon_net | [TK]D-Fender ? |
| 12:57.59 | *** join/#asterisk mchou (n=mchou@c-71-198-127-234.hsd1.ca.comcast.net) |
| 12:58.14 | slavon_net | [TK]D-Fender its help to see that line to provider is up |
| 12:58.28 | mchou | anyone here happen to have a dlink vta ATA? |
| 12:58.29 | slavon_net | you can get status OK or not |
| 12:58.37 | [TK]D-Fender | slavon_net: That has nothing to do with Channels... |
| 12:58.59 | [TK]D-Fender | slavon_net: And thats assuming you're even using qualify. They could be refusing your calls, etc. |
| 12:59.32 | slavon_net | also can check DIALSTATUS after dial |
| 12:59.43 | Grash | yes |
| 12:59.54 | Grash | but if i make the call, how to put it again if it fail? |
| 13:00.05 | slavon_net | use flag to dial function |
| 13:00.24 | slavon_net | g - if i remember |
| 13:00.34 | *** join/#asterisk ming_zy1 (n=ming_zym@220.181.54.113) |
| 13:00.49 | slavon_net | its not stop dialplan after return of dial |
| 13:00.56 | Grash | yes |
| 13:01.29 | Grash | but if it fail cause anything but provider is down? |
| 13:01.58 | Grash | example, providers problem, not asterisk or my connection to provider. |
| 13:02.43 | slavon_net | check dialstatus... if ISP send to you normal codes - you can see ANSWERED, BUSY or other (FAILED) status and recall to another ISP |
| 13:03.08 | Grash | ok |
| 13:03.12 | Grash | i'll try |
| 13:03.28 | Grash | and, when and why to user ChanIsAvail? |
| 13:03.35 | Grash | and, when and why to use ChanIsAvail? |
| 13:04.14 | slavon_net | its use for check free channels in ZAP |
| 13:04.18 | Grash | ok |
| 13:04.24 | Grash | i suppose |
| 13:04.32 | Grash | thanks |
| 13:04.38 | slavon_net | i also use it for check busy state for multiline phones |
| 13:04.53 | slavon_net | or check if exten is exists in dialplan |
| 13:04.59 | slavon_net | (use LOCAL channel) |
| 13:05.00 | Grash | but not for sip, or other tecnology |
| 13:05.11 | Grash | ? |
| 13:07.41 | slavon_net | i think its wrong use of this function but i not find function like isExtenExists and use it =) |
| 13:09.02 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.136) |
| 13:10.43 | *** join/#asterisk angryuser (n=nononon@df01t2-195-36-136-82.d4.club-internet.fr) |
| 13:11.31 | angryuser | have configured SPA8000 it took me 5 min, working fine |
| 13:11.33 | *** join/#asterisk fskrotzki (n=fskrotzk@64.129.187.131) |
| 13:12.46 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:13.41 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 13:14.42 | Katty | hmmhesays: morning (= |
| 13:14.53 | *** part/#asterisk ndlovu (n=rudivs@wbs-41-208-242-62.wbs.co.za) |
| 13:14.54 | [TK]D-Fender | Katty: Mew. |
| 13:14.59 | hmmhesays | Morning Katty |
| 13:15.11 | hmmhesays | Soon my FC6 box will be no more. I'm excited for FC8 |
| 13:15.27 | Katty | [TK]D-Fender: hai. |
| 13:15.55 | hmmhesays | Katty: you get your stuff figured out yesterday? Sorry i wasn't much help I actually have to work for a living these days |
| 13:16.16 | Katty | hmmhesays: i gave up and called Sangoma. they requested 20 billion log files and conf files and such. |
| 13:16.20 | Katty | hmmhesays: haven't heard back from them yet. |
| 13:16.37 | Katty | hmmhesays: think i'm gonna have the telco do a call trace and watch what the pri does when i dial a number. maybe that'll help |
| 13:17.08 | hmmhesays | Yeah troubleshooting the pri card isn't my specialty I hardly use them. Usually go with sip gateways. |
| 13:17.35 | Katty | all good (= |
| 13:18.01 | hmmhesays | So I finished my awesome directory dialplan yesterday. This thing rocks |
| 13:18.42 | Katty | did you blog it? |
| 13:18.56 | *** join/#asterisk cjk (n=cjk@d90-129-20-198.cust.tele2.lu) |
| 13:19.15 | hmmhesays | I will, when I get my webserver back up. Some j@ckh0le took advantage of it yesterday |
| 13:19.25 | Katty | :< |
| 13:19.28 | cjk | hi, how can i check i an isdn line is in point-to-point mode or point-to-multi-point |
| 13:20.19 | hmmhesays | I takes info from a database and builds an IVR on the fly that asks you want endpoint you would like to contact your user by. IE: mobile number, business number, free world dialup, skype |
| 13:20.43 | Katty | that's neat. |
| 13:21.15 | yang | hey [TK]D-Fender are you using portech products? |
| 13:21.17 | hmmhesays | Yep, I hope to port it over to our friends in the other channel when I get a chance too |
| 13:21.52 | [TK]D-Fender | yang: Never heard of them. |
| 13:22.19 | coppice | lots of suppliers ship poor tech |
| 13:22.28 | yang | [TK]D-Fender: its a badly documented Taiwanesse project |
| 13:22.40 | [TK]D-Fender | yang: YGWYPF |
| 13:22.59 | Katty | jbot: ygwypf? |
| 13:23.00 | jbot | somebody said ygwypf was You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 13:23.09 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:23.26 | yang | [TK]D-Fender: I prefer MBIAS |
| 13:23.35 | yang | My Boss Is A Scottsman |
| 13:23.42 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 13:24.00 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:24.21 | yang | Gives me headaches with these low cost products brrr |
| 13:24.26 | coppice | portech's web site takes me back - "最佳解析度 800*600" :-) |
| 13:24.58 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.225) |
| 13:25.26 | keith4_ | "portech" seems like a bad name choice |
| 13:25.33 | mchou | haha |
| 13:25.35 | yang | coppice: their manual comes on a printed paper sheets |
| 13:25.54 | mchou | yang: why is that so bad? |
| 13:26.04 | coppice | yang: so does a Nortel or Lucent manual |
| 13:26.12 | mchou | yang: at least they have printing :) |
| 13:26.55 | yang | I feel like I would have to learn some chinesse |
| 13:27.12 | mchou | haha |
| 13:27.29 | coppice | most people I know called yang can read chinese :-\ |
| 13:27.49 | mchou | considering they are going to be the dominant economy soon learning Chinese might not be such a bad investment of time :) |
| 13:28.02 | yang | coppice: that is mostly true |
| 13:29.34 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:29.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:30.49 | coppice | 如果你不識看中文,這個台灣 web wites 好難 :-) |
| 13:31.11 | yang | Its the only GSM adapter, which doens't support the incoming calls in default manual *sigh* |
| 13:31.50 | *** join/#asterisk ph0ne (n=ph0ne@206-248-164-27.dsl.teksavvy.com) |
| 13:33.19 | *** join/#asterisk jbigB (n=jBigB@216.207.245.1) |
| 13:34.19 | mchou | I dont get it. |
| 13:34.38 | *** join/#asterisk Lsodi (n=Lsodi@213.168.26.50) |
| 13:34.44 | mchou | what's the product used for? terminating cell phone calls? |
| 13:35.06 | waKKu | folks.. does someone there that can help me with pickupgroups under IAX? Asterisk 1.4 .. I post my confs and CLI output here: http://pastebin.ca/936404 ... any help would be appreciate :) |
| 13:35.08 | yang | no its GSM gateway adapter used to make calls VOIP->GSM and opposite |
| 13:35.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:35.59 | mchou | yang: which market segment would this product be useful for? |
| 13:36.08 | ktwilight[m] | taiwan and hk should form a country :P |
| 13:36.29 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:36.32 | mchou | ktwilight[m]: haha. China wont let them |
| 13:36.32 | yang | mchou: well its usefull if you make VOIP->GSM calls |
| 13:36.59 | mchou | ktwilight[m]: at least not w/o a lot of sabre rattling |
| 13:37.03 | coppice | the taiwanese speak a foreign language. they'd never fit in here |
| 13:37.23 | yang | mchou: or you can for example call from your car with a mobile to GSM adapter, which connects you over VOIP to anywhere |
| 13:37.30 | mchou | coppice: so do most chinese, as a matter of fact. (regional dialects) |
| 13:38.10 | hmmhesays | anyone in here using ip power switches? I need to install one |
| 13:38.14 | mchou | yang: why not just have a sip client on your mobile phone? :) |
| 13:38.35 | ktwilight[m] | mchou, am sure the taiwanese military can die hard against chinese :P |
| 13:38.40 | [TK]D-Fender | coppice: foreign language? That happens every couple hundred miles in Cina as it is... |
| 13:38.43 | yang | mchou: out GSM provider blocks 5060, also you would pay for internet much more even if it was available |
| 13:38.56 | [TK]D-Fender | China* |
| 13:39.01 | yang | mchou: out/our |
| 13:39.15 | ktwilight[m] | foreign language or not, 普通话 rules |
| 13:39.27 | KillerQueen_86 | ~普通话 |
| 13:39.33 | mchou | lol |
| 13:39.37 | KillerQueen_86 | ~china |
| 13:39.38 | jbot | i guess china is a country, or what you eat from on holidays |
| 13:39.56 | mchou | KillerQueen_86: I'd really be surprised if jbot understood that :) |
| 13:40.12 | coppice | 廣東話 is the true Chinese :-) |
| 13:40.13 | ktwilight[m] | only problem is the damn traditional vs simplified ;) |
| 13:40.18 | KillerQueen_86 | hahaha, yep... me too |
| 13:40.21 | ktwilight[m] | coppice, haha |
| 13:41.03 | mchou | coppice: I wasnt even talking about writing, just spoken |
| 13:41.03 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:41.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:41.24 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 13:41.29 | mchou | writing is another bag of problems altogether :) |
| 13:41.44 | ktwilight[m] | not really, writing is the least of all, it's either this or that. |
| 13:41.48 | mchou | not to mention idioms :) |
| 13:41.50 | *** join/#asterisk joshaidan (n=brianj@S0106001c1023e838.tb.shawcable.net) |
| 13:42.03 | ktwilight[m] | dialects and accents are huge problems. |
| 13:42.04 | coppice | this or that or kanji :-) |
| 13:42.14 | mchou | ktwilight[m]: you've obviously never put in an order of something |
| 13:42.27 | ktwilight[m] | meaning? |
| 13:42.32 | mchou | and gotten something else completely |
| 13:42.51 | mchou | due to "writing" |
| 13:43.06 | coppice | technical chinese is a headache between china and taiwan. most terms introduced since 1949 are different :-\ |
| 13:43.12 | ktwilight[m] | whoever wrote it must suck. |
| 13:43.29 | Katty | do do do. |
| 13:43.34 | mchou | ktwilight[m]: nope. writing and idioms make a lethal combo |
| 13:43.34 | Katty | ktwilight[m]: so much negativity! |
| 13:43.35 | ktwilight[m] | of course, there aren't standards, but that doesn't mean it's wrong. it's like UK and US english. |
| 13:43.39 | Katty | ktwilight[m]: don't complain about free stuff. |
| 13:43.46 | ktwilight[m] | Katty, ain't complaining :) |
| 13:44.04 | Katty | ktwilight[m]: well you're ruining my happy complaint-free morning! |
| 13:44.06 | coppice | US English is the wrong one |
| 13:44.09 | nixguy | does reloadin extensions.conf affect a running asterisk? |
| 13:44.09 | mchou | ktwilight[m]: you are sadly mistaken, my friend. US and UK is no comparison |
| 13:44.11 | ktwilight[m] | Katty, :( |
| 13:44.25 | dandre | I have these log from a tcpdump: |
| 13:44.25 | dandre | 14:42:55.361637 provider_ip.5060 > nat_routeur-ip.63681: udp 484 |
| 13:44.25 | dandre | 14:42:55.361745 nat_routeur-ip > provider_ip: icmp: 82.236.48.30 udp port 63681 unreachable [tos 0xc0] |
| 13:44.31 | ktwilight[m] | mchou, really? have you lived in UK before? |
| 13:44.39 | mchou | ktwilight[m]: yup |
| 13:44.46 | ktwilight[m] | mchou, are you brought up with US english? |
| 13:44.54 | mchou | ktwilight[m]: nope |
| 13:45.05 | dandre | How could I fix this to receive inbound call from my sip provider? |
| 13:45.14 | ktwilight[m] | mchou, and how nlong have you lived in UK? |
| 13:45.31 | mchou | ktwilight[m]: my language upbringing is rather mongrel |
| 13:45.40 | ktwilight[m] | mchou, same here |
| 13:46.40 | mchou | but the English vs. Chinese analogy is very far off |
| 13:47.07 | *** join/#asterisk UVSoft (n=jnk467@80.254.48.215) |
| 13:47.10 | UVSoft | hi |
| 13:47.36 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 13:47.42 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:47.51 | ktwilight[m] | mchou, i kinda disagree |
| 13:48.18 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.117.5) |
| 13:48.19 | ZaVoid | morning all |
| 13:48.35 | mchou | ktwilight[m]: get back to me when you've submitted some orders in Chinese :) |
| 13:48.51 | *** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 13:49.17 | mchou | ktwilight[m]: there are no such issues in English |
| 13:49.28 | UVSoft | if asterisk is under the terms of GPL, how lumenvox can hide the code of its res_speech_lumenvox.so? |
| 13:49.32 | davidcsi | anyonw knows exactly what "Spawn extension (whatever, h, 1) exited non-zero on ..." means? |
| 13:49.54 | ZaVoid | doesn't that mean it ran the extension macro fine? |
| 13:50.04 | ktwilight[m] | mchou, heh been there done that |
| 13:50.57 | davidcsi | but does it exits NON-ZERO? |
| 13:51.30 | mchou | ktwilight[m]: think "napkins" and that might be the closest you every come to confusion in US vs. UK English |
| 13:51.41 | mchou | ktwilight[m]: hehe |
| 13:51.51 | *** join/#asterisk ccvp (n=ax@66.0.46.210) |
| 13:52.13 | mchou | ktwilight[m]: let's just say you wouldn't want to confuse those two :) |
| 13:52.19 | hmmhesays | holy crap ip kvm switches are expensive |
| 13:52.26 | mchou | when submitting a purchase order |
| 13:52.33 | ZaVoid | i think it does davidcsi |
| 13:52.35 | ktwilight[m] | mchou, "pants" too ;) |
| 13:52.57 | Katty | hmmhesays: holy crap, vegetables are expensive too! |
| 13:53.02 | ZPertee | will this http://pastebin.ca/939641 work for a .call file or do I need to include more information? |
| 13:53.08 | davidcsi | no, i mean,WHY is it exiting NON-ZERO? that's not an error? |
| 13:53.16 | Katty | ZPertee: hi! |
| 13:53.49 | ZPertee | Katty: mornin |
| 13:55.25 | SwK | any anyone answer the following 2 questions for me |
| 13:55.25 | SwK | does libpri support sending Identity Verify? |
| 13:55.26 | SwK | is libpri stable in TEI assigned state S4? |
| 13:55.54 | hmmhesays | no not really |
| 13:56.10 | hmmhesays | that wasn't for you SwK |
| 13:56.13 | Katty | SwK: my answers are strawberries and blueberries. |
| 13:56.20 | Katty | SwK: i do not gaurantee they are correct. |
| 13:56.41 | hmmhesays | I need an ip kvm switch that allows console access so I can get to my cisco's and quintums |
| 13:56.44 | ccvp | i know this makes you all warm & cozy inside, but a company just paid a conulting company $121,000 to deploy call manaager, I had a hand in it. when they could of done it for 99% cheaper with an asterisk-based solution |
| 13:56.46 | ccvp | lulz :) |
| 13:56.59 | SwK | hmmhesays, when it comes to IP KVMs its cheaper to get them built in with dell hp or sun hardware... out of those 3 only Dell Charges extra for the addon |
| 13:57.10 | Katty | ccvp: cheers. congrats (= |
| 13:57.31 | SwK | hmmhesays, dont all the ciscos and quintums have serial ports on them? |
| 13:57.41 | Katty | ccvp: i hope you saw some of that profit. |
| 13:57.56 | ccvp | equipment purchased was around 35,000 |
| 13:57.57 | hmmhesays | The do I need remote access to said ports |
| 13:58.02 | ccvp | rest was labor |
| 13:58.19 | *** join/#asterisk LemensTS (i=CustomGT@70.238.191.87) |
| 13:58.46 | Katty | ccvp: and you got a nice chunk, not just by the hour? |
| 13:59.01 | LemensTS | If i have a TDM400 card and put in a TE122P, how do I know what channels to set? (i assume TDM is 1-4, then the T1 would be 5-28) |
| 13:59.34 | SteveTotaro | depends what order you load the modules |
| 13:59.41 | Katty | morning SteveTotaro (= |
| 13:59.54 | SteveTotaro | morning |
| 14:00.10 | SteveTotaro | what a difference a day makes, huh? |
| 14:01.05 | zeeesh | i m trying to install asterisk-addons but still unable to find "res_config_mysql.so"? |
| 14:01.30 | hmmhesays | Swk: yeah I know, these servers are already in place though. I wonder if there is a pci addon you can buy somewhere |
| 14:01.40 | mvanbaak | zeeesh: try 'make menuconfig' |
| 14:01.51 | mvanbaak | there you can see if res_config_mysql is going to be installed |
| 14:01.58 | SwK | hmmhesays, what kinda servers are they? |
| 14:02.11 | mvanbaak | and you can also see why not if it's not going to be installed |
| 14:02.22 | hmmhesays | Couple older visionman servers |
| 14:02.41 | SwK | i dunno |
| 14:02.43 | Lsodi | I have following notices on asterisk: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1, line is E1, zaptel.conf, interrupts zapata.conf are here http://pastebin.com/d30efd454 , what might be causing problem IRQ or wrong timing? |
| 14:02.51 | hmmhesays | as for the cisco and quintums, i'd like something standalone that I can just access the console ports with |
| 14:04.10 | SteveTotaro | quintum rocks |
| 14:04.11 | SwK | hmmhesays, http://www.data-connect.com/Patton_2120.htm |
| 14:04.28 | dandre | I have these log from a tcpdump: |
| 14:04.28 | dandre | 14:42:55.361637 provider_ip.5060 > nat_routeur-ip.63681: udp 484 |
| 14:04.28 | dandre | 14:42:55.361745 nat_routeur-ip > provider_ip: icmp: 82.236.48.30 udp port 63681 unreachable [tos 0xc0] |
| 14:04.28 | dandre | How could I fix this to receive inbound call from my sip provider? |
| 14:08.26 | *** join/#asterisk TheH (n=tommy@86.43.150.138) |
| 14:08.32 | TheH | hi |
| 14:10.16 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 14:10.20 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 14:10.52 | TheH | anyone noticed the timezone of the grandstreams 2000 is wrong (for europe) and even changing the ntp pool will not change a thing.. |
| 14:11.45 | mchou | TheH: ntp pool has nothing to do with TZ |
| 14:12.09 | mchou | TheH: think about it for 1 sec and you'd understand |
| 14:12.20 | TheH | yeha ok ok |
| 14:12.25 | TheH | BUT |
| 14:12.31 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:12.40 | TheH | there is still something wrong with those gxp2000's |
| 14:12.46 | mchou | lol |
| 14:13.02 | mchou | now that's insightful :) |
| 14:13.05 | potsboy | oh can i do it pleeease |
| 14:13.09 | potsboy | ~gs |
| 14:13.09 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:13.15 | potsboy | :) |
| 14:13.19 | TheH | true true |
| 14:13.31 | TheH | i got a few aalstra's here as well , not really happy with them either |
| 14:13.42 | Lsodi | dandre: can you register with your provider? unregister and register again, try to make call to sip account |
| 14:13.56 | TheH | although i love the xml progam features |
| 14:14.15 | angryuser | aastra, hate how they manage rensfer&attended transfer |
| 14:14.21 | angryuser | transfer* |
| 14:14.33 | TheH | ah yeah i know, i got some users complaining bout that crap. |
| 14:14.49 | TheH | also after a restart it gives you a nice pimp error #500 msg |
| 14:15.02 | TheH | cause some counters get messed up |
| 14:15.21 | angryuser | that's why i prefer snom |
| 14:15.56 | *** join/#asterisk Chris-NB (n=chris@213162066153.public.t-mobile.at) |
| 14:15.57 | angryuser | everyone is happy&no complaining |
| 14:16.05 | TheH | when i started in this comp. they already had 200 gxp200's |
| 14:16.06 | TheH | 2000 |
| 14:16.12 | dandre | Lsodi: Yes I can register and I can make call with this sip account |
| 14:16.45 | angryuser | 200? that's a shame ;) |
| 14:16.50 | TheH | i do have a nice failover setup now with my 4 asterisk boxes in 4 counties and the gxp's using dns srv (in broadsoft mode) to failover to each other |
| 14:16.55 | *** join/#asterisk JoseBravo (n=jbravo@190.156.225.15) |
| 14:17.25 | potsboy | TheH: doesnt help with the phones you have |
| 14:18.10 | angryuser | i have installed spa8000 this morning, at least it works fine |
| 14:18.14 | TheH | pots: it does it does , i tested it , the "normal" dns serv does not work but when you use the special feature: broadsoft the dns srv works |
| 14:18.38 | potsboy | fair enough |
| 14:18.48 | angryuser | TheH : extrernal ip's fixed or dynamic ? |
| 14:19.04 | TheH | pots: in normal mode the gxp would fail over on the 2nd server but it would not fall back if the 1st server would come back up , in broadsoft mode it does |
| 14:19.30 | TheH | all fixed ip, in a big cloud |
| 14:19.34 | potsboy | good to know, not that i intend on using grandstram |
| 14:19.41 | JoseBravo | I have a TDM400P connected to my asterisk, but it don't detect hang up. Im from Colombia, and I don't know what tones can I use. Is posible to record some tones and try to generate the codes (Frequency and Durarions)? |
| 14:20.01 | JoseBravo | *durations |
| 14:21.40 | potsboy | JoseBravo: yes it is, check the contribution by steve davies.. does exactly what you require... i am not from that part of the world and cant imagine how is doesnt work with ground or loop |
| 14:22.21 | TheH | anyone played with the n95 yet ? using wifi to connect to the asterisk box ? it works sweet only the n95 seems to not re-register the sip when it lots the wifi connection... only a small change for example in the profile name of the sip would force it to re-register |
| 14:25.44 | tzafrir | JoseBravo, IIRC you can get some information about that from the debug output |
| 14:26.04 | tzafrir | but are you sure you don't have polarity for hangup? |
| 14:26.07 | Mw3 | off: do you know some kind of device which delays to power on of the devices by 10, 20, 30 seconds after a power outage? |
| 14:27.08 | JoseBravo | tzafrir well, I have the hanguponpolarityswitch=yes in my zapata.conf but It dons't work. |
| 14:28.16 | potsboy | JoseBravo: we have the same issue here in south africa, the alcatel exchange dont produce nothin.. we use steves patch commited into 1.2 a while back |
| 14:29.48 | mpwizard | Hi, is it possible to use realtime integration of voicemail account and store messages as files? Using Asterisk 1.4 with Comedian Voicemail. |
| 14:30.04 | Lsodi | dandre: can you recive calls when you have just registered, try to call to sip account immediately when you have registered, and are you using software phone or hardware phone? |
| 14:30.51 | *** join/#asterisk ph0ne (n=ph0ne@206-248-164-27.dsl.teksavvy.com) |
| 14:31.19 | JoseBravo | postboy how works the patch? where can I found it? |
| 14:31.34 | jblack | huh. looks like * got confused here for the first time. |
| 14:31.38 | *** join/#asterisk adjohn (n=adjohn@p2103-ipad512marunouchi.tokyo.ocn.ne.jp) |
| 14:31.59 | *** join/#asterisk nighty^ (n=nighty@p2071-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 14:32.25 | JoseBravo | potsboy if the telco dons't send nothing, how can we determine the hang up? |
| 14:33.23 | potsboy | its in zaptel, heres the link http://www.connection-telecom.com/development.html |
| 14:33.38 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 14:34.19 | *** join/#asterisk flo (n=flo@p54969D47.dip0.t-ipconnect.de) |
| 14:34.23 | *** part/#asterisk flo (n=flo@p54969D47.dip0.t-ipconnect.de) |
| 14:34.48 | potsboy | it trains on the busy pattern i.e 500 ms silence 1500 non silnce |
| 14:35.06 | lirakis | this is a radom question... |
| 14:35.25 | lirakis | does anyone know if granstream GXP series phones handle sip/3XX redirect messages? |
| 14:35.37 | JoseBravo | potsboy So, are you still using asterisk 1.2? |
| 14:35.43 | lirakis | .. or for that matter... if any handset does? |
| 14:35.59 | potsboy | no it was first commited into the asterisk trunk at 1.2 |
| 14:36.12 | potsboy | i.e. zaptel 1.2.X |
| 14:36.16 | sysreq | mpwizard: the messages are still stored as files (in the spool directory) if you store your voicemail configurations in a database using realtime. |
| 14:36.23 | LemensTS | SteveTotaro: how do i specify the order the modules load? |
| 14:36.43 | SteveTotaro | when you modprobe them |
| 14:36.46 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:36.50 | JoseBravo | potsboy do you know a solution for 1.4.* ? |
| 14:36.56 | mpwizard | sysreq: ok. ty. |
| 14:37.07 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 14:37.22 | sysreq | mpwizard: np |
| 14:37.34 | potsboy | JoseBravo: its is _in_ zaptel 1.2.X and up.... i.e. official asterisk release.. just into .conf what i pasted in the link |
| 14:37.39 | SteveTotaro | modprobe wctdm, modprobe qozap |
| 14:37.55 | potsboy | JoseBravo: i.e. no patching |
| 14:38.27 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:39.02 | JoseBravo | potsboy can you listen my tone in www.colombiahosting.com.co/busy.wav and see if its the same? |
| 14:41.32 | *** join/#asterisk johndo (n=hef@castor.tavros.net) |
| 14:41.37 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:41.37 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:43.10 | potsboy | JoseBravo: use any audio program the can display the spectrum.. its will be easy to see.. i used audacity |
| 14:43.20 | jblack | Hmm. It seems * went a little squirrely on me yesterday without me knowing it. Is anything other than /var/log/asterisk/event_log that I can look at to figure out what happened? |
| 14:43.47 | jblack | pardon, s/event_log/messages |
| 14:44.37 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:45.32 | *** join/#asterisk steliosk (n=Stelios@79.131.65.61) |
| 14:47.08 | johndo | in asterisk 1.2.x I used VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) for a phone to get into its own voicemail |
| 14:47.32 | johndo | since ${CALLERIDNUM} doesn't work in 1.4.x, how would I accomplish the same thing |
| 14:47.35 | johndo | ? |
| 14:47.38 | Qwell | ${CALLERID(number)} |
| 14:47.57 | johndo | thanks |
| 14:48.10 | clyrrad | Any idea why even when using NoCDR() you still see the 's' extenstion CDR's and NO ANSWER() as disposition? |
| 14:48.35 | clyrrad | I want so that the CDR's are not going to record extension to extenstion calls, but NoCDR() is not working, if the extension does not answer a CDR is still saved for this |
| 14:48.44 | clyrrad | any idea how I can resolve this? |
| 14:49.16 | *** join/#asterisk itzbipin (n=bipin@122.171.12.24) |
| 14:49.44 | *** part/#asterisk UVSoft (n=jnk467@80.254.48.215) |
| 14:49.56 | potsboy | JoseBravo: before you go on a wild goose chase try kewl start, busy patterning should be a last resort |
| 14:50.11 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-206-182.bltmmd.east.verizon.net) |
| 14:50.53 | itzbipin | any one had done conferencing |
| 14:51.34 | itzbipin | how can we invite a friend with it |
| 14:51.43 | clyrrad | Is there something inherient that if you execute Dial inside your Macro to an extension that a CDR "must" be written if the extension does not answer? I am using NoCDR, but still a CDR is being written, cant figure out why.... any idea guys? |
| 14:52.25 | draygon-w | has anyone here ever intergrated nagios and asterisk? |
| 14:52.43 | *** join/#asterisk angryuser (n=nononon@df01t2-195-36-136-82.d4.club-internet.fr) |
| 14:53.43 | *** join/#asterisk amalgamated (n=krufty_t@gnu/amalgamated) |
| 14:54.09 | amalgamated | hi! when i run "iax2 show peers" from console, each host has either an (S) or a (D) after it -- what do these mean? |
| 14:55.09 | JunK-Y | amalgamated: static, dynamic |
| 14:56.04 | amalgamated | static or dynamic what? |
| 14:57.03 | Qwell | peers |
| 14:57.08 | draygon-w | Hi. Im currently using asterisk -rx "originate SIP/14072222222@callwithus-trunk extension s@context" command which calls the number successfully,however i need to know what i need to add to this command to PLAY a .wav file once the user picks up.. can someone help me? |
| 14:58.00 | draygon-w | TheH, was that for me? |
| 14:58.25 | TheH | drayon: no was just my msg :) sending to the room that i am gone |
| 14:58.32 | draygon-w | ah, heh ok |
| 14:58.37 | draygon-w | do you have any idea how to do what i am asking? |
| 14:58.42 | TheH | yeah |
| 14:58.46 | TheH | its the same |
| 14:58.49 | draygon-w | sweet! |
| 14:58.51 | TheH | ill give you a example |
| 14:58.52 | draygon-w | what'd you mean? |
| 14:58.54 | JunK-Y | Qwell: did you read my privmsg? |
| 14:59.01 | draygon-w | can you tie it in to my current command so it would make it easy for me please |
| 14:59.06 | dandre | Lsodi: yes just after registering, I can make an inbound call |
| 15:00.14 | TheH | 1. Users answers the phone 2. Playback(filename) so you need to convert your wav file to a asterisk readable format and then after pickup use playback or Background(filename) |
| 15:00.17 | Lsodi | dandre: you have nat problem then, are you using soft phone or hardware phone? |
| 15:00.23 | badcfe | if i do Transfer on a sip-chan before answering it then it just leads to a 302 Redirect huh? |
| 15:00.44 | badcfe | (i hope it will ...) |
| 15:01.06 | draygon-w | just pm'd you real quick theh, i hope you don't mind its too confusing in this channel :/ |
| 15:01.17 | dandre | My asterisk is behind a nat and my phones are on local net |
| 15:01.38 | *** join/#asterisk AlienPenguin (n=Miranda@213.188.207.153) |
| 15:01.48 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:02.08 | AlienPenguin | hi all, is there a way to disbale globally the call pickup feature? (*8) and not on a per user basis? |
| 15:04.00 | waKKu | see features.conf ? |
| 15:04.14 | *** join/#asterisk L|NUX (i=L_NUX@unaffiliated/lnux/x-10290) |
| 15:04.28 | AlienPenguin | waKKu : if i commet it out it still aloows picking up with *8 i want to disable it |
| 15:04.34 | waKKu | (and I'm trying to two days get it working with iax ;/) |
| 15:05.17 | *** join/#asterisk ManxPower (n=manxpowe@127.sub-75-202-166.myvzw.com) |
| 15:05.18 | waKKu | hm.. you can set it for a impractical code... or, unload features modules |
| 15:05.32 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:05.55 | mpwizard | sysreq: Hmmm... It seems that Asterisk want to fetch audiofile from database anyway; app_voicemail.c:1125 retrieve_file: Failed to obtain database object for 'asterisk'! |
| 15:05.55 | *** join/#asterisk thrills3k3r (n=jcrum@66-193-179-217.static.twtelecom.net) |
| 15:06.03 | AlienPenguin | waKKu: uhmmm if i unload the feature module i loose all the feature don't i? |
| 15:06.11 | waKKu | AlienPenguin yeah.. |
| 15:06.17 | AlienPenguin | i'd like just to disable some of them |
| 15:06.31 | jameswf | just pull out a yard stick and if a user hits *8 slap em on the hands and scream no touchie |
| 15:07.12 | ZPertee | anyone know how I can do an automated find and replace? I have a bunch of .call files that go to the same place. For example I have dean_one.call points to context get_dean_1. Currently dean_two.call also points there but I want it to go to get_dean_2. possible to automate it with batch script or not? |
| 15:07.19 | AlienPenguin | jameswf: now... that is a sensible suggestion! :-) |
| 15:07.43 | *** part/#asterisk amalgamated (n=krufty_t@gnu/amalgamated) |
| 15:08.01 | ManxPower | or just remove them from features.conf or remove T/t |
| 15:08.15 | ManxPower | ZPertee: ^KF |
| 15:08.20 | jameswf | if you cant babysit you may be able to link *8 to a shock collar |
| 15:08.32 | *** join/#asterisk al123 (n=alben@190.42.192.157) |
| 15:08.33 | AlienPenguin | ManxPower: removing from features.conf does not work |
| 15:08.52 | ManxPower | AlienPenguin: removing t/T and/or w/W will work |
| 15:08.52 | Qwell | jameswf: link it to the HR database |
| 15:08.53 | AlienPenguin | jameswf: the shock collar would cost too much for all the users |
| 15:09.28 | jameswf | heh still considers putting together shock_collar.agi |
| 15:09.30 | jameswf | :) |
| 15:09.36 | Qwell | UPDATE users SET salary = salary - (salary / 20) WHERE userid = ${EXTEN} |
| 15:09.43 | ManxPower | AlienPenguin: depending on the phone you can just disable that on the phone. I assume you are trying to NOT to call pickup |
| 15:10.36 | AlienPenguin | ManxPower : yeah i am trying NOT to call pick up but i'd like to do it at pbx level instead of configuring all the phones |
| 15:10.57 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 15:11.00 | ManxPower | AlienPenguin: remove callgroup= and pickupgroup= from sip.conf |
| 15:11.20 | ccvp | wtf |
| 15:11.23 | ccvp | firefox 3B4 blows |
| 15:11.24 | ManxPower | Why do you have them in there in the first place? |
| 15:11.29 | jblack | you can do better than that, qwell. |
| 15:11.46 | Qwell | ccvp: got a contact at Cisco that I can yell at for making the firmware downloads so stupid? |
| 15:11.47 | AlienPenguin | ManxPower: : i know, but that would have been my last resort... |
| 15:11.57 | AlienPenguin | cos i am working on yet another configuring interface |
| 15:12.07 | AlienPenguin | and work with text files not realtime asterisk |
| 15:12.09 | jblack | update users SET salary = salary*.95 where userid = ${EXTEN} |
| 15:12.17 | Qwell | pfft |
| 15:12.32 | AlienPenguin | hance to disable the functionality globally i would have prefereed to work on features.conf instead of modifying the sip.conf |
| 15:12.36 | Qwell | okay, fine, you win |
| 15:12.37 | jblack | sorry. Can't help it. We do a lot of math here. |
| 15:12.42 | *** join/#asterisk kannan (n=kann@123.201.60.116) |
| 15:13.05 | Qwell | jblack: in that case, better make it 94% |
| 15:13.10 | ManxPower | AlienPenguin: why would that be your last resort??? |
| 15:13.18 | waKKu | ppl... talking about pickups ... could someone help me with pickups with IAX ? http://pastebin.ca/936404 |
| 15:13.34 | ManxPower | waKKu: IAX does not support call pickups. |
| 15:13.42 | waKKu | ManxPower 1.4 ? |
| 15:13.45 | jameswf | http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 15:13.47 | AlienPenguin | ManxPower: cos editing the sip conf and commenting it out on a per user basis it is more lengthy :) |
| 15:13.57 | ManxPower | AlienPenguin: It sucks to be you. |
| 15:14.13 | ManxPower | waKKu: I believe so. |
| 15:14.22 | jameswf | you know what really sucks.... those dyson vacume cleaners |
| 15:14.43 | waKKu | ManxPower http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups here says that supoport |
| 15:14.51 | ManxPower | waKKu: It lies. |
| 15:14.55 | ZaVoid | anyone know of any sip ata's that work with openvpn? |
| 15:14.59 | ManxPower | much of the wiki lies. |
| 15:15.05 | ManxPower | ZaVoid: any of them |
| 15:15.09 | waKKu | omfg! i'll kill that wiki :D |
| 15:15.11 | waKKu | :/ |
| 15:15.15 | Qwell | I think he means openvpn directly on the ata |
| 15:15.18 | Qwell | which would be a bit silly |
| 15:15.23 | ManxPower | Qwell: it is silly |
| 15:15.33 | aydiosmio | would adding a context to a SIP peer entry prevent asterisk from matching that peer on calls that do not match that context? |
| 15:15.34 | ManxPower | waKKu: the wiki is one of the last places to look |
| 15:15.56 | ManxPower | aydiosmio: SIP does not provide the context as part of the auth, only IAX |
| 15:16.00 | waKKu | :/ |
| 15:16.03 | waKKu | damnit |
| 15:16.20 | ZaVoid | i haven |
| 15:16.21 | ZaVoid | t seen any manx |
| 15:16.48 | [TK]D-Fender | ZaVoid: You need to read his answer in context... |
| 15:17.17 | ManxPower | ZaVoid: ATAs don't care how the packets get from point A to point B. |
| 15:17.43 | ManxPower | Now if you want to do something utterly stupid like try to make the ATA a VPN endpoint - Well just be lucky no products out there support that. |
| 15:17.56 | jblack | 2 people in the house, 6 extensions, and 226 lines of dialplan..... |
| 15:18.06 | ManxPower | There are some linksys boxes that combine a router with an ATA, maybe one of those will do what you want to do. |
| 15:18.06 | aydiosmio | well, maybe someone can help me then. I have two servers exchanging calls, and asterisk needs to match an "outbound" peer with g729 only, and an "inbound" peer with g711 only, is there a way to prevent asterisk from matching the call to the "outbound" peer when calls come in, even thoguh the two peers have the same ip address? |
| 15:18.25 | ZaVoid | le me rephrease that then.. yes ManxPower i want an ata that has openvpn support as an endpoint built in to |
| 15:18.34 | *** join/#asterisk ber___ (i=brad@neu.cow.org) |
| 15:18.34 | ManxPower | aydiosmio: Asterisk really isn't designed for that, but just use two different peers |
| 15:18.34 | ZaVoid | and for certain markets its not stupid |
| 15:18.39 | jblack | Clearly, * is an addictive drug. * devs, be on notice that I'm filing a federal complaint against you guys for distributing crack. |
| 15:18.49 | ber___ | does software fax with asterisk work yet or is it still too buggy? |
| 15:19.01 | ber___ | last time i tried it, it was 90-95% reliable |
| 15:19.08 | ManxPower | ZaVoid: *nod* And we say that is stupid and you should not do that, but if you did actually want to do that I doubt you'll find any. |
| 15:19.18 | aydiosmio | ManxPower: I am using two different peers, but they have the same ip address, so the wrong peer is being matched and when the codec mismatches, the call is rejected |
| 15:19.22 | ZaVoid | stupid why? |
| 15:19.24 | ManxPower | ber___: WHICH fax software? Asterisk doesn't include any. |
| 15:19.31 | jblack | damn. I forgot the fax line. Make that 7 extensions |
| 15:19.32 | jameswf | in the winter I have a friend call silent witness and report drugs hidden in the fire wood... never have to chop wood :) |
| 15:19.32 | ber___ | asterfax could be one |
| 15:19.37 | ber___ | they all seem to use the spandsp |
| 15:19.51 | jameswf | ~spandsp |
| 15:19.52 | jbot | from memory, spandsp is cool : http://www.soft-switch.org/installing-spandsp.html, or a softmodem that comes with app_rxfax and app_txfax for asterisk |
| 15:19.52 | ber___ | all i want to know is there any VOIP based fax stuff which is working 99.99% reliable |
| 15:20.03 | ZaVoid | is as stupid as building a vpn endpoint connection into a soft client? |
| 15:20.06 | ManxPower | ber___: there is none |
| 15:20.11 | ber___ | ok thats what i thought |
| 15:20.19 | jameswf | jbot: spandsp is also the shizzznit |
| 15:20.20 | jbot | okay, jameswf |
| 15:20.20 | ber___ | but just checking to see if anything new came up :) |
| 15:20.21 | Qwell | ZaVoid: pretty much, yeah |
| 15:20.21 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.227.137) |
| 15:20.22 | jblack | ber___: I'm using a receive_fax agi, that I think works... however, I don't get any faxes, so.... |
| 15:20.27 | ber___ | hahah |
| 15:20.34 | *** join/#asterisk mitcheloc (n=mitchelo@216.207.245.1) |
| 15:20.40 | ManxPower | ZaVoid: Yes. |
| 15:20.50 | jblack | You're welcome to test faxing against me if you like. |
| 15:20.55 | mpwizard | sysreq: Found it... I had compiled in support for odbcstorage into app_voicemail. |
| 15:20.56 | ZaVoid | based on what? |
| 15:21.03 | ber___ | no its ok, i will just use hardware fax boards |
| 15:21.18 | jblack | That's probably smart. |
| 15:21.23 | ber___ | yeah, time is money :) |
| 15:21.25 | ZaVoid | ever consider that some markets/areas of the world have sip blocked/filted and if they don't want to use iax/h323 the only toher method that is effective is vpn.... |
| 15:21.34 | ManxPower | ber___: those won't get you %99.99 either. |
| 15:21.44 | ber___ | thats what efax uses |
| 15:21.48 | jblack | With the testing I did, faxing was fine... but receive_fax does have a bad reputation. |
| 15:21.50 | ManxPower | that's like 1 bad fax out of 1000 |
| 15:21.53 | ber___ | so i am happy with efax reliability |
| 15:21.57 | Qwell | ManxPower: haha |
| 15:21.59 | ManxPower | you get that with two pots lines and two fax machines |
| 15:22.01 | ber___ | spandsp is about 90% reliable |
| 15:22.04 | jameswf | mmm VPN and Out burger |
| 15:22.05 | draygon-w | Hi. Im currently using asterisk -rx "originate SIP/14072222222@callwithus-trunk extension s@context" command which calls the number successfully,however i need to know what i need to add to this command to PLAY a .wav file once the user picks up.. can someone help me? http://pastebin.com/m8d4a892 is my extensions.conf |
| 15:22.09 | Qwell | you won't get *that* with two fax machines directly connected |
| 15:22.12 | ManxPower | You DON'T get that with two pots lines and two fax machines |
| 15:22.15 | johndo | so our upstream provider just went offline |
| 15:22.33 | Qwell | ber___: fax is about 90% reliable... |
| 15:22.37 | johndo | anyone know of a good provider that will let me set callerID in a bery short period of time? |
| 15:22.42 | ber___ | not in my experience |
| 15:22.47 | Qwell | (okay, maybe 99.. but definitely not more) |
| 15:22.59 | jblack | johndo: Huh? |
| 15:23.05 | johndo | *very |
| 15:23.07 | itd | ~itsplist-us |
| 15:23.07 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 15:23.11 | ber___ | i dont know i probably have faxed 1k documents i dont remember many transmission errors |
| 15:23.14 | ber___ | its one thing to be busy |
| 15:23.20 | Qwell | many? |
| 15:23.20 | ber___ | or no answer |
| 15:23.29 | ber___ | like 1 in 1000 |
| 15:23.29 | ManxPower | Qwell: We FINALLY trained our users to stop complaining to IT when a fax does not go thru. The fax machines are connected directly to telco POTs lines. Nothing IT can do. |
| 15:23.30 | Qwell | again - 99.99% is exactly one out of one thousand. |
| 15:23.32 | ber___ | definitely good enough for business |
| 15:23.59 | draygon-w | when you set your caller id, lets say someone has t-mobile and you use a t-movile # do they get billed for it? |
| 15:24.16 | ber___ | draygon that is authenticated by the cell tower |
| 15:24.18 | itd | I dont understand why people still use fax machines wtf is wrong with a scanner and email |
| 15:24.22 | Qwell | draygon-w: unless t-mobile has fixed that bug |
| 15:24.25 | ber___ | you cant scam it like you are trying |
| 15:24.32 | draygon-w | Qwell which bug? |
| 15:24.39 | Qwell | the bug you're asking about |
| 15:24.43 | draygon-w | ah |
| 15:24.48 | draygon-w | so people were doing that I assume? lol |
| 15:24.52 | Qwell | mobile-to-mobile minutes used to be based on callerid.. I don't know if they still are |
| 15:24.54 | ber___ | yeah people use fax because it takes less time |
| 15:24.57 | Qwell | they probably are though |
| 15:25.03 | ber___ | simple stupid process |
| 15:25.08 | ber___ | paper is dumb but peopel are attached to it |
| 15:25.16 | ManxPower | itd: because most people are technical morons and lazy. It takes much more time to import a digital signature, after getting the document electronically than it does to sign the paper and stuff it into a machine. |
| 15:25.30 | ber___ | exactly manx |
| 15:25.48 | jblack | There's no way that one can avoid cell charges just by setting callerid. Large carriers are much smarter than that. |
| 15:25.58 | ManxPower | Why do people still use phones, it's 100 year old technology! |
| 15:25.59 | Qwell | jblack: you'd be surprised |
| 15:26.11 | Qwell | hell, t-mobile used to authenticate *voicemail* by callerid |
| 15:26.11 | jameswf | If you cater to morons well frankly its like feeding a cat... do it once they neer go away |
| 15:26.21 | ber___ | recurring billing s:) |
| 15:26.25 | ber___ | yum yum |
| 15:26.28 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 15:26.42 | jblack | Well, that's user data, which doesn't have any value to them... but billing? Cmon! |
| 15:26.42 | ManxPower | jblack: most carriers seem to allow you to call a cellphone with the cellphone number as the callerid and get directly into the user's mailbox without a password. |
| 15:26.43 | jameswf | lmao I accessed my brothers voicemail by simply setting my caller ID |
| 15:26.59 | Qwell | jblack: again - you'd be surprised |
| 15:27.07 | draygon-w | Hi. Im currently using asterisk -rx "originate SIP/14072222222@callwithus-trunk extension s@context" command which calls the number successfully,however i need to know what i need to add to this command to PLAY a .wav file once the user picks up.. can someone help me? http://pastebin.com/m8d4a892 is my extensions.conf |
| 15:27.10 | Qwell | it worked for a very long time, and I'd bet that it still works |
| 15:27.30 | ber___ | you know there is some russian voip carrier out there doing that now |
| 15:27.32 | ber___ | if its true :) |
| 15:27.33 | jblack | So... I wonder waht cell providers do if you set the callerid to the handset you're calling.... |
| 15:27.36 | jameswf | ~repeat |
| 15:27.37 | jbot | methinks repeat is probably If nobody answers your question, don't just repeat it. Spamming the channel and getting ignored/banned/silenced isn't going to get a faster answer; spamming is a function of time. More likely, nobody knows the answer (/msg apt ask), or you need to provide more information (/msg apt sicco)), or ask me about "rephrase", or ask smart ... |
| 15:27.43 | ber___ | 'grey routes to US cell phone providers' |
| 15:27.46 | Qwell | jblack: send you to their voicemail |
| 15:27.50 | ManxPower | jblack: We just TOLD you. |
| 15:27.53 | Qwell | or "your voicemail", as it were |
| 15:28.29 | jblack | I'm gonna have to test verizon wireless later today. |
| 15:28.49 | ManxPower | jbwhat is your cell number |
| 15:28.50 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.63) |
| 15:28.53 | Qwell | jblack: not sure about verizon, but I know for a fact it worked on t-mobile |
| 15:29.07 | ber___ | thats hilarious |
| 15:29.14 | ber___ | you would think they woudl authenticate the handset |
| 15:29.27 | ber___ | as teh originator of the call |
| 15:29.32 | jblack | ManxPower: I don't have a cell phone. I'll wait for my ex-wife to come over later, and dick with her system |
| 15:29.33 | Qwell | in fact, people once changed Mark Spencer's voicemail greeting... Allison recorded it live |
| 15:29.34 | ber___ | or block calls coming in from external carriers |
| 15:29.38 | jameswf | yeah my brother is on t mobile... some providers let you set "always ask for a password" |
| 15:29.39 | ber___ | with t-mobile numbers |
| 15:29.51 | Qwell | jameswf: that's the reason for it :) |
| 15:29.52 | ber___ | (external non-roaming_) |
| 15:30.19 | ber___ | btw, thanks for the answer on the fax stuff |
| 15:30.49 | jblack | Speaking of faxes, can someone fax me? I don't care what. Just wanna see that it still works. |
| 15:30.54 | KillerQueen_86 | ~thwack ber___ |
| 15:30.55 | jbot | ACTION beats ber___ on the knee with a UNIX Manual |
| 15:31.09 | jblack | I'm at home, so NSFW is fine |
| 15:31.35 | jblack | sure |
| 15:31.45 | jblack | 206-203-5386 |
| 15:34.48 | jameswf | actualy "The Bet" by anton Checkhov ( http://en.wikipedia.org/wiki/The_Bet_(short_story) ) is a good add to the book shelf |
| 15:35.02 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 15:35.13 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:38.04 | jameswf | I kinda liked oregon... 20 degrees outside you dont even have to get out and pump your gas :) |
| 15:40.01 | ZaVoid | nj is like that you cant pump your own gas.. takes for damn ever |
| 15:40.38 | SteveTotaro | link a cron to the HR database that checks for an entry in date terminated for your record, if not null then rm -rf * |
| 15:42.18 | aydiosmio | or, better yet, gzip+tar, mv /olddata, rm |
| 15:42.31 | aydiosmio | because... deletion is a last resort. |
| 15:43.00 | aydiosmio | never delete what you can archive |
| 15:43.14 | SteveTotaro | the whole point is deleting |
| 15:43.15 | Qwell | why archive, if you're doing it to be malicious? |
| 15:43.17 | *** join/#asterisk jdg (n=jdg@203.185.182.6) |
| 15:43.26 | SteveTotaro | unless you are going to ftp and then delete |
| 15:44.08 | *** join/#asterisk xezz (n=asdasd@87.203.212.82) |
| 15:44.12 | SteveTotaro | or setup some random extension system(service iptables stop) |
| 15:45.29 | SteveTotaro | qwell i call it insurance, not malicious ;) |
| 15:46.17 | *** join/#asterisk svenna (n=svenna@p548D06F6.dip0.t-ipconnect.de) |
| 15:46.57 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 15:47.08 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 15:48.17 | BCS-Satori | How do you set a password on a dial? For example, when the user enters extension 19 from the auto attendent i want it to ask for a password before going to the next step in the ext. |
| 15:49.07 | SteveTotaro | authenticate |
| 15:49.34 | BCS-Satori | thanks |
| 15:50.56 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:51.06 | *** join/#asterisk Victor_Yure (n=Victor_Y@unaffiliated/victoryure/x-837844) |
| 15:51.53 | BBHoss | anybody know anything like lumenvox that is free and worth looking at? |
| 15:52.24 | clyrrad | in cdr.conf with unanswered = no, why am I still getting CDR's recorded for calls that are not answered? I am also using NoCDR() in the dial plan but the CDR's are still beting recorded. Why is this happening? I have been searching google to no avail.,..... any help? |
| 15:52.58 | clyrrad | <PROTECTED> |
| 15:57.29 | badcfe | when doing Transfer leading to 302 Temporarily Moved on SIP channel then should i just forget about it or check something? will the 302 be retransmitted until ACK is coming even if i do Hangup on the channel immediately after doing Transfer ? |
| 15:58.08 | badcfe | (i hop retransmit on sip channel will be taken care of even if my dialplan execution is long ago finished huh) |
| 15:58.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:58.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:59.00 | nixguy | what happens when multiple sip clients try to auth to asteris with same username /pass |
| 15:59.19 | Qwell | auth, or register? |
| 15:59.22 | Qwell | those are completely different |
| 15:59.33 | Juggie | nixguy, the last one to register is the active registration |
| 15:59.38 | hmmhesays | nothing like a morning of walker texas ranger to start the day |
| 16:00.02 | nixguy | Juggie: ok but since i guess they try to reregister they will compete |
| 16:00.10 | nixguy | so it will be random what sip user gets the dial |
| 16:00.16 | Qwell | nixguy: exactly |
| 16:00.19 | Juggie | nixguy, yep |
| 16:00.28 | nixguy | ok not perfect so to speak |
| 16:00.32 | Qwell | and there's nothing that can be done to change that, except register with 2 different accounts |
| 16:00.32 | Juggie | no. |
| 16:00.38 | nixguy | was thinking about how a user who has a hard sip phone inthe office |
| 16:00.43 | nixguy | could use his extension when "mobile" |
| 16:00.50 | nixguy | ok |
| 16:00.55 | Juggie | he can, but do it with two registrations |
| 16:01.03 | Qwell | yeah, just dial both |
| 16:01.05 | nixguy | different numbers |
| 16:01.06 | Juggie | like if you had extension 4000 |
| 16:01.17 | Juggie | you cuold have sip reg, 4000 and 4000_mobile |
| 16:01.26 | Juggie | and dial both when you receive a call for 4000 |
| 16:01.41 | SteveTotaro | ~ron paul |
| 16:01.41 | jbot | Ron paul could kick chuck Noris; Arse |
| 16:01.49 | nixguy | true true |
| 16:02.01 | Juggie | i wish there was a better way w/ sip registrations too |
| 16:02.06 | Juggie | but, nat breaks the whole thing. |
| 16:02.17 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 16:02.38 | Juggie | you could say allow one registration per user per ip, but then nat breaks that model.. because what if you have dif users who want to use the same registration behind nat. |
| 16:03.40 | nixguy | nat wont be a real issue for me since this implementation is only going tobe used internally on the company |
| 16:03.50 | nixguy | users on laptops will have to use vpn anyway |
| 16:03.53 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:03.53 | Juggie | i thought about this problem before and sadly there is no ideal solution |
| 16:04.03 | Juggie | your best bet is just to make two sip profiles |
| 16:04.31 | nixguy | ok thnx for your time and help |
| 16:05.14 | tzafrir | http://www.voiceroute.org/ - anybody tried this? |
| 16:06.14 | JoseBravo | tzafrir can you explain me "JoseBravo, IIRC you can get some information about that from the debug output" |
| 16:06.55 | clyrrad | Can anyone help me with my CDR question? |
| 16:07.59 | clyrrad | :/ |
| 16:08.06 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 16:08.07 | teknoprep | hey all |
| 16:08.45 | clyrrad | unanswered = yes and unanswered = no seems to have the exact same result...... I dont get it... |
| 16:12.18 | ManxPower | clyrrad: perhaps your docs don't match your version |
| 16:13.47 | clyrrad | ManxPower: hrm, I have checked on the net and the docs, I am using 1.4.18, and unanswered = no should "not" be recording calls that are not answered... |
| 16:14.18 | alrs | tzafrir: downloading |
| 16:14.49 | tzafrir | alrs, one thing to note: IIRC it installs a DHCP server by default |
| 16:15.07 | alrs | tzafrir: yikes, thank you |
| 16:15.36 | tzafrir | We expect every router to do that, but not a PC :-( |
| 16:15.56 | asteriskmonkey | can you do includes in realtime? |
| 16:17.18 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:22.25 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 16:22.56 | atis_work | asteriskmonkey: you can't |
| 16:23.50 | atis_work | asteriskmonkey: realtime is for automatically generated parts of dialplan, so you can just duplicate upon writing, or use goto |
| 16:25.47 | *** join/#asterisk VaNNi (n=VaNNi@216.70.165.200) |
| 16:28.24 | Kobaz | it's not necessarily for generated parts. it's for pulling dialplan parts out of a database |
| 16:28.55 | ManxPower | No, Realtime is for pulling CONFIG INFORMATION from a database. You can store SIP accounts, voicemail account, etc |
| 16:28.59 | drmessano | tzafrir: I think the idea of the "DHCPd on the PBX" came from needing to push tftp parameters to clients and not being to do that with a SOHO router.. Trixbox has a DHCP GUI too, as I understand. |
| 16:29.28 | *** join/#asterisk tld (n=terje@217-14-11-114-dhcp-osl.bbse.no) |
| 16:31.34 | jameswf | gui |
| 16:31.35 | jameswf | heh |
| 16:31.55 | *** join/#asterisk datachomper (n=russ@75.146.194.59) |
| 16:32.06 | JoseBravo | I need help to determine the frequency of my busy tone: its my tone: www.colombiahosting.com.co/busy.wav |
| 16:32.17 | jameswf | []D[][]V[][]D |
| 16:32.42 | datachomper | Was there a problem with the asterisk 1.2 dial application, where the L() macro doesn't play the sound files? It works correctly and hangs up the call but it's not playing any of the files I assigned to it. |
| 16:33.34 | *** join/#asterisk bantu (n=Miranda@p54A33E95.dip0.t-ipconnect.de) |
| 16:35.05 | *** join/#asterisk boster (n=boster@chi-pat.cashnetusa.com) |
| 16:35.07 | ManxPower | JoseBravo: why? |
| 16:35.58 | boster | I'm reading through the change log and I found a change/patch that might relate to a problem that I'm having. It says it references an issue in "BE-396" |
| 16:36.24 | ManxPower | boster: that is BUSINESS EDITION |
| 16:36.40 | boster | does anyone know where I can find this? It sounds like some sort of bug or trouble ticket number, but it's not the right format for bugs.digium |
| 16:36.43 | boster | ah |
| 16:37.00 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 16:37.07 | boster | thanks |
| 16:38.16 | *** join/#asterisk ederaam (n=ederaam@unaffiliated/ederaam) |
| 16:38.27 | *** part/#asterisk ederaam (n=ederaam@unaffiliated/ederaam) |
| 16:39.27 | JoseBravo | ManxPower My asterisk don't detect the hang up. I save the tone into a wav.. I need to know what frequency is for put it in indications.conf |
| 16:40.11 | ManxPower | JoseBravo: I don't believe indications.conf is used by the busy detection. |
| 16:40.24 | ManxPower | Unless you have evidence that says different |
| 16:40.53 | JoseBravo | ManxPower then zonedata. |
| 16:40.59 | JoseBravo | zonedata.c |
| 16:42.02 | ManxPower | JoseBravo: there are several places in Asterisk with that information, you should look at the callprogress detection code to see which one it uses |
| 16:42.08 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 16:42.31 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 16:43.07 | jameswf | JoseBravo: Have you tried calling your OEM to check for updates... |
| 16:44.15 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
| 16:44.34 | JoseBravo | jameswf my telco don't say nothing.... |
| 16:44.50 | jameswf | no the maker of your card not your teco |
| 16:45.19 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 16:45.35 | __freedom__lover | does anyone use asterisk 1.4 on freebsd? |
| 16:45.54 | __freedom__lover | i cannot use iax with trunk mode enabled.. |
| 16:46.46 | outtolunc | you need a timing source, which iirc *bsd doesn't have a zaptel driver |
| 16:46.58 | outtolunc | (but there might be one nowdays) |
| 16:47.40 | *** part/#asterisk boster (n=boster@chi-pat.cashnetusa.com) |
| 16:48.05 | JoseBravo | ManxPower callprogress=yes may help me? |
| 16:48.10 | __freedom__lover | outtolunc: yes, there's zaptel-bsd and i've installed it.. |
| 16:48.43 | __freedom__lover | in my cli it shows: Unable to support trunking on user 'teste' without zaptel timing |
| 16:48.55 | *** join/#asterisk greekguy8888 (n=alex@c-76-118-204-95.hsd1.ma.comcast.net) |
| 16:49.17 | mort_gib | question.. One of my users wants to have his calls transferred if he is in a call, but the Snom phones I use allow several incoming calls, any ideas as to how this is done?? |
| 16:49.17 | outtolunc | does user 'teste' have access to the zaptel device? |
| 16:49.42 | outtolunc | my guess would be nope <G> |
| 16:50.53 | ZPertee | I have a bunch of files dean_line_one, dean_line_two, bob_line_one, bob_line_two, etc. So what I want to be able to do is is make the name a variable and the the line number a variable. Then change each file accordingly. Here is what my file looks like http://pastebin.ca/939832. |
| 16:50.58 | outtolunc | that came out wrong, but you get the idea |
| 16:51.01 | *** part/#asterisk fedya (n=fedya@75.112.143.226) |
| 16:52.14 | __freedom__lover | outtolunc: how can see if my iax user 'teste' has permission for that? |
| 16:52.49 | outtolunc | thats the part that came out wrong |
| 16:53.00 | JoseBravo | ManxPower ? |
| 16:53.04 | outtolunc | your asterisk process, needs access to the zap device |
| 16:53.54 | *** join/#asterisk jbigB (n=jBigB@216.207.245.1) |
| 16:54.09 | __freedom__lover | hum... |
| 16:54.50 | outtolunc | unload your chan_zap and reload it |
| 16:54.57 | __freedom__lover | outtolunc: you are a genius, or i'm a fool |
| 16:54.58 | __freedom__lover | ahuahau |
| 16:55.00 | outtolunc | see what it 'says' |
| 16:56.46 | *** join/#asterisk hohum (n=dcorbe@68.26.92.34) |
| 16:57.54 | __freedom__lover | outtolunc: i'm using asterisk running as asterisk user |
| 16:58.28 | __freedom__lover | my zap devices are owned by root, and just it has permissions, then i changed the owner to asterisk |
| 16:59.00 | __freedom__lover | crw------- asterisk wheel timerXX |
| 16:59.11 | __freedom__lover | but no success with iax trunk :( |
| 16:59.43 | outtolunc | hold on, let me find the *bsd one |
| 17:00.00 | *** join/#asterisk af_ (n=getsmart@88-149-240-207.dynamic.ngi.it) |
| 17:00.54 | JayTee52 | hyperthreading on a dual Xeon system running Asterisk is a bad thing I'm thinking. Yes? |
| 17:01.11 | jameswf | ~fleamarket |
| 17:01.11 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
| 17:01.25 | jameswf | heh |
| 17:03.34 | *** join/#asterisk CVirus (n=GoD@196.205.192.125) |
| 17:03.39 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 17:04.05 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:04.52 | BCS-Satori | When using the monitor command is there a way to save the audio file as $CID_$DATE.wav ? insead of audio-0.wav |
| 17:05.18 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:05.48 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 17:07.16 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:08.26 | BBHoss | BCS-Satori, yes |
| 17:09.02 | BBHoss | exten => _X.,n,Set(CallFileName=${CALLERID(all)}--${STRFTIME(${EPOCH},GMT-6,%C%y%m%d%H%M)}) |
| 17:09.02 | BBHoss | exten => _X.,n,Monitor(wav,${CallFileName},mb) |
| 17:09.04 | BBHoss | is what I use |
| 17:09.21 | ice_croft | hi all |
| 17:09.53 | BCS-Satori | BBHoss; thank you ill try it out |
| 17:10.00 | BBHoss | its works in 1.4 |
| 17:14.17 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 17:15.14 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 17:15.22 | Sci_05 | afternoon all! |
| 17:17.53 | JoseBravo | How can I do to generate the busy tone that my telco are sending me? I have the sound into a file. But I don't know the frequency... its the sound: www.colombiahosting.com.co/busy.wav |
| 17:18.27 | *** part/#asterisk greekguy8888 (n=alex@c-76-118-204-95.hsd1.ma.comcast.net) |
| 17:18.40 | BCS-Satori | BBHoss: thanks a lot that works perfectly |
| 17:18.45 | BBHoss | np |
| 17:18.51 | angryuser | <JoseBravo> play with playtone(!frequency) |
| 17:19.02 | angryuser | for europe it is !440 |
| 17:19.36 | Sci_05 | anyone setup an audiocodes mp118 with asterisk before? I am having problems getting the caller ID CNAME and stutter tone working correctly |
| 17:20.30 | BCS-Satori | BBHoss: do you know if i remove the ,GMT-6, if it would default to system time? |
| 17:26.13 | *** join/#asterisk gego (n=gego@dyndsl-091-096-097-072.ewe-ip-backbone.de) |
| 17:31.17 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 17:34.28 | errr | BCS-Satori: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+strftime |
| 17:35.06 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:35.34 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-230-175-33.dsl.irvnca.pacbell.net) |
| 17:35.37 | UnixDog | mornign |
| 17:35.44 | alrs | Hey, UnixDog |
| 17:35.54 | UnixDog | well the 1.0.1 asterisk-now cd is fubared |
| 17:36.07 | UnixDog | it wont install to a usb drive |
| 17:36.27 | UnixDog | het alrs |
| 17:36.31 | UnixDog | hey even |
| 17:37.35 | Katty | ohai |
| 17:42.22 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 17:43.39 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:45.10 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:45.19 | jameswf | elastix 1.0 lookin pretty good |
| 17:46.20 | SteveTotaro | clownpbx rocks |
| 17:47.29 | SteveTotaro | elastix 1.0beta9? |
| 17:48.08 | SteveTotaro | why not just call it elastix .99999 or elastix 1.0rc9? |
| 17:48.13 | *** join/#asterisk greekguy8888 (n=alex@c-76-118-204-95.hsd1.ma.comcast.net) |
| 17:48.56 | greekguy8888 | "sharks sighted today'" |
| 17:49.00 | greekguy8888 | what does that mean? |
| 17:49.02 | greekguy8888 | :) |
| 17:49.12 | greekguy8888 | does someone live near the ocean? |
| 17:49.59 | harryv | in the console: Executing [99400505@incoming:1] DIal("Zap/1-1", "Zap/g2") in new stack |
| 17:50.12 | harryv | means that channel 1 on span 1 is calling group 2, right? |
| 17:50.20 | greekguy8888 | yep |
| 17:50.20 | waKKu | i live at florianopolis/brazil.. an island ;) |
| 17:50.50 | greekguy8888 | no really am i just naive or something? what does the sharks sighted thing mean |
| 17:51.00 | greekguy8888 | i've seen it say no sharks allowed, etc |
| 17:51.23 | greekguy8888 | what's a shark supposed to mean in this context? |
| 17:51.26 | greekguy8888 | hackers? |
| 17:51.27 | harryv | so |
| 17:51.37 | harryv | Zap/31-1 is proceeding passing to to Zap/1-1 |
| 17:51.39 | harryv | means, what? |
| 17:51.48 | harryv | seems opposite of what is happening, to me. |
| 17:52.16 | greekguy8888 | span 1 channel 31 is passing to spanm 1 channel 1 |
| 17:52.42 | harryv | but, that's not what happenede. span 1 channel 1 dialed g2, which is span2 starting from channel 32 |
| 17:52.55 | *** join/#asterisk oej (n=olle@114.62-97-206.bkkb.no) |
| 17:53.00 | greekguy8888 | i would check your zaptel config then |
| 17:53.03 | harryv | oh. wait. |
| 17:53.22 | harryv | span 1 is chan 1-31 .. why is it passing from chan 1 to 31 on same span!? |
| 17:53.41 | greekguy8888 | "i would check your zaptel config then" :) |
| 17:54.03 | SteveTotaro | ~shark |
| 17:54.04 | jbot | There are NO sharks in #Asterisk. We want to be perfectly clear on that. |
| 17:54.14 | harryv | ah yeah, i forgot my fine paste. |
| 17:54.26 | greekguy8888 | lol |
| 17:55.24 | SteveTotaro | druid looks pretty cool |
| 17:55.44 | SteveTotaro | opensource edition |
| 17:55.55 | SteveTotaro | www.voiceroute.org |
| 17:56.00 | harryv | http://trueaffection.net/paste/zapzap.txt |
| 17:56.19 | harryv | argh! wtf, my fault. s/31/32. |
| 17:56.31 | harryv | typo. and then i just looked at it here. and got confused. |
| 17:57.35 | SteveTotaro | ~stevetotaro |
| 17:57.36 | jbot | rumour has it, stevetotaro is an IRC nub |
| 17:58.13 | SteveTotaro | canceling my elastixs download and going for druid |
| 18:02.56 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
| 18:03.25 | kannan | heelo, can an AGI script inherit a variable defined as global in a dialplan extensions? |
| 18:04.04 | ZPertee | anyone know how to fix this http://pastebin.ca/939945 |
| 18:04.36 | ice_croft | ZPertee> wtf? |
| 18:04.52 | flush | yo what does this mean http://pastebin.com/m38284f2a |
| 18:05.08 | flush | when i call in, the CLI shows this but the phone plugged in the asterisk box wont ring |
| 18:05.11 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 18:05.26 | hmmhesays | I think I'm in love with fedora core 8 |
| 18:05.43 | ice_croft | flush> lines 3, 4 |
| 18:05.57 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:06.02 | ZPertee | ice_croft: I made a bunch of .call files by copying one file and now I am trying to automatically go in and fix each file based on the filename |
| 18:06.40 | flush | ice_croft == Starting Zap/3-1 at from-pstn,s,1 failed so falling back to exten 's' is why the phone doesnt ring ? |
| 18:07.07 | kannan | or should the variable be passed to the script as an argument only? |
| 18:07.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:07.21 | ice_croft | flush> well, to ring it must reach it, i think |
| 18:07.47 | ice_croft | flush> at least |
| 18:08.07 | flush | yes i understand.. how would i make it recognize where its supposed to go? |
| 18:08.15 | flush | its in extensions.conf ? |
| 18:08.15 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-ff7704d1aee3a39b) |
| 18:08.25 | harryv | when bridging a dial, Dial(Zap/32-2) will the callerid from the origin follow by default? |
| 18:08.46 | harryv | eh |
| 18:08.51 | harryv | the callerid for the called id. |
| 18:09.32 | harryv | so, if someone calls 4000 and i bridge that dial, will Zap/32-2 receive the call as destined for '4000' ? |
| 18:09.53 | *** join/#asterisk danilom (n=danilom@201.209.158.232) |
| 18:10.37 | danilom | hi, i have a p4 with 512 ram with a digium and 4 fxo, is this enought for 40 users and 10 calls at the same time? |
| 18:10.51 | ZPertee | new pastebin with corrections http://pastebin.ca/939962 |
| 18:10.57 | [TK]D-Fender | flush: You told your zapata.conf channels to use "from-pstn,s,1" which doesn't exist. Go fix your incoming context. |
| 18:11.11 | [TK]D-Fender | flush: Either where you are pointing zapata.conf to, or your dialplan. Your choice. |
| 18:11.28 | JoseBravo | The hangup detection works with the busy tones that are in zonedata.c? |
| 18:12.00 | flush | my zapata.conf file is "#include zapata-channels.conf" |
| 18:12.17 | alrs | has anyone put more than two Sangoma A108DX cards in one box? |
| 18:13.37 | flush | [TK]D-Fender ok i see "context=from-pstn" in zapata-channels.conf |
| 18:13.49 | flush | i create a new entry [from-pstn] in extensions.conf now |
| 18:14.08 | flush | now this is what i barely get.. how do i tell it to ring the phone when a call comes in.. |
| 18:14.19 | flush | sorry im a total newb just trying to make it work |
| 18:14.33 | hmmhesays | dial |
| 18:14.36 | hmmhesays | ~dial |
| 18:14.43 | hmmhesays | no? argh |
| 18:14.55 | flush | ?dial |
| 18:15.36 | BCS-Satori | when i run this command, exten = 18,n,Set(CallFileName=${CALLERID(num)}--${STRFTIME(${EPOCH},GMT-6,'%F--%H-%M-%S-%p')}) right now it makes the %H for hour 00 and %p AM. but minutes and seconds are correct. when i do system date it show the corerct time, why is it stating that the time is 00AM? |
| 18:15.52 | [TK]D-Fender | flush: what is "the phone" you are referring to? |
| 18:16.27 | flush | its an analog phone plugged on channel 2, which is green/fxs module |
| 18:18.12 | [TK]D-Fender | flush: "exten => s,1,Dial(Zap/2,30) |
| 18:20.13 | tzafrir | flush, does your zapata.conf have [channels] on its top? |
| 18:20.33 | [TK]D-Fender | tzafrir : that seems fine |
| 18:21.08 | *** join/#asterisk zapa (n=hzavala@206.132.198.6) |
| 18:21.10 | flush | tzafrir http://pastebin.com/m8a7162a |
| 18:21.14 | flush | [TK]D-Fender let me try |
| 18:22.53 | tzafrir | flush, so it does have [channels] . fine |
| 18:23.00 | flush | ahoy it rings |
| 18:23.16 | [TK]D-Fender | flush: Good |
| 18:23.22 | flush | [TK]D-Fender what does the "30" in Zap/2,30 stands for |
| 18:23.32 | [TK]D-Fender | flush: ring for 30 seconds |
| 18:23.43 | flush | after it hangs up ? |
| 18:23.44 | [TK]D-Fender | flush: Time to sit down and read the book.. |
| 18:23.46 | [TK]D-Fender | ~book |
| 18:23.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:24.01 | flush | i read oreilly book |
| 18:24.09 | flush | but it wasnt clear to me.. |
| 18:24.53 | flush | now i answer the phone i get dial tone.. how would i make it actually work ? that i can speak with whos calling |
| 18:25.16 | [TK]D-Fender | flush: I think you answered after the calling end has already hung up... |
| 18:25.45 | flush | hrm let me try again |
| 18:25.45 | [TK]D-Fender | flush: thats a timing issue with analog lines if thats the case |
| 18:25.50 | *** join/#asterisk rsc (n=robert@fedora/rsc) |
| 18:25.53 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 18:26.01 | FarrisG | Does anyone have a door phone setup using an FXO port? |
| 18:26.12 | rsc | Somebody around having knowledge about kerberos and/or app_voicemail (IMAP)? |
| 18:26.14 | FarrisG | I had one working beautifully, but a recent asterisk/zaptel upgrade killed it. None of the conf files (zapate/zaptel) have been changed, and it still rings through properly, but now neither end can hear the other. No audio at all. |
| 18:26.23 | flush | hrm [TK]D-Fender actually when i pick up phone on channel 2 it gives dial tone |
| 18:26.34 | [TK]D-Fender | flush: while its ringing? |
| 18:26.37 | Sci_05 | anyone setup an audiocodes mp118 with asterisk before? I am having problems getting the caller ID CNAME and stutter tone working correctly |
| 18:26.43 | flush | and i have another phone plugged with a splitter in the wall jack, so only the other phone wil lcontinue to ring (the one thats not plugged via asterisk) |
| 18:26.47 | flush | [TK]D-Fender affirmative |
| 18:26.58 | [TK]D-Fender | flush: No idea... |
| 18:27.00 | *** join/#asterisk kalevra (n=dallas@adsl-76-203-166-4.dsl.rcsntx.sbcglobal.net) |
| 18:27.46 | flush | cause my [from-pstn] context only has "exten => s,1,Dial(Zap/2,30)" |
| 18:27.57 | flush | this is enough? it should work when i pick up the phone? |
| 18:28.24 | flush | and when i pick up the phone and press a key, it gives me fast busy tone.. like im not able to place call thats all a dialplan issue ? |
| 18:28.51 | flush | i pick up the phone, CLI shows: |
| 18:29.03 | flush | <PROTECTED> |
| 18:29.03 | flush | <PROTECTED> |
| 18:29.05 | [TK]D-Fender | flush: Corrent. We just set it up to handle an INCOMING call on your FXO port. This has nothing to do with allowing your FXS port to dial anything. |
| 18:29.08 | rsc | WHICH library/code section causes this error? WARNING[16064] app_voicemail.c: IMAP Warning: Can't use Kerberos: invalid /authuser |
| 18:29.11 | [TK]D-Fender | flush: GET READING. |
| 18:29.20 | flush | HQ we copy |
| 18:29.53 | [TK]D-Fender | flush: You need to set a context for your FXS port and set up dialplan patterns for what you want that phone to be able to do. |
| 18:30.53 | flush | kk, and this book http://downloads.oreilly.com/books/9780596510480.pdf is pretty much complete, i should be able to receive and place calls after i read all the section about the dialplan you think ? |
| 18:31.13 | *** join/#asterisk DirtyD (n=DirtyD@ool-18bddaa0.dyn.optonline.net) |
| 18:31.16 | DirtyD | Howdy |
| 18:31.26 | DirtyD | any lumenvox users here? |
| 18:32.12 | kalevra | whats up dirty |
| 18:32.15 | [TK]D-Fender | flush: probably |
| 18:32.20 | kalevra | not sure what lumenvox is... |
| 18:33.24 | bkruse | lumenvox is voice recognition |
| 18:33.28 | bkruse | It rox |
| 18:43.10 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 18:49.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:51.44 | flush | yo |
| 18:51.52 | flush | when i type "dialplan reload" in the cli i get WARNING[3080]: pbx.c:4695 add_pri: Unable to register extension 's', priority 1 in 'from-pstn', already in use |
| 18:53.38 | *** join/#asterisk svenna (n=svenna@p548D06F6.dip0.t-ipconnect.de) |
| 18:53.54 | bkruse | flush: you have to of the same extensions |
| 18:54.10 | bkruse | eg: exten => 1,1,Answer() |
| 18:54.10 | bkruse | exten => 1,1,Playback("demo-congrats") |
| 18:57.23 | flush | ohhh, h4x |
| 18:57.53 | bkruse | flush: Totally :] |
| 18:57.58 | flush | i use asterisk 1.4 its good ? |
| 18:57.59 | *** join/#asterisk oddn (n=fufufu@0x55508034.adsl.cybercity.dk) |
| 18:58.32 | oddn | what is isdn transfer capability? can't find much info on google? and the according SetTransferCapability app ? |
| 19:01.34 | rsc | nobody here using app_voicemail? |
| 19:02.58 | FarrisG | how do you restart zap channels in 1.2 without restarting * ? |
| 19:07.32 | *** join/#asterisk vap0rtranz (n=jpittman@75.110.17.157) |
| 19:07.34 | ZPertee | if I copy a .call file into /var/spool/asterisk/ and it just sits in /var/spool/asterisk/tmp what does that mean? |
| 19:08.32 | BBHoss | you should be copying to /var/spool/asterisk/outgoing if i remember correctly |
| 19:08.33 | tzafrir | OT: someone asked me about iax client for Windows Mobile. I saw some references to ppciax but only dead links |
| 19:08.44 | tzafrir | any ideas where I can find ppciax? |
| 19:08.55 | BBHoss | ZPertee, also make sure that the permissions are right if asterisk is running non-root |
| 19:09.16 | tzafrir | FarrisG, generally: unload chan_zap.so ; load chan_zap.so |
| 19:10.55 | ZPertee | BBHoss: I am running Asterisk as root. here is my .call file http://pastebin.ca/940053 |
| 19:11.36 | BBHoss | ZPertee, are you copying it to /var/spool/asterisk/outgoing ? |
| 19:13.05 | ZPertee | BBHoss, I am using this command "mv /etc/asterisk/call_files/tmp /var/spool/asterisk/outgoing" |
| 19:13.24 | ZPertee | BBHoss: nm I am an idiot! |
| 19:13.27 | BBHoss | heh |
| 19:13.36 | BBHoss | its the simple things always |
| 19:13.49 | ZPertee | ain't that the truth |
| 19:14.51 | flush | yo |
| 19:14.56 | flush | whats wrong with this context http://pastebin.com/m156fb745 |
| 19:15.46 | BBHoss | yo flush check this fixed version out: http://pastebin.com/m3ff9ac4 |
| 19:16.21 | BBHoss | that wont really do anything unless you have a DID of 1 |
| 19:16.27 | bkruse | flush: Wow, I thought you understood asterisk |
| 19:16.34 | bkruse | 1= priority, you cannot have 3, number 1 steps |
| 19:17.19 | bkruse | flush: http://pastebin.com/m362f8874 |
| 19:17.24 | BBHoss | flush, what exactly do you want to do, dialout on a zap trunk and playback demo-congrats? |
| 19:18.47 | *** join/#asterisk oddn (n=fufufu@0x55508034.adsl.cybercity.dk) |
| 19:18.47 | bkruse | BBHoss: In either case, that post I did solves the problem |
| 19:18.53 | oddn | what is isdn transfer capability? can't find much info on google? and the according SetTransferCapability app ? |
| 19:19.00 | BBHoss | bkruse, true |
| 19:19.12 | bkruse | flush: You really should not call your box h4xt3r1sk....tad lame |
| 19:19.15 | bkruse | and I will hack it. :] |
| 19:23.07 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:27.45 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.144) |
| 19:27.51 | vap0rtranz | is there somthing more minimal than audacity for smashing two of Allison's recordings together? assuming my server has a working sound car for remixing ... |
| 19:29.48 | [TK]D-Fender | vap0rtranz: What for? And probably sox... |
| 19:32.47 | vap0rtranz | [TK]D-Fender: my peeps tell me that shhheeee slllooooooow. even with & concat there's a background() call, so i'm hoping smashing it all together will speed up things |
| 19:33.45 | [TK]D-Fender | vap0rtranz: Ok, well SOX is it... |
| 19:38.29 | *** join/#asterisk cleone (i=cleo@41.251.86.87) |
| 19:38.37 | *** part/#asterisk cleone (i=cleo@41.251.86.87) |
| 19:39.20 | *** join/#asterisk oej (n=olle@114.62-97-206.bkkb.no) |
| 19:43.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:43.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:44.33 | *** join/#asterisk yenno (n=yenno@92.104.83.58) |
| 19:45.56 | yenno | hi, how would you monitor your asterisk uptime? i'd like to automatically initiate a call every 5 minutes, what program would you use? |
| 19:46.14 | waKKu | cron ? |
| 19:46.29 | yenno | yes, but what program should i call.. like, sipp |
| 19:46.50 | yenno | a client that can authenticate with asterisk, call a number and check for the 200 ok |
| 19:46.53 | waKKu | just creata callfile and put it on /var/spool/asterisk/outgouing |
| 19:46.55 | waKKu | outgoing |
| 19:48.32 | yenno | ok thanks, never heard of that one |
| 19:49.23 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 19:57.49 | *** join/#asterisk Noise98 (n=moi@modemcable193.193-59-74.mc.videotron.ca) |
| 19:58.14 | Noise98 | hi , anyone know what the accronyme NOA mean on Calling NOA and Called NOA ? |
| 20:01.08 | *** part/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 20:06.05 | *** join/#asterisk robeph (n=robf@router.asteriasgi.com) |
| 20:07.05 | robeph | I'm trying to register a sip phone with a foreign asterisk server, I have everything correctly set up yet I get 403 forbidden from sip debug on attempted registration failing with the error "Bad Password" I know my secrets match up, and can find no clear cause of this, any ideas what could be at fault? |
| 20:07.27 | *** join/#asterisk svenna (n=svenna@p548D06F6.dip0.t-ipconnect.de) |
| 20:08.11 | iCEBrkr | Maybe the foreign astrisk box speaks French and your phone speaks english |
| 20:08.11 | jblack | robeph: You're not matching up to the account you think you are? |
| 20:08.53 | robeph | yes |
| 20:08.59 | robeph | both are showing the same |
| 20:09.16 | robeph | it is very odd... |
| 20:09.49 | robeph | this box is behind a cisco firewall which I have no access, though I assume since I'm seeing the registration attempt I'm not just passing 80/8080/22 traffic |
| 20:09.52 | robeph | so I don't think it's that |
| 20:10.05 | robeph | I really have no clue unless for some reason that cisco is mangling my packets |
| 20:10.20 | robeph | is there a way to have it display its recepit password |
| 20:10.24 | robeph | to show what it thinks I'm sending? |
| 20:10.35 | robeph | receipt* |
| 20:10.39 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 20:10.43 | iCEBrkr | set core verbose 9 |
| 20:10.50 | iCEBrkr | and then try to register |
| 20:10.52 | robeph | thanks lemme try that |
| 20:10.57 | iCEBrkr | it won't show the secret |
| 20:11.01 | robeph | k |
| 20:11.03 | iCEBrkr | but it'll spew more junk to the screen. |
| 20:11.39 | robeph | yeah, thats what it was already set at |
| 20:11.46 | robeph | also have sip debug ip <myip> goin |
| 20:12.46 | robeph | 403 forbidden bad auth on sip debug, and' Registration from '"xxxxxxx"<sip:xxxxxxxxx@xxxxxxxxx>;transport=UDP' failed for '24.214.206.xxx' - Wrong password' |
| 20:13.01 | robeph | i mean its hard to get 1234 wrong |
| 20:13.13 | ice_croft | whoa |
| 20:13.14 | ice_croft | :)) |
| 20:13.26 | robeph | ? |
| 20:13.39 | iCEBrkr | hehe |
| 20:14.00 | robeph | xxxxx'd out for security reasons... |
| 20:14.01 | ice_croft | funny |
| 20:14.03 | ice_croft | :)) |
| 20:14.20 | robeph | eh? |
| 20:14.27 | iCEBrkr | you have your extension in the [] or using regexten=? |
| 20:14.36 | robeph | [] |
| 20:15.01 | robeph | all standard... I mean, I've set it up the same way 100 times |
| 20:15.16 | robeph | this isn't an error on that part, something else is going on |
| 20:15.53 | robeph | the config there has all been checked and double checked... |
| 20:18.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 20:19.59 | robeph | actually I'm also getting a 401 unauth |
| 20:20.07 | robeph | host is set to dynamic |
| 20:20.27 | robeph | nm |
| 20:20.37 | robeph | thats normal.. |
| 20:21.03 | *** join/#asterisk nitram (i=nitram@superblob.com) [NETSPLIT VICTIM] |
| 20:21.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:21.21 | *** join/#asterisk harryv (n=fufufu@0x55508034.adsl.cybercity.dk) |
| 20:21.32 | harryv | hey, how to convert mp3 to a-law? |
| 20:21.35 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 20:22.48 | fujin | use sox |
| 20:23.12 | fujin | sox file.wav -t raw -r 8000 -c 1 -b -A file.alaw |
| 20:23.16 | iCEBrkr | robeph: I can't say I ever had this sort of problem *shrug* |
| 20:23.23 | robeph | iCEBrkr: me niether :p |
| 20:23.40 | robeph | fujin: thats not mp3 :p |
| 20:23.44 | fujin | s/wav/mp3/ |
| 20:23.50 | fujin | foad |
| 20:23.55 | robeph | mp3 -t raw? |
| 20:24.29 | clyrrad | Any of you guys know how to get access to the 411 database? Both to list numbers as well as do lookups via Asterisk? |
| 20:24.30 | fujin | 1.4 also has that convert console command anyway |
| 20:24.42 | fujin | file_convert |
| 20:24.46 | robeph | well you're telling it's raw format, but I don't think mp3 is raw |
| 20:24.48 | jblack | clyrrad: Try whitepages.com |
| 20:24.49 | fujin | file convert <infile> <outfile> |
| 20:24.54 | fujin | no, I'm telling it to put the alaw in raw format |
| 20:24.55 | fujin | gg. |
| 20:24.58 | fujin | man sox |
| 20:24.59 | fujin | gtfo |
| 20:25.08 | robeph | oh yeh its after lol |
| 20:25.09 | clyrrad | jblack: ok ill try there, have you done this before? |
| 20:25.12 | robeph | I use sox a lot |
| 20:25.16 | robeph | I've just been up going on 30hrs |
| 20:25.17 | robeph | =\ |
| 20:25.21 | jblack | what? Look up numbers on a website? Yes. |
| 20:25.37 | clyrrad | jblack: no, though Asterisk |
| 20:25.47 | clyrrad | jblack: to provide 411 functionality to an Asterisk PBX |
| 20:25.55 | jblack | write an asterisk app that lets people type in names, then screenscrape whitepages.com? No. |
| 20:26.36 | clyrrad | jblack: was more thinking along the lines if there is a service like how e911 works where you pay for the access |
| 20:27.15 | jblack | Perhaps you can find a whitepages xmlrpc service |
| 20:27.35 | clyrrad | yea, something like that is more along the lines of what I am looking for |
| 20:27.38 | fujin | does the 411 database contain ddi -> SIP mappings? |
| 20:27.45 | clyrrad | was just wondering if any of you guys have setup 411 on your PBX servers |
| 20:27.46 | fujin | what do you want to actually lookup? |
| 20:27.49 | jblack | Oh, was he looking for sip? |
| 20:27.53 | clyrrad | yes lookup |
| 20:27.57 | fujin | lookup what |
| 20:28.02 | fujin | name -> number? |
| 20:28.05 | clyrrad | People and Busiensses |
| 20:28.05 | fujin | number -> sip address? |
| 20:28.06 | jblack | if you're looking for sip addresses, then what you want is enum164.org |
| 20:28.13 | jameswf | r2 even... |
| 20:28.22 | clyrrad | No actaull phone numbers, like how 411 works on a POTS line or your cell phone |
| 20:28.36 | fujin | pretend you're talking to people who have no idea what 411 is. |
| 20:28.51 | fujin | is this simply a phone directory service? |
| 20:28.58 | jameswf | 1800 goog 411 |
| 20:29.01 | clyrrad | Exacly how e911 works..... you pay for getting access to the service, if someone calls 911 its routed to an operator, I am looking for the same kind of service for 411 |
| 20:29.22 | fujin | ok, pretend you're talking to someone who doesn't know what e911 is either. |
| 20:29.23 | *** join/#asterisk RoyK (n=roy@ip-219-54-149-91.dialup.ice.no) |
| 20:29.27 | clyrrad | fujin: yes, correct 411 is a direcotry service to find a phone number for a person or business |
| 20:29.35 | fujin | right. That's like 018 here. |
| 20:29.51 | fujin | an operator directory |
| 20:30.00 | clyrrad | fujin: yea, you call it, you get an operator that asks you whom your trying to find... then they give you the number, and you pay for the lookup |
| 20:30.00 | fujin | Asterisk shouldn't do that functionality. |
| 20:30.11 | fujin | wrong tool for the job |
| 20:30.11 | jameswf | ok so a phone is a thingy with 12 dealios called keys... keep up |
| 20:30.34 | clyrrad | fujin: yea, I am more looking for an a "service provider" to use for my askterisk box, so when users dial 411 I route the call to them |
| 20:30.50 | fujin | ah. |
| 20:30.54 | fujin | now you're making sense. |
| 20:30.55 | fujin | well |
| 20:30.57 | fujin | buy $provider |
| 20:31.00 | fujin | and then handle 411 in your dialplan |
| 20:31.01 | fujin | eof |
| 20:31.03 | clyrrad | Wondering if any of you guys have done this? |
| 20:31.05 | jameswf | clyrrad: route all 411 calls to GOOG 411 |
| 20:31.40 | clyrrad | jameswf: I am not famaliar with that looking it up on Google |
| 20:32.04 | fujin | http://www.google.com/goog411/ |
| 20:32.18 | clyrrad | checking that right now |
| 20:32.21 | clyrrad | how good is it? |
| 20:32.35 | fujin | As good as any free google service, one would hope |
| 20:32.45 | clyrrad | hehe |
| 20:32.52 | clyrrad | does it work for Canada too? |
| 20:32.58 | clyrrad | looks like just USA from what I see |
| 20:33.06 | fujin | ohhh canadaaa |
| 20:33.11 | fujin | how the hell would I know |
| 20:33.14 | fujin | fuckinggoogleit |
| 20:33.21 | clyrrad | Canada and the USA would be what im after |
| 20:33.31 | fujin | fwiw, we're not google-proxies |
| 20:33.44 | clyrrad | LOL |
| 20:33.54 | fujin | and this is heavily offtopic |
| 20:34.03 | fujin | if your question was 'how can I setup my dialplan to handle 411 calls and route them somewhere else' |
| 20:34.04 | clyrrad | im not asking for a google-proxy just if any of you have done it b4 |
| 20:34.06 | fujin | that would have been ontopic |
| 20:34.14 | jameswf | ~rickroll |
| 20:34.15 | jbot | [rickroll] http://www.internetisseriousbusiness.com |
| 20:34.26 | jameswf | ~fleamarket |
| 20:34.27 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
| 20:34.33 | fujin | http://www.mylazysunday.com |
| 20:36.14 | *** join/#asterisk harryv (n=fufufu@0x55508034.adsl.cybercity.dk) |
| 20:36.17 | harryv | crappy connection. |
| 20:36.57 | harryv | converted a mp3 to alaw with sox, named it mysound.alaw (tried .pcm aswel) and loaded codec_alaw.so |
| 20:37.08 | harryv | [Mar 12 22:37:16] WARNING[4412]: file.c:569 ast_openstream_full: File mintest does not exist in any format |
| 20:37.11 | harryv | [Mar 12 22:37:16] WARNING[4412]: file.c:868 ast_streamfile: Unable to open mintest (format 0x48 (alaw|slin)): No such file or directory |
| 20:37.14 | harryv | s/mysound/mintest |
| 20:39.32 | *** join/#asterisk kyron (n=kyron@211-217-static-ppp.3menatwork.com) |
| 20:40.43 | fujin | ls -la mintest? |
| 20:40.52 | harryv | format_pcm.so was missing.. |
| 20:40.54 | *** join/#asterisk jdg (n=jdg@203.185.181.87) |
| 20:49.20 | *** join/#asterisk clive- (n=pirch@dsl-242-162-91.telkomadsl.co.za) |
| 20:50.00 | clive- | hi, would you guys recommend centos 5 for running asterisk ? |
| 20:51.03 | mvanbaak | clive-: if you like centos and are comfortable with it, I would say: go for it |
| 20:51.38 | clive- | mvanbaak, hi thanks, I still have astreisk running on centos 3, but I havent built a system for a while ..:) |
| 20:51.57 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 20:52.15 | harryv | for load statistics, what do you recommend ? |
| 20:52.23 | mvanbaak | top |
| 20:52.28 | infinity3 | heh |
| 20:52.54 | JayTee52 | clive-, I just got done installing CentOS 5.1 for a new Asterisk box. |
| 20:52.56 | infinity3 | ya. i think everyone starts with top. if that doesn't help, well, keep digging |
| 20:53.15 | JayTee52 | it's just RHEL5 without the artwork |
| 20:53.23 | clive- | jaytee, and its playing nicely with asterisk ? |
| 20:53.45 | *** join/#asterisk harryv (n=fufufu@0x55508034.adsl.cybercity.dk) |
| 20:53.52 | harryv | for load statistics, what do you recommend ? |
| 20:54.12 | harryv | like, when i run a stress test, i would like to have some data afterwards |
| 20:54.13 | JayTee52 | well, it has in the past in a VM and my RHEL5 64 bit Asterisk system works like a champ so I'm confident CentOS5.1 will do just as well. |
| 20:54.23 | clive- | this is also my first attempt at using asterisk 1.4 on a production box |
| 20:55.00 | JayTee52 | well, ya gotta start somewhere. My first Asterisk box ran on Ubuntu Server 6.06 |
| 20:55.30 | clive- | cool, . I have servers running 1.2 , and once it works, I dont touch .... call me old fashioned |
| 20:55.32 | JayTee52 | I've found Asterisk works on most of the major linux distros just fine. |
| 20:55.49 | JayTee52 | ok, you're old fashioned! :-) |
| 20:56.00 | clive- | my first was asterisk 0.8 on redhat9 |
| 20:56.05 | JayTee52 | hey, I'm 54 so I know what you mean. If it ain't broke, don't fix it. |
| 20:56.39 | clive- | thats my motto.. sweated enough to get it working once |
| 20:57.01 | JayTee52 | I've had good results with 1.4 so far |
| 20:57.16 | clive- | I want 1.4 becaue of the sip jitter buffering |
| 20:57.21 | JayTee52 | and I'm eager to get 1.6 installed on a system to test. |
| 20:57.41 | clive- | 1.2 only does iax2 jitter buffering.... and only sip jitter buffering with lots of patching |
| 20:58.02 | clive- | unfortunately I dont have too much time to play and test.... |
| 20:58.04 | JayTee52 | sip jitter buffering? I thought jtter buffering was for IAX2 |
| 20:58.20 | *** join/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 20:58.27 | JayTee52 | hmmm, gonna have to look into that |
| 20:58.36 | clive- | it was only for iax2.... but now it can be used for sip as well.....yay.... |
| 20:59.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:59.48 | clive- | in places with less than ideal internet, jitter buffering is a pre-requisite |
| 20:59.52 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 21:01.46 | tld | clive-: On the UAS. Depending on how you use Asterisk, might not matter. |
| 21:01.52 | tld | UAs even |
| 21:02.01 | JayTee52 | is it a minor version addition? I'm running 1.4.15 |
| 21:02.36 | clive- | what are UA's ? |
| 21:03.12 | tld | clive-: User Agents |
| 21:03.17 | tld | endpoints |
| 21:03.20 | tld | phones and adapters |
| 21:03.47 | JayTee52 | ok, I'm done and it's quittin time. bye all! |
| 21:03.53 | *** part/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 21:04.42 | clive- | ok, tld, Its for when asterisk is the endpoint with a digium pri card |
| 21:05.03 | tld | for example |
| 21:05.27 | tld | but if everything is on SIP, and Asterisk is just passing the RTP streams around, it doesn't matter as much |
| 21:05.29 | clive- | 2 cisco phones passing rtp via astersisk do not need jitter buffering |
| 21:05.35 | tld | exactly |
| 21:05.45 | clive- | yup, ... :) |
| 21:05.48 | tld | or rather, they could probably benefit from it if their own buffer is crappy |
| 21:06.31 | clive- | tld, Its for when asterisk is the endpoint with a digium pri card |
| 21:09.14 | clive- | tld ok, Time for some shuteye .. nice chattin |
| 21:10.09 | *** part/#asterisk clive- (n=pirch@dsl-242-162-91.telkomadsl.co.za) |
| 21:12.15 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 21:17.27 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 21:17.46 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 21:18.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:22.14 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 21:23.10 | *** join/#asterisk RobH (n=RobH@216.207.245.1) |
| 21:25.44 | *** join/#asterisk jbigB (n=jBigB@216.207.245.1) |
| 21:28.41 | *** join/#asterisk kowalsky440 (n=ldesfoss@gra94-2-82-66-142-248.fbx.proxad.net) |
| 21:31.42 | tzanger | wow tk's not here? |
| 21:32.04 | kowalsky440 | hie :) |
| 21:33.35 | johndo | Can someone point me to documentation on regexten |
| 21:33.42 | johndo | ? |
| 21:34.03 | johndo | I don't think it works the way I thought it worked. |
| 21:35.33 | riddlebox | tzafrir, he is saving up his wit until later tonight |
| 21:37.06 | mvanbaak | johndo: what's the way you think it works ? |
| 21:37.33 | johndo | I figured that if I did regexten=666, dialing 666 would call that line. |
| 21:37.42 | johndo | I need to figure out how to include that in the dialplan |
| 21:37.42 | mvanbaak | ah |
| 21:37.48 | mvanbaak | no, that's not true |
| 21:37.50 | johndo | (: |
| 21:38.06 | mvanbaak | if you do regexten=666 |
| 21:38.29 | mvanbaak | it means when the phone registers it will create 'exten => 666,1,NoOp()' in the regcontext |
| 21:38.41 | johndo | I see |
| 21:38.44 | johndo | and what does that do? |
| 21:39.07 | mvanbaak | you can use that with dundi |
| 21:39.10 | johndo | is NoOp just a logging thing? |
| 21:39.24 | Katty | pretty much. |
| 21:39.32 | mvanbaak | NoOp == No Operation |
| 21:40.16 | johndo | It would have been nice if I could have overredden the action witha macro(stdexten,${EXTEN},${CONTEXT}) |
| 21:41.24 | johndo | I was hoping to use names for usernames |
| 21:41.41 | johndo | and then use regexten to setup the line numbers in order to keep the dialplan simple |
| 21:41.53 | johndo | it looks like it will be easier to use numbers for usernames |
| 21:42.29 | mvanbaak | yup |
| 21:42.46 | mvanbaak | exten => _6XX,1,Dial(SIP/${EXTEN}) |
| 21:42.57 | johndo | yeah, pretty much |
| 21:43.30 | johndo | I am getting the hang of contexts |
| 21:43.38 | johndo | I can use it like virtual hosting |
| 21:44.28 | mvanbaak | yup |
| 21:44.44 | adeel | i thought with SIP extensions, the extension could be any alphanumeric character? |
| 21:45.08 | johndo | adeel: it can, it is just easier to configure it to be all numbers |
| 21:45.19 | adeel | johndo, yeah, that i agree with |
| 21:46.13 | johndo | I was kind of hoping to use our ldap tree to authenticate users for asterisk |
| 21:46.26 | johndo | and then have an atrribute be their extensions. |
| 21:46.54 | *** join/#asterisk joetester (n=patrick@216.191.34.13) |
| 21:48.51 | joetester | Hey Corydon? Trying to run a read ODBC_ query from AGI... this one takes multiple arguments... I used GET VARIABLE ODBC_MYQUERY(Argument1,Argument2). Is that syntax all right? |
| 21:57.57 | jameswf | I should get an award for most amaizing waste of resources..... Quad T1/E1 designed to route 2 phones in an over complicated intercom :) |
| 22:00.22 | tzanger | jameswf: haha |
| 22:00.29 | tzanger | that's not a waste of resources |
| 22:00.32 | tzanger | that is future proofing |
| 22:02.42 | jameswf | [1line phone]---[channel bank]---[t1 LOOP]---[asterisk]---[e1 mfcr2]---[e1 mfcr2]---[t1 loop]---[channelbank]---[1line phone] |
| 22:03.59 | mvanbaak | wow |
| 22:06.49 | *** join/#asterisk anthm (n=anthm@mb10736d0.tmodns.net) |
| 22:06.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:08.35 | flush | k yo |
| 22:08.37 | flush | http://pastebin.com/m2a8c08f1 |
| 22:08.40 | flush | what does this mean |
| 22:08.47 | flush | the asterisk phone doesnt ring.. |
| 22:12.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:14.45 | jameswf | step 1 Cut a hole in the box |
| 22:20.30 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.or.comcast.net) |
| 22:23.52 | flush | hrmmm |
| 22:23.57 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:26.04 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 22:27.09 | vap0rtranz | wow. i just got dtmf working over channels via inband & ulaw. so sad :( |
| 22:28.14 | vap0rtranz | damn affinity |
| 22:28.30 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:32.48 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:33.55 | mvanbaak | sipdebug &= ~sip_debug_console; |
| 22:33.59 | mvanbaak | eh ? |
| 22:34.07 | obnauticus | does anyone ehre use ISPhone? |
| 22:34.15 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 22:39.43 | *** join/#asterisk klauwhamer (n=felixdhc@ipd50af070.speed.planet.nl) |
| 22:41.01 | *** join/#asterisk UserReg_CL (n=COB@200.113.99.247) |
| 22:43.32 | JT | obnauticus: i've used them |
| 22:43.41 | obnauticus | JT do you have a CSV rate table? |
| 22:43.45 | obnauticus | because i cannot find one on their site |
| 22:44.03 | *** part/#asterisk UserReg_CL (n=COB@200.113.99.247) |
| 22:44.24 | JT | no i don't |
| 22:44.31 | obnauticus | They provide one though, right? |
| 22:44.33 | JT | why would they have it on their site |
| 22:44.35 | JT | you need to ask |
| 22:44.40 | JT | they go through resellers |
| 22:44.49 | obnauticus | ya, that's why i asked you |
| 22:45.05 | obnauticus | this dude bought ISPhone for his * boxen he wants me to help setup routes to resel |
| 22:45.07 | obnauticus | resell* |
| 22:45.14 | JT | ask them |
| 22:45.30 | obnauticus | I jsut sent an email |
| 22:45.46 | JT | i don't have them |
| 22:45.48 | obnauticus | k |
| 22:45.55 | obnauticus | are they very good? |
| 22:46.17 | JT | they're alright |
| 22:46.33 | JT | i don't think any voip provider is "very" good :) |
| 22:47.06 | obnauticus | JT ya. |
| 22:47.27 | obnauticus | this guy is thinking about getting a DS1 or DS3 though... |
| 22:47.38 | *** join/#asterisk ph0ne (n=ph0ne@206-248-164-27.dsl.teksavvy.com) |
| 22:47.59 | JT | is he in australia? |
| 22:48.02 | obnauticus | No |
| 22:48.07 | obnauticus | Turkey.... |
| 22:48.11 | JT | oh |
| 22:48.12 | obnauticus | for istanbul... |
| 22:48.26 | JT | isphone is australian is it not? |
| 22:48.26 | obnauticus | I wonder if any datacenter will actually route a DS line to your cage. |
| 22:48.47 | JT | i think it's up to you to buy the relevant service from the telco |
| 22:48.57 | JT | the datacentre shouldn't care what cables you have running |
| 22:49.23 | obnauticus | well they would have to route the fiber and stuff wouldn't they |
| 22:50.09 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 22:50.53 | vodak | I think the only time a Datacenter would have a problem when routing DS1 or DS3 to hardware is if you are hosting at a bottom of the barrel cheap place. |
| 22:51.12 | obnauticus | lol |
| 22:51.34 | vodak | aka rinky dink small |
| 22:52.37 | JT | obnauticus: depends totally on the datacentre |
| 22:52.44 | JT | some datacentre do cabling for you at a cost |
| 22:52.47 | obnauticus | Ya, I would imagene, I told the dude to ask it. |
| 22:52.48 | JT | some don't |
| 22:52.56 | JT | some charge per month |
| 22:52.58 | JT | some don't |
| 22:53.08 | obnauticus | I haven't asked a datacenter above just upgrading what i get from their switches. |
| 22:53.16 | obnauticus | not for special dedicated lines... |
| 22:53.30 | JT | i prefer to use datacentres that are completely carrier neutral and do not levy monthly charges on cross connects and circuits |
| 22:54.16 | obnauticus | Ya, I have a cogent datacenter near me that is like that. |
| 22:55.07 | danilom | i have to use a E1 what card is better a sangoma or a digium |
| 22:55.35 | JT | danilom: none |
| 22:55.43 | danilom | uhmm |
| 22:55.46 | mvanbaak | danilom: with the digium card you sponsor asterisk development |
| 22:56.06 | danilom | JT so what to use ? |
| 22:56.40 | JT | danilom: sangoma, or pri to sip hardware gateways |
| 22:56.59 | danilom | mvanbaak, ok want to use a digium with 1 E1/T1/J1 port with echo cancellation |
| 22:57.12 | JT | lol |
| 22:57.16 | JT | do they even have one yet? |
| 22:57.25 | danilom | ? |
| 22:57.27 | mvanbaak | JT: they have one for some time now |
| 22:57.36 | JT | what, a few months? |
| 22:57.41 | mvanbaak | danilom: pci or pci-expres ? |
| 22:57.50 | danilom | mvanbaak, is pci.. |
| 22:57.51 | mvanbaak | JT: more like 6 months |
| 22:58.19 | mvanbaak | danilom: http://store.digium.com/productview.php?product_code=1TE122BF |
| 22:58.19 | danilom | $730 in the web store |
| 22:58.22 | obnauticus | Well, I'll need an E1/T1/J1 card, which one do you guys suggest? |
| 22:58.29 | obnauticus | i want max quality! |
| 22:58.30 | riddlebox | haha you want to hear something funny, we have a customer who wants to record every call for training purposes, so my boss quotes them asterisk, for the recording and to use as a voicemail for their legacy system, instead of just buying the 9 phones and just use asterisk!!! |
| 22:59.06 | JT | mvanbaak: i guess they had to catch up after having a gap in their product range |
| 22:59.44 | vodak | riddlebox: I've seen a bunch of places that want to hold on to the old P.O.S. systems because it makes them feel secure. Even if an Asterisk/Polycom solution is cheaper |
| 23:00.37 | mvanbaak | JT: digium is the only one with a 1port PRI card with hardware EC |
| 23:00.45 | JT | mvanbaak: you're crazy |
| 23:00.52 | mvanbaak | all the other vendors only give hardware EC on dual or higher cards |
| 23:00.55 | JT | sangoma's A101d came out well before any digium card |
| 23:01.01 | JT | single port with HWEC |
| 23:01.17 | mvanbaak | JT: no, that was released after the digium card |
| 23:01.25 | JT | that is not correct |
| 23:01.38 | JT | there was no digium card with HWEC at the time of the sangoma release |
| 23:01.48 | JT | at least not released |
| 23:01.54 | danilom | ok im going to use with asterisk in a debian server... digium or sangoma |
| 23:02.32 | mvanbaak | ah well |
| 23:02.41 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 23:03.07 | *** join/#asterisk oddn (n=fufufu@0x55508034.adsl.cybercity.dk) |
| 23:03.17 | mvanbaak | danilom: right now I would contact my reseller to see which one is available |
| 23:03.24 | mvanbaak | it really doesn't matter that much |
| 23:03.45 | danilom | mvanbaak, well is my first asterisk server i want some opinions |
| 23:04.10 | mvanbaak | I have installed both, and both without trouble |
| 23:04.18 | danilom | ok |
| 23:04.33 | mvanbaak | the nice thing with the digium cards is, you dont need additional drivers |
| 23:04.38 | mvanbaak | it's all in zaptel |
| 23:04.47 | mvanbaak | and you are installing that anywayz |
| 23:05.00 | danilom | how much hardware do i need to manage about 40 extensions the E1 and 2 analog.. |
| 23:05.32 | mvanbaak | E1 has 30 voice channels |
| 23:06.05 | danilom | ok, but 40 internal voip phones |
| 23:06.20 | obnauticus | an E1 has 23B+D right? |
| 23:06.22 | obnauticus | 24 channels |
| 23:06.30 | mvanbaak | obnauticus: that's T1 |
| 23:06.39 | obnauticus | we's talking about PRI's? |
| 23:06.40 | mvanbaak | E1 has 30B+D |
| 23:07.09 | mvanbaak | obnauticus: PRI in USA is not the same as PRI in EU |
| 23:07.16 | obnauticus | oh mmk |
| 23:07.21 | mvanbaak | danilom: really depends on the setup |
| 23:07.25 | obnauticus | so if i were in Europe and got an E1 |
| 23:07.28 | JT | PRI in rest of world really ;) |
| 23:07.28 | obnauticus | it would be 31 channels? |
| 23:07.33 | JT | no, 30 |
| 23:07.37 | obnauticus | 30B+D |
| 23:07.38 | obnauticus | ? |
| 23:07.41 | JT | it's not available in RBS |
| 23:07.52 | JT | that i've ever seen |
| 23:07.52 | mvanbaak | danilom: we have several locations where I have 100 phones on a soekris 4801 |
| 23:07.52 | obnauticus | is it 30 channels total or 30B+D? |
| 23:08.00 | JT | it's 32 |
| 23:08.03 | danilom | ok... |
| 23:08.03 | JT | 30B |
| 23:08.05 | JT | 1D |
| 23:08.08 | JT | 1 multiframe sync |
| 23:08.09 | obnauticus | 30B + 1 D = 31 |
| 23:08.11 | obnauticus | oh |
| 23:08.15 | mvanbaak | obnauticus: 30 B, 1D, 1 sync |
| 23:08.16 | danilom | what is a soekris 4801 |
| 23:08.24 | JT | T1 doesn't quite have a full channel for multiframe sync |
| 23:08.36 | oddn | i have loaded cdr_csv.so and [general]; enabled=yes; size=1 |
| 23:08.39 | JT | but if you do 24 * 64 it doesn't equal a T1 linerate |
| 23:08.41 | mvanbaak | danilom: http://www.soekris.com/net4801.htm |
| 23:08.43 | oddn | but nothing is written.. |
| 23:09.18 | mvanbaak | oddn: did you restart asterisk ? |
| 23:09.24 | oddn | yes |
| 23:09.32 | mvanbaak | oddn: did you check file/folder permissions ? |
| 23:09.34 | danilom | OK!! |
| 23:09.44 | danilom | is an embedded solution |
| 23:09.57 | mvanbaak | danilom: of course, those are installs where we have no transcoding |
| 23:10.01 | oddn | mvanbaak: this is a test installation, asterisk is running as root. |
| 23:10.04 | oddn | [Mar 13 01:07:23] NOTICE[5398]: cdr.c:1369 do_reload: CDR simple logging enabled. |
| 23:10.08 | mvanbaak | all phones and the PRI run the same codec |
| 23:10.34 | obnauticus | JT if i get an E1 |
| 23:10.35 | obnauticus | what do i get? |
| 23:10.45 | obnauticus | and what do i need to configure if i plug it directly into an E1 card? |
| 23:11.02 | mvanbaak | obnauticus: you will get an RJ45 cable |
| 23:11.12 | mvanbaak | able to run 30 concurrent voice calls |
| 23:11.20 | obnauticus | ... |
| 23:11.25 | obnauticus | what do i need to configure |
| 23:11.32 | mvanbaak | your E1 card |
| 23:11.37 | JT | obnauticus: zaptel if that's what you're using |
| 23:11.38 | danilom | mvanbaak, what linux can be installed |
| 23:11.44 | JT | hdb3 crc4 usually |
| 23:11.52 | mvanbaak | danilom: any |
| 23:12.10 | mvanbaak | danilom: we are running debian on a 1GB CF card on those |
| 23:12.19 | danilom | mvanbaak, cool |
| 23:12.31 | mvanbaak | we also use them for our firewalls |
| 23:12.49 | danilom | iptables and squid? |
| 23:13.23 | mvanbaak | no |
| 23:13.25 | mvanbaak | OpenBSD |
| 23:13.39 | mvanbaak | danilom: this is our default firewall: http://soekris.kd85.com/images/tn/DSC05702.med.JPG |
| 23:13.56 | obnauticus | mvanbaak i like. |
| 23:13.57 | mvanbaak | redundant dual firewall in a 1U case |
| 23:14.14 | obnauticus | with CARP? |
| 23:14.15 | obnauticus | and pfsync |
| 23:14.17 | mvanbaak | yup |
| 23:14.20 | obnauticus | I do that |
| 23:14.23 | obnauticus | with 3 firewalls |
| 23:14.37 | obnauticus | Quad core xeon with 2GB of ram each |
| 23:14.41 | mvanbaak | gheh |
| 23:14.43 | obnauticus | i had them sitting around. |
| 23:14.44 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 23:14.46 | mvanbaak | we do that with the soekris |
| 23:14.47 | mvanbaak | :) |
| 23:15.05 | mvanbaak | here's the setup I used in our initial install: http://michiel.vanbaak.info/soekrisobsdcarp.htm |
| 23:15.23 | mvanbaak | I know, I should update it to obsd 4.2 |
| 23:15.27 | JT | how man Mbps and pps do they push? |
| 23:15.57 | mvanbaak | JT: 30% cpu on our 100mbit internet line |
| 23:16.22 | mvanbaak | avg usage is 80mbit |
| 23:16.35 | JT | all on the 4801? |
| 23:16.41 | mvanbaak | yup |
| 23:16.54 | mvanbaak | you'll need OpenBSD 4.2 for that |
| 23:16.56 | mvanbaak | or newer |
| 23:17.05 | JT | why is that |
| 23:17.09 | mvanbaak | older versions will do like 30 to 40 mbit |
| 23:17.18 | JT | ah ok |
| 23:17.27 | mvanbaak | JT: they rewrote the whole interrupt stuff in the network layer |
| 23:17.41 | JT | isn't the chipset realtek or something cheap on the 4801s? |
| 23:17.50 | mvanbaak | via |
| 23:18.27 | mvanbaak | in 4.2 they did a great job to really use the via stuff |
| 23:18.51 | mvanbaak | so now you can get close to 90mbit real traffic on it |
| 23:18.56 | JT | will the push much in the >100Mbit/s range? |
| 23:19.12 | mvanbaak | JT: you'll need the 5501 for that |
| 23:19.37 | mvanbaak | http://www.soekris.com/net5501.htm |
| 23:20.01 | JT | the problem is all the economically priced embedded stuff with gigabit use crappy chipsets |
| 23:20.02 | ph0ne | Oh brilliant my OS of choice with my hardware of choice |
| 23:20.03 | mvanbaak | if you put a 10Ge card in there, you will get a lot of bandwidth |
| 23:20.21 | JT | if it's not intel or broadcom, it's a waste of time with gigabit |
| 23:20.52 | mvanbaak | JT: all them boards have a pci slot |
| 23:21.02 | mvanbaak | so feel free to put a 4port 10Ge card in it |
| 23:21.08 | ph0ne | I just bought a net5501 |
| 23:21.26 | mvanbaak | ph0ne: should have bought 2 |
| 23:21.34 | mvanbaak | and do carp+pfsync on them |
| 23:21.40 | JT | haha as if they'd be able to push 10Gbit/s |
| 23:21.49 | ph0ne | why VPN? |
| 23:21.52 | JT | you need hardware ASICs and FPGAs for that |
| 23:22.05 | mvanbaak | JT: a good NIC will have that on board |
| 23:22.23 | JT | yes but the os won't be able to feed them |
| 23:22.32 | mvanbaak | obsd can handle that |
| 23:22.33 | JT | with all the interrupts |
| 23:22.59 | mvanbaak | nice thing in the latest obsd: they can do more then one opperation per interrupt |
| 23:23.13 | mvanbaak | that's why they are able to push 90mbit on the via nics |
| 23:23.16 | JT | acting purely as a router/packet mover? |
| 23:23.24 | JT | not off any local source i assume |
| 23:23.27 | mvanbaak | no, even with pf enabled |
| 23:23.42 | JT | and any links to real world scenarios with close to 10Gbit/s performance? |
| 23:23.47 | JT | i meant as opposed to a server |
| 23:23.54 | fujin | Anyone here familiar with ARA queues? there are a couple of settings missing from the db schema I installed, which I'd like to set |
| 23:23.58 | JT | making the content that goes in packets |
| 23:24.01 | fujin | mostly (ringinuse = no) |
| 23:24.17 | mvanbaak | JT: www.undeadly.org had some articles about the latest hacketon where they did this shit |
| 23:24.19 | fujin | is it simply a matter of adding a value to the db with the same name, or is ARA configured to only look at certain things |
| 23:24.57 | mvanbaak | fujin: I have no idea. I only use flat configfiles |
| 23:25.09 | fujin | blagh |
| 23:25.11 | fujin | get out! |
| 23:25.31 | mvanbaak | I dont want my phone system to be dependant on a database system |
| 23:26.27 | mvanbaak | I put all my configs in a database, create config files from it, put those in svn and use that to provision my servers |
| 23:26.47 | mvanbaak | works great for asterisk, apache, sshd, bind etc |
| 23:27.26 | fujin | I'm doing one-way replication of mysql to slave mysqld's running on both of my asterisk cluster nodes |
| 23:27.36 | fujin | so, it's not reliant on network access to a remote db |
| 23:27.48 | mvanbaak | fujin: yeah, but that becomes a bitch when you go over 10 asterisk boxen |
| 23:28.03 | fujin | not really, the database traffic is minimal |
| 23:28.08 | fujin | and one-way replication in sql is easy |
| 23:28.08 | mvanbaak | no, it's not |
| 23:28.20 | mvanbaak | what if only 8 of the 10 boxen can reach the master |
| 23:28.25 | mvanbaak | and some are out of sync ? |
| 23:28.25 | fujin | I dunno, maybe in the day of 10baseTx |
| 23:28.32 | fujin | then 2/10 aren't active? |
| 23:28.42 | mvanbaak | with ARA they are |
| 23:28.55 | mvanbaak | they just look at the local mysql database |
| 23:29.00 | fujin | I alert based on slave status, anyway |
| 23:29.08 | fujin | so if slave status on any of my two nodes is out, i whack em |
| 23:29.14 | fujin | and it's not active/active anyway |
| 23:29.32 | mvanbaak | if it works, nice setup :) |
| 23:29.33 | mvanbaak | really |
| 23:29.41 | mvanbaak | my setup is just one way to do it |
| 23:29.49 | mvanbaak | no _THE_ way |
| 23:30.20 | mvanbaak | but I trust md5sum and sha1sum more then some db replication thing |
| 23:30.59 | fujin | mm, sure |
| 23:31.02 | mvanbaak | ok, time to get some sleep |
| 23:31.06 | fujin | haven't really had much issue with one-way mysql master->slave replicate |
| 23:31.09 | fujin | althkugh i'd rather use LDAP tbh |
| 23:31.11 | fujin | hehe |
| 23:31.11 | fujin | seeya |
| 23:31.16 | mvanbaak | latero all |
| 23:31.35 | ph0ne | bye |
| 23:32.51 | riddlebox | anyone using broadvoice? |
| 23:33.18 | fujin | wrong channel buddy |
| 23:34.05 | riddlebox | fujin, you talking to me? |
| 23:34.14 | fujin | yea |
| 23:34.20 | fujin | this is asterisk, that other phone system you know? |
| 23:39.53 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 23:40.43 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 23:41.07 | *** join/#asterisk thinko (i=jdoe6alp@smaug.rackdragon.com) |
| 23:41.19 | ZX81 | hi all, anyone know the meaning of "Internal RTCP NTP clock skew detected" - machine is synched to NTP with very little drift |
| 23:42.20 | *** join/#asterisk bigdawggi (n=bigdawgg@mx1.hnwo.org) |
| 23:43.37 | riddlebox | fujin, do you even know what broadvoice is? |
| 23:43.58 | bigdawggi | Hi everyone, my boss is looking for a couple bullet points on the benefits of upgrading to Asterisk 1.4 from 1.2 -- can anyone help? |
| 23:44.02 | ph0ne | is it an itsp riddlebox? |
| 23:44.34 | ZX81 | bigdawggi, it gets bugfixes |
| 23:44.35 | ZX81 | :) |
| 23:44.39 | bigdawggi | :D lol |
| 23:44.41 | bigdawggi | thanks! |
| 23:44.53 | riddlebox | ph0ne, yeah, I am having problems making calls out, calls can come in, but I get their error message that says to contact them, but get put on hold right away haha |
| 23:44.57 | ZX81 | I did an article on the changes when it came out |
| 23:44.58 | ZX81 | ~adn |
| 23:44.59 | jbot | well, adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 23:45.56 | ph0ne | riddlebox: never had that problem? |
| 23:46.16 | riddlebox | yeah, my mom uses broadvoice as well and hers is doing the same thing |
| 23:47.42 | ph0ne | riddlebox: change to real itsp? |
| 23:48.30 | ZX81 | noone knows about the RTCP NTP clock skew? |
| 23:49.08 | TJNII | riddlebox: I'm using broadvoice |
| 23:50.09 | TJNII | Their system was kind of flaky earlier today. Swinging from unreachable to reachable several times a minute for about an hour. It does that every now and again. |
| 23:50.50 | riddlebox | ph0ne, like who vonage broadvoice is good |
| 23:51.06 | ph0ne | ~itsp-us |
| 23:51.14 | ph0ne | ~itsp |
| 23:51.15 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:51.31 | ph0ne | ~itsplist-us |
| 23:51.31 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 23:52.20 | TJNII | ph0ne: ^ oh look, there it is. |
| 23:52.35 | riddlebox | :) |
| 23:52.50 | ph0ne | if you are in Canada I would say recommend les.net |
| 23:53.15 | riddlebox | I still like that someone thought it was another phone system |
| 23:55.24 | _ShrikE | ~mini-mall |
| 23:55.25 | jbot | It's just like, it's just like, a russell.. b |
| 23:55.31 | *** join/#asterisk RoyK (n=roy@ip-27-3-149-91.dialup.ice.no) |
