| 00:15.07 | *** join/#asterisk Cyford (i=geegs1@c-24-99-118-189.hsd1.ga.comcast.net) |
| 00:30.40 | *** join/#asterisk Raky-2 (n=John@220.157.75.246) |
| 00:32.27 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9ca503e1fb459a47) |
| 00:35.44 | *** join/#asterisk ScurvyDawg (n=scurvyda@S0106000d883f28a0.gv.shawcable.net) |
| 00:38.21 | Iamnacho | ~book |
| 00:38.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 00:49.11 | *** join/#asterisk rummey (n=mike@c-75-72-151-125.hsd1.mn.comcast.net) |
| 00:49.33 | rummey | I have a digium question: If I get a Silver Subscription, am I required to renew it each year in order to keep using the software? |
| 00:51.02 | rummey | a stumper? |
| 00:54.24 | *** join/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 00:54.36 | frocos11292 | hey guys... got a problem |
| 00:55.12 | frocos11292 | need to originate a call thru ast api, but i need to know the channel right away |
| 00:55.15 | frocos11292 | any ideas? |
| 00:55.46 | rummey | I think zombies ate them all |
| 01:00.10 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 01:15.43 | *** part/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 01:17.19 | *** join/#asterisk litage|w (n=nick@70.55.220.203.static.comindico.com.au) |
| 01:18.00 | *** join/#asterisk Corydon76-dig (i=silver@pdpc/supporter/bronze/Corydon76-home) |
| 01:18.00 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 01:18.22 | Qwell | test |
| 01:18.47 | Iamnacho | echo |
| 01:22.19 | khronos | Back. |
| 01:22.30 | khronos | Just saw some messages a couple posted. |
| 01:22.50 | khronos | From my iaxy I had it configured for iax to talk to three different servers. |
| 01:23.21 | khronos | These servers had conferences on each as well as had sip trunks out to Axvoice. |
| 01:25.10 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:25.10 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:25.39 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 01:25.46 | dmz | howdy, anyone use cisco IP phones? |
| 01:26.59 | WilliamK | quite a few do |
| 01:27.37 | WilliamK | do you know Cisco has sold over 1 million of them? |
| 01:28.07 | dan__t | Hrm... So this Polycom phone supposedly supports 802.1Q. It also has a LAN port. So I have inet -> polycom -> pc. Wondering if the bridge on the phone is transparent, so I can VLAN the phone, but keep the PC behind it on a different VLAN. |
| 01:28.19 | dan__t | I think this one has your name all over it, [TK]D-Fender heh |
| 01:31.28 | dan__t | I don't even know if I can do 802.1Q on this router |
| 01:34.33 | *** join/#asterisk Braxus (n=bhsieh@66.147.214.164) |
| 01:44.17 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:44.42 | *** join/#asterisk Twister (n=bob@71-213-215-72.sxcy.qwest.net) |
| 01:45.04 | Iamnacho | can anyone tell me where to start troubleshooting this problem: i have no audio from any of my extensions. They are all on the same lan and no firewalls are running. I am at a total loss of what to check. |
| 01:46.53 | *** join/#asterisk Zenith77 (n=moose@c-76-110-200-130.hsd1.fl.comcast.net) |
| 01:49.54 | *** join/#asterisk coppice (n=chatzill@142.204.17.210.dyn.pacific.net.hk) |
| 01:53.35 | TJNII | Iamnacho: No audio phone to phone, no audio on an echo test, no audio on playback(hello-world)? |
| 01:53.37 | *** join/#asterisk TheCops (n=henri@got.securebinary.com) |
| 01:54.19 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 01:55.44 | Iamnacho | im definantly a newbie. i have only tested phone to phone. i will test the others |
| 01:56.08 | TJNII | Create an exten that plays back a sound file and try that. |
| 01:56.20 | TJNII | Also asterisk -rvvvvvv is your friend. |
| 01:56.57 | Iamnacho | thanks for the tips |
| 01:59.28 | Zenith77 | Hi. |
| 01:59.37 | Zenith77 | Does this channel only offer Linux Asterisk help? |
| 01:59.44 | Zenith77 | Or Asterisk Win32 help as well? |
| 02:00.08 | Zenith77 | However, it should be noted I don't think my problem has to do with my OS (although it does have a signficant impact) |
| 02:00.34 | Zenith77 | When I make a test call to Asterisk, everything works through the dial plan okay |
| 02:00.43 | Zenith77 | Just for some reason Playback() doesn't play back the sound file... |
| 02:01.05 | Zenith77 | If he anyone is willing to look into this further, please say so (don't want to waste breath ^^) |
| 02:01.37 | Twister | Zenith77: paste the section of your dialplan that pertains to your playback to me in a pm |
| 02:01.49 | Zenith77 | err hang on |
| 02:01.51 | Zenith77 | on my other comp ^^ |
| 02:02.31 | TJNII | Zenith77: I've found when playback doesn't work its a path problem |
| 02:02.40 | Zenith77 | That's what I though. |
| 02:02.50 | Zenith77 | I've tried both just the sound file name, and a relative path |
| 02:03.03 | Zenith77 | but now that I come to think about it, would it help if I tried a direct path? |
| 02:03.06 | TJNII | Remember, don't put the extension on the filename. |
| 02:03.18 | TJNII | Asterisk uses the easiest codec and tacs on the extension. |
| 02:03.29 | Zenith77 | http://zenith.ampaste.net/109922 |
| 02:03.38 | Zenith77 | there's my dial plan |
| 02:03.49 | TheCops | suggestion for a good softphone on windows XP ? (other then eyebeam) |
| 02:03.51 | Zenith77 | TJNII, nope no extension. |
| 02:04.02 | TJNII | Playback(/var/lib... |
| 02:04.10 | Zenith77 | The_Ball, X-Lite or WengoPhone |
| 02:04.17 | TJNII | No leading /, so it won't start from root |
| 02:04.29 | Zenith77 | ummm |
| 02:04.40 | Zenith77 | It doesn't have a leading / |
| 02:04.43 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 02:05.01 | phix | WARNING[14348]: chan_zap.c:3958 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. |
| 02:05.03 | Zenith77 | exten => 77, n, Playback(var/lib/sounds/hello-world) |
| 02:05.14 | Twister | zen |
| 02:05.16 | phix | <PROTECTED> |
| 02:05.17 | Zenith77 | yes? |
| 02:05.21 | Twister | Playback(/var/lib/ |
| 02:05.25 | *** join/#asterisk J4k3 (n=jsuter@pimpin.aint.easy.in.grapeland.us) |
| 02:05.31 | phix | What do these messages mean and how do I resolve it? |
| 02:05.58 | Zenith77 | Twister, okay I will try this. But I thought this makes it look from the root? |
| 02:06.15 | Twister | well where are you trying to make it.. |
| 02:06.30 | Zenith77 | here let me get you the exact path |
| 02:06.32 | Twister | why dont you just put your sound file in the same directory as the default asterisk files |
| 02:06.42 | Zenith77 | they are |
| 02:06.43 | Zenith77 | -.- |
| 02:06.49 | Zenith77 | I'm just trying to play a default sound file |
| 02:07.03 | Zenith77 | (remember I'm using if Asterisk Win32 if that makes a difference) |
| 02:07.06 | TJNII | Hmmm.. And Playback(hello-world) doesn't work? |
| 02:07.11 | Zenith77 | nope |
| 02:07.27 | TJNII | What does the console say? |
| 02:07.36 | Twister | then you need to do x:\dir\dir2\sound |
| 02:08.27 | Zenith77 | okay will try, thank you Twister :) |
| 02:09.29 | javb | when i enter "wget http://downloads.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz" i get an html file, any ideas? |
| 02:10.10 | TJNII | That command works for me |
| 02:10.40 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:12.17 | javb | It works on my laptop but not on the server i `m installing |
| 02:12.22 | javb | weird, isnt it? |
| 02:12.31 | TJNII | Indeed |
| 02:12.37 | Zenith77 | Twister, tried but to no avail :'( |
| 02:12.50 | Zenith77 | Is there any setting that would prevent the user from receiving the media? |
| 02:12.55 | TJNII | Zenith77: Does the console give any clues? |
| 02:12.56 | Raky-2 | javb, are you behind a proxy? |
| 02:12.57 | Zenith77 | Perhaps reinvite or something of the sort? |
| 02:13.05 | javb | Raky-2 not at all |
| 02:13.09 | Zenith77 | TJNII, yea, it prints everything, I just don't know what I'm looking for. |
| 02:13.13 | Zenith77 | I have debug, etc enabled. |
| 02:13.23 | Zenith77 | Would you like me to paste the logs? |
| 02:13.29 | Raky-2 | try lynx --source http://downloads.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz > asterisk-1.4-current.tar.gz |
| 02:13.46 | TJNII | Zenith77: The last 15 or so lines might be handy |
| 02:13.52 | Zenith77 | kk brb |
| 02:13.58 | javb | now, i did ssh to the server, which is next to my, and from my laptop i did the SAME cmd, and walla |
| 02:14.16 | TJNII | javb: Any DNS issues? |
| 02:14.38 | javb | TJNII: none |
| 02:18.06 | Zenith77 | TJNII, http://zenith.ampaste.net/109923 |
| 02:18.46 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:19.08 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:20.13 | Zenith77 | Err, like I said before just realized something |
| 02:20.19 | Zenith77 | Could this be caused by careinvite = no |
| 02:20.20 | Zenith77 | ? |
| 02:20.49 | Zenith77 | in sip.conf |
| 02:23.09 | riddlebox | is it not considered proper to put a t at the end of exten => 522,1,Dial(SIP/522,20,t)? |
| 02:25.59 | *** join/#asterisk blq (n=Bl@dslb-088-064-156-077.pools.arcor-ip.net) |
| 02:26.02 | Zenith77 | TJNII, any ideas? |
| 02:26.10 | TJNII | afk |
| 02:26.13 | Zenith77 | ah |
| 02:26.14 | Zenith77 | sorry |
| 02:26.54 | clyrrad | I had some files recorded in GSM format which sound great on PC, but on asterisk they sound terrible, any idea how to convert them so they sound good in Asterisk? |
| 02:27.17 | Zenith77 | clyrrad, perhaps the codec you're using? |
| 02:27.41 | clyrrad | Zenith77: channel is ULAW..... as far as I know - is that what you mean? |
| 02:27.48 | Zenith77 | err no |
| 02:28.00 | Zenith77 | I'm a bit to new to Asterisk myself, so I can't really explain it in detail. |
| 02:28.13 | clyrrad | uh? |
| 02:28.14 | Zenith77 | I remember reading it in the manual somewhere, you can set a codec to use or something. |
| 02:28.24 | TJNII | Zenith77: I spotted "Cannot find extension context 'demo'" Is that the entire extensions.conf |
| 02:28.39 | Zenith77 | err TJNII, actually that's from the default sip friends |
| 02:28.42 | Zenith77 | IT's unused. |
| 02:28.45 | Zenith77 | in extension 77 |
| 02:28.52 | Zenith77 | in fact |
| 02:28.59 | Zenith77 | I should probably change those just to get rid of that error |
| 02:29.31 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 02:30.45 | Zenith77 | err oh wow |
| 02:30.52 | Zenith77 | It's in the [general] section |
| 02:30.55 | Zenith77 | That might not be good ^^ |
| 02:33.45 | *** join/#asterisk tax0n (n=malder@host-84-9-229-171.bulldogdsl.com) |
| 02:33.46 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:36.48 | clyrrad | any takers? |
| 02:37.10 | clyrrad | the GSM files sound very distorted, tried using SOX to convert but am having no luck |
| 02:37.33 | clyrrad | I would like to convert the GSM to WAV or antying that will sound decent on Asterisk |
| 02:37.44 | clyrrad | any takers? |
| 02:37.44 | TJNII | Zenith77: Context mistake? ;) |
| 02:38.26 | Zenith77 | lol, Now WengoPhone is acting up :X |
| 02:47.12 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:48.15 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 02:54.03 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 02:56.58 | *** join/#asterisk Flauto (n=zhao@71.194.141.225) |
| 02:57.48 | Flauto | i have a question on followme.conf. can i put it to use for sending calls to multi-people by config this file |
| 02:58.46 | Flauto | like calls to my extension send followme call to my cell, my wife's extension would send follow me call to her cell |
| 02:58.47 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 02:59.39 | Flauto | hello all |
| 03:00.11 | Qwell | clyrrad: if your source format sounds bad, your destination format will to |
| 03:00.13 | Qwell | too |
| 03:00.31 | javb | Zup Flauto |
| 03:00.50 | Flauto | javb, how are you doing |
| 03:01.26 | javb | Chill, looking for a Red Bull :p |
| 03:02.27 | Flauto | javb, i have never used followme.conf to send out followme calls, what i do is set a macro in dialplan to send calls out. what is the advantage by using followme.conf? |
| 03:02.56 | clyrrad | Qwell: my source format sounds great on my PC, only the Playback on Asterisk sounds bad |
| 03:03.16 | clyrrad | Qwell: its like the encoding is not correct for Asterisk |
| 03:03.36 | clyrrad | Qwell: I have tried using SOX to correct the encoding, but am having no luck |
| 03:04.00 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:05.39 | Flauto | Qwell, for tdm 400 card with two fxs and two fxo posts, when i install zaptel, how do i modprobe them |
| 03:06.03 | *** join/#asterisk blq (n=Bl@dslb-088-064-157-128.pools.arcor-ip.net) |
| 03:06.33 | Flauto | modprobe zaptel modprobe wcfxo modprobe wcfxs ztcfg? |
| 03:06.47 | hmmhesays | oh my roomate is retarded |
| 03:07.01 | hmmhesays | i love it when people as "what file extension do I need to play this file" |
| 03:08.08 | Zenith77 | implode.exe >:) |
| 03:09.48 | Zenith77 | okay I give up, everything all of a sudden stops working |
| 03:09.52 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 03:09.56 | Zenith77 | thank you TJNII and Twister for your guys help :). |
| 03:11.10 | TJNII | That's not good |
| 03:11.40 | Zenith77 | well asterisk is working |
| 03:11.53 | Zenith77 | just WengoPhone, and X-Lite has never done anything for me anyways. |
| 03:13.07 | Zenith77 | X-Lite that is... |
| 03:15.01 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 03:17.41 | *** join/#asterisk ozus (n=ozus@61.152.175.216) |
| 03:18.30 | Zenith77 | TJNII, I finally got to test it. |
| 03:18.36 | Zenith77 | Nothing changed :S |
| 03:18.52 | TJNII | Zenith77: Which context is your phone in? |
| 03:18.58 | Zenith77 | default |
| 03:19.23 | Zenith77 | the one I pasted for you, but there is one thing that sticks out |
| 03:20.13 | TJNII | Oh wait |
| 03:20.22 | TJNII | Are you running asterisk and the softphone on the same box? |
| 03:20.28 | Zenith77 | yea :S |
| 03:20.38 | Zenith77 | I have no choice atm... |
| 03:20.44 | TJNII | That may not work as they will both try to bind to port 5060 to listen for sip |
| 03:21.03 | TJNII | Try the softphone on another computer or changing the SIP port on the softphone |
| 03:21.21 | TJNII | Assuming you're using SIP, of course.... |
| 03:21.38 | Zenith77 | yea |
| 03:21.45 | Zenith77 | I'll have to wait to try another box |
| 03:21.48 | Zenith77 | I have two sitting here |
| 03:21.55 | Zenith77 | just that the one I'm typing on right, something is wrong with it :S |
| 03:22.02 | *** join/#asterisk circas (n=dom_paq@CPE0015e985d53c-CM0011aec7a4c6.cpe.net.cable.rogers.com) |
| 03:22.06 | Zenith77 | I can ping it, but it can't ping the other machine (the one asterisk) is one back. |
| 03:22.22 | TJNII | You can try to muck with the ports, but if you have another machine that would be best for testing |
| 03:22.28 | Zenith77 | yea |
| 03:22.31 | TJNII | As changing ports adds another variable. |
| 03:22.48 | Zenith77 | yea |
| 03:22.49 | Zenith77 | :S |
| 03:23.14 | Zenith77 | I'll wait till tomorrow |
| 03:23.19 | Zenith77 | thanks for you help TJNII :) |
| 03:23.21 | TJNII | np |
| 03:25.21 | circas | hi fellow * users! |
| 03:26.51 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 03:26.54 | circas | anyone know queues a lot! |
| 03:26.55 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 03:27.47 | TJNII | I got one working. Does that count? |
| 03:28.10 | circas | no it dosent |
| 03:28.12 | circas | lol |
| 03:28.17 | TJNII | mmmkay |
| 03:30.10 | TJNII | Out of curiosity, what are you trying to find out? |
| 03:30.55 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.57) |
| 03:32.02 | circas | well i think i'm trying to do strange stuff! |
| 03:32.39 | circas | like inserting @ a specific pos in the que |
| 03:33.04 | TJNII | Don't the call back scripts do that? |
| 03:33.29 | circas | call back scripts? |
| 03:34.10 | TJNII | Aah, never mind. I was thinking of something else |
| 03:34.33 | circas | and i also wana like go back in the dialplan instead of distributing to an agent |
| 03:35.20 | TJNII | So go to an extension instead of a phone? How does it know how long to keep someone in the queue? |
| 03:35.53 | circas | one of the the things i was thinking of doing is this |
| 03:36.50 | circas | i think it would make call centers a hole lot better... |
| 03:37.26 | circas | you call and have the system call you back when its your turn to speak to an agent |
| 03:37.39 | circas | instead of waiting |
| 03:37.48 | TJNII | Yea, I think there are some scripts that do that. |
| 03:38.11 | circas | really |
| 03:38.16 | TJNII | I know I read about it somewhere, don't remember where. |
| 03:38.24 | circas | like AGIs |
| 03:38.29 | TJNII | Indeed |
| 03:38.44 | TJNII | Though I don't remember where I read about it, so it may not be for * |
| 03:39.37 | TJNII | http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback |
| 03:40.15 | circas | thx, i'll check it out |
| 03:40.33 | TJNII | Sorry to take the wind out of your sails. :) |
| 03:42.31 | TJNII | I should put my mpd control AGI up on voip-info.... |
| 03:42.59 | circas | i dident read it yet lol |
| 03:44.42 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 03:47.24 | circas | interesting |
| 03:50.59 | circas | do you think its a good idea to modify app_queue.c directly |
| 03:51.19 | circas | and have it do what i want |
| 03:51.36 | TJNII | I don't really know. If you do it right, I'd say no. |
| 03:51.40 | *** part/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 03:51.56 | TJNII | Heck, you could make a new feature. |
| 03:52.07 | TJNII | Then again I'm not a dev |
| 03:52.13 | *** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.109) |
| 03:52.19 | vitaminmoo | avahi |
| 03:52.36 | *** join/#asterisk bmg505 (n=leon@196.209.183.36) |
| 03:52.52 | circas | im a dev but i dident look at the asterisk code yet |
| 03:59.26 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.18 | *** join/#asterisk Jeremy223 (n=no@CPE-75-81-42-52.kc.res.rr.com) |
| 04:00.36 | Jeremy223 | hellooo |
| 04:00.40 | *** join/#asterisk serpent-fly (n=serpent@194.79.34.10) |
| 04:00.59 | Jeremy223 | anyone awake and care to help with a bizarro issue we're having? |
| 04:01.37 | circas | yeah if i can lol |
| 04:02.23 | Jeremy223 | well we didn't make any changes recently, and all of a sudden our main phone # / menu works, but a lot of extensions are giving fast busy signals, and the few that don't do that, just ring and ring and never go to voicemail (comedian) |
| 04:02.35 | Jeremy223 | <PROTECTED> |
| 04:02.35 | Jeremy223 | Oct 15 23:00:21 DEBUG[7663]: chan_zap.c:1405 zt_enable_ec: Enabled echo cancellation on channel 1 |
| 04:02.35 | Jeremy223 | <PROTECTED> |
| 04:02.35 | Jeremy223 | Oct 15 23:00:21 NOTICE[8313]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 04:02.35 | Jeremy223 | <PROTECTED> |
| 04:02.37 | Jeremy223 | Oct 15 23:00:21 DEBUG[8313]: app_dial.c:1587 dial_exec_full: Exiting with DIALSTATUS=CHANUNAVAIL. |
| 04:02.39 | Jeremy223 | <PROTECTED> |
| 04:03.48 | Jeremy223 | our asterisk guy who set up the server from scratch bailed on us and just left me with a guide on setting up new extensions (yay)... not really sure how to troubleshoot, already restarted asterisk / reloaded sip/extensions/etc / rebooted the server even |
| 04:04.15 | circas | hmmm |
| 04:04.32 | circas | it used to work before? |
| 04:04.48 | TJNII | Unable to create channel of type 'SIP' (cause 3 - No route to destination): Anything happen to the network? |
| 04:04.51 | Jeremy223 | yep randomly quit working a few hours ago, we have 50+ employees that had working extensions until then |
| 04:05.10 | citats | Jeremy223: you should use ${EXTEN} instead of BYEXTENSION. did someone upgrade from a really old asterisk to something more recent? |
| 04:05.11 | TJNII | They're using SIP phones? |
| 04:05.20 | *** join/#asterisk serpent-fly (n=serpent@194.79.34.10) |
| 04:05.20 | circas | there all sip phones right |
| 04:05.25 | Jeremy223 | yep all SIP phones |
| 04:05.37 | TJNII | sip show peers shows them all ok? |
| 04:05.43 | Jeremy223 | I'm not sure about the setup, we built this out maybe a year or so ago |
| 04:05.57 | citats | Jeremy223: what version of asterisk is it? |
| 04:06.07 | Jeremy223 | I see a ton of them yep, all Unmonitored |
| 04:06.13 | Jeremy223 | 1.2.5 |
| 04:06.18 | circas | are u doing sip trunking with a sip provider? |
| 04:06.32 | Jeremy223 | I believe so yeah, we have a whole block of lines 816-222-xxxx |
| 04:06.57 | circas | maybee youre provider changed something? |
| 04:07.03 | circas | did you call them? |
| 04:07.20 | TJNII | Jeremy223: Does internal calling work OK? |
| 04:07.47 | Jeremy223 | internal gives fast busy too - can dial our main number and get our main menu ok but dialing extensions from there gives fast busy - and one or two extensions don't give a fast busy but just ring and ring - so I haven't tried talking to them yet |
| 04:08.01 | citats | well i'll be, 1.2 still has BYEXTENSION in it. i'd still change it over to use ${EXTEN} |
| 04:08.25 | Jeremy223 | I grepped the configs for BYEXTENSION but its only on some weird lines I'm not familiar with like |
| 04:08.25 | Jeremy223 | extensions.conf:exten => _XXXX,1,Dial,SIP/BYEXTENSION |
| 04:08.26 | Jeremy223 | extensions.conf:exten => _NXXNXXXXXX,1,Dial,Zap/g0/BYEXTENSION |
| 04:08.26 | Jeremy223 | e |
| 04:08.54 | circas | yeah I'dd start by connecting a sip phone (entry in sip.conf) and then getting a dial to work |
| 04:09.09 | TJNII | Jeremy223: Can you dial out on the phones whose extens you can't dial? |
| 04:09.21 | citats | the Dial,SIP/BYEXTENSION must be what is being used to dial when you get an incoming call. i assume your sip peers are all numbered the same as the DIDs |
| 04:09.25 | Jeremy223 | one sec I'm at home, installing eyebeam |
| 04:09.32 | *** join/#asterisk jsaunders (n=super@70.70.0.33) |
| 04:09.39 | Jeremy223 | yep numbered the same |
| 04:09.57 | TJNII | If you're ssh'd in, since you reset the server, does it know the IPs of all the sip phones? |
| 04:10.06 | Jeremy223 | most of our setup in our extensions config is like |
| 04:10.06 | Jeremy223 | exten => 1213,hint,SIP/1213,Jeremy Martin ; Jeremy Martin, 1213 |
| 04:10.06 | Jeremy223 | exten => 1213,1,Macro(gsiexten,1213,Jeremy Martin) ; |
| 04:10.37 | dmz | yeah 7960 now has 6 lines active :) and local pbx is sending & receiving calls w/pbx in colo. life is good again. |
| 04:10.46 | Jeremy223 | most of them |
| 04:10.47 | Jeremy223 | phone2*CLI> sip show peers |
| 04:10.47 | Jeremy223 | Name/username Host Dyn Nat ACL Port Status |
| 04:10.47 | Jeremy223 | 1297/1297 198.247.174.254 D N 4891 Unmonitored |
| 04:10.47 | Jeremy223 | 1228/1228 198.247.174.254 D N 4845 Unmonitored |
| 04:10.47 | Jeremy223 | 1295/1295 198.247.174.254 D N 1093 Unmonitored |
| 04:10.49 | Jeremy223 | 1270/1270 198.247.174.254 D N 18042 Unmonitored |
| 04:10.51 | Jeremy223 | 2006 (Unspecified) D N 0 Unmonitored |
| 04:10.53 | Jeremy223 | 2020 (Unspecified) D N 0 Unmonitored |
| 04:10.55 | Jeremy223 | 2021 (Unspecified) D N 0 Unmonitored |
| 04:10.57 | Jeremy223 | 2019 (Unspecified) D N 0 Unmonitored |
| 04:10.59 | Jeremy223 | 2 |
| 04:11.10 | jsaunders | Already tried asking in #asterisknow but now answer. :( Is there an easy cli network interface setup script or something for AsteriskNOW? |
| 04:11.22 | TJNII | I assume 2006,202, 2021, 2019, etc. don't work? |
| 04:11.35 | citats | what about extension 1213? thats the one you pasted the error with. does it have an IP in there? |
| 04:12.00 | Jeremy223 | that's mine, I haven't fired up eyebeam for over a week though, one sec looking |
| 04:12.24 | citats | does calling 1297, 1228, 1295, or 1270 work? |
| 04:12.49 | TJNII | How about 2006, 2020, 2021, 2019? Do those yeild your problem? |
| 04:12.53 | Jeremy223 | rings and no fast busy |
| 04:13.09 | TJNII | Jeremy223: Which ones? |
| 04:13.35 | Jeremy223 | 1297 works, trying others, the 2xxx ones do not have DID's so dialing menu to test those |
| 04:14.08 | Jeremy223 | 2006 fast busy, 1297 rings 1228 rings but never goes to voicemail |
| 04:14.34 | serpent-fly | hta, problem te410p card , work onli 1,4. other ports (2,3) all time get errors " PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" and pri restart... |
| 04:14.34 | Jeremy223 | my 1213 extension is "1213 (Unspecified) D N 0 Unmonitored |
| 04:14.56 | TJNII | Jeremy223: You said 2006 isn't in the dialplan. Is that supposed to work or is that something special? |
| 04:15.00 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4e70a4b71e74bf9f) |
| 04:15.23 | TJNII | Yea, your phones arn't connecting to the server. Thus the (Unspecified) under the IP fiels |
| 04:15.36 | TJNII | Check the network. |
| 04:15.43 | Jeremy223 | the 2xxx extensions are for our support guys that are just in queues and don't need people dialing them direct |
| 04:15.50 | [TK]D-Fender | serpent-fly, pastebin your "cat /proc/interrupts" |
| 04:15.51 | [TK]D-Fender | ~pb |
| 04:15.52 | jbot | A Pastebin is a place to paste your stuff 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 |
| 04:15.52 | TJNII | Do you do any centrailzed configing? |
| 04:15.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 04:16.14 | Jeremy223 | ah thanks been about 4 years since I've been on IRC ;-) |
| 04:16.46 | TJNII | Jeremy223: Do you do centralized configs for your phones? |
| 04:16.46 | Jeremy223 | we just have server mainly, just edit the config files manually usually |
| 04:17.07 | Jeremy223 | not sure what centralized configs means, everyone manually configures their sip info in Eyebeam |
| 04:17.29 | Jeremy223 | but even one of our queues is doing a fast busy x1232 |
| 04:17.58 | [TK]D-Fender | serpent-fly, ask your telco to run a check for frame slips/flips |
| 04:18.08 | serpent-fly | http://pastebin.com/m311fa903 |
| 04:18.17 | Jeremy223 | x1232 setup: http://pastebin.com/d20edd3f9 |
| 04:18.31 | [TK]D-Fender | serpent-fly, pastebin "dmesg" |
| 04:18.33 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:18.40 | Jeremy223 | the server can ping its default gateway and google etc, what network connectivity would it need if its our only asterisk server? |
| 04:19.41 | serpent-fly | http://pastebin.com/d5b4c41eb |
| 04:19.44 | circas | well youll prob need to be able to connect to your sip provider no? |
| 04:19.45 | TJNII | I assumed you were using hardphones that would always be connected. If you are using softphones I'm probably wrong. |
| 04:20.15 | serpent-fly | [TK]D-Fender, http://pastebin.com/d5b4c41eb |
| 04:20.27 | Jeremy223 | I'm not exactly sure what our setup is with our sip provider, how would our main line be working though and some extensions like 1200, but not others? |
| 04:20.35 | Jeremy223 | yeah a few hardphones but mostly soft |
| 04:21.05 | TJNII | And this just dies a few hours ago? Nobody messed with it? |
| 04:21.50 | Jeremy223 | TJNII: yep |
| 04:22.13 | Jeremy223 | sip show subscriptions and channels shows 0 subscriptions/channels, how can I see if our connectivity to the sip provider is ok? |
| 04:22.15 | TJNII | And voicemail fails too? It's not something simple like a full disk, is it? |
| 04:22.17 | [TK]D-Fender | serpent-fly, Intel(R) PRO/1000 Network Driver - version 7.3.20-k2 <-- this is a known compatability item with Digium cards & zaptel |
| 04:22.39 | [TK]D-Fender | serpent-fly, Try disabling the nic and removing its module completely. |
| 04:22.45 | TJNII | I may be way out in left field with that, but it is something to check. |
| 04:22.49 | Jeremy223 | TJNII: yeah voicemail never kicks in, and the server has like 60 gigs free, we have 24/7 monitoring on it so it didn't fill up earlier either |
| 04:22.56 | [TK]D-Fender | serpent-fly, Especaill as you're on a 1st gen TE series card |
| 04:23.31 | Jeremy223 | watching the console I do see a bunch of stuff like this, if that means anything? http://pastebin.com/m4507df7a |
| 04:23.43 | Jeremy223 | the stopping retransmission / auto destroying lines |
| 04:24.15 | circas | weird that it worked before thought |
| 04:25.05 | Jeremy223 | yep. we have had this fast busy issue come up maybe once in the past 6 months and a reboot fixed it then.. this is for a company of about 50 employees working 24/7 that use the phones heavily and no one reported any problems until a few hours ago |
| 04:25.42 | Jeremy223 | we did add a couple new employees last week but I restored backup configs from before then and it made no difference |
| 04:25.45 | serpent-fly | [TK]D-Fender, thx |
| 04:26.13 | circas | did you reload lol |
| 04:26.13 | Jeremy223 | if I can dial our main menu though and try dialing an extension from there and get the same fast busy though, could that be a SIP provider issue or would that rule it out? |
| 04:26.21 | Jeremy223 | hehe yes extensions / sip / voicemail.so or whatever |
| 04:26.43 | TJNII | Jeremy223: If it does it internally too your VoIPSP is probably fine |
| 04:26.53 | TJNII | Or not the root problem, anyways |
| 04:27.05 | circas | yeah thats true |
| 04:27.12 | TJNII | Jeremy223: Is there any pattern to the extensions that don'r work |
| 04:27.52 | Jeremy223 | no, its weird. i.e. 1216 works but 1217 doesn't and they are set up exactly the same as far as I can tell |
| 04:28.00 | Jeremy223 | its almost like we are getting DOS'd and all lines are full or something but I'm not sure how to tell. |
| 04:28.07 | Jeremy223 | except the same ones don't work repeatedly |
| 04:28.21 | Jeremy223 | one sec I'll pastebin some working/nonworking ones |
| 04:28.37 | Jeremy223 | but none of them go to voicemail anymore even the non-fast-busy |
| 04:28.52 | TJNII | Call one and pastebin everything that comes up on the console |
| 04:30.48 | Jeremy223 | 1216: fast busy http://pastebin.com/m588e7831 |
| 04:30.48 | Jeremy223 | 1217: rings but no voicemail, no fast busy though: http://pastebin.com/m57083b54 |
| 04:31.03 | Jeremy223 | ok calling 1216 one sec |
| 04:31.46 | Jeremy223 | 1216 console log http://pastebin.com/m36a4678c |
| 04:31.58 | *** part/#asterisk nickzxcv (i=nick@schmalenberger.us) |
| 04:32.10 | circas | I've solved some weird problems by looking at the sip debugging info! |
| 04:32.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:32.55 | Jeremy223 | 1217 console log (the one that rings and never goes to voicemail, but doesn't fast busy): http://pastebin.com/m5d333fd0 |
| 04:33.12 | Jeremy223 | ah also my boss said "we dont have a SIP provider. We have a PRI (ISDN Line) as our inbound telco, but once it gets to the phone system it is all controled by asterisk" |
| 04:33.41 | TJNII | Is 1216 a direct line? |
| 04:34.04 | Jeremy223 | yeah they both are, I can call our main menu and just dial those extensions from there though if that would shed any light |
| 04:36.20 | TJNII | Is whatever you're using as SIP/1216 on? |
| 04:36.39 | Jeremy223 | both those employees are gone for the day, one sec though I'll set the sip show peers for both |
| 04:36.42 | TJNII | Because I see "Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 04:37.00 | Jeremy223 | yeah actually 1216 is an employee who got fired so it hasn't been on a while ;-) his extension is still set up though |
| 04:37.14 | Jeremy223 | but still it should kick over to his voicemail I'd think since we haven't disabled it |
| 04:37.28 | TJNII | hmmm |
| 04:37.33 | Jeremy223 | is that message normal if their soft phone is offline though? |
| 04:38.00 | TJNII | I think so. |
| 04:38.11 | TJNII | Can you post your dial macro? |
| 04:38.29 | Jeremy223 | here is me reloading voicemail http://pastebin.com/m586144eb |
| 04:39.22 | Jeremy223 | looking for it, would it be [dial] to grep for? |
| 04:39.52 | TJNII | grep for [Macro-gsiexten] |
| 04:41.31 | Jeremy223 | here is it http://pastebin.com/m533c8cdc ... also some people use macro-stagingexten for optional call forwarding which is http://pastebin.com/mb6ab9e8 |
| 04:43.03 | Jeremy223 | stagingexten guys have fast busy too though it seems not just gsiexten |
| 04:46.01 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 04:46.45 | TJNII | hmmm... So 1216 fast busys with no voicemail, right? |
| 04:47.07 | Jeremy223 | ahh shit my boss remembered what he did to break it :-( evidently last weekend and no one noticed the problem until a few hours ago, argh |
| 04:47.15 | javb | i get this when trying to make a call using zap.. "Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) |
| 04:47.15 | javb | " |
| 04:47.29 | Jeremy223 | in extensions.conf we had "#include "extensions/*.txt" |
| 04:47.39 | Jeremy223 | well this weekend he removed the last *.txt file in there so that was not matching anything. |
| 04:47.48 | Jeremy223 | I commented that line out and now I'm getting voicemail again after reloading, phew |
| 04:47.52 | Jeremy223 | weird. |
| 04:48.08 | TJNII | Heh. |
| 04:48.09 | Jeremy223 | I restored backups of the configs we normally edit but not the entire /etc/asterisk folder |
| 04:48.13 | [TK]D-Fender | javb, "load chan_zap.so" |
| 04:48.36 | TJNII | Jeremy223: I had a feeling it was something like that. I was just about to ask you to bypass the macro as a test. |
| 04:49.11 | javb | now getting :1111 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 04:49.11 | TJNII | Well, on that note, I'm on call for work in 5 hours so I need to get off the computer and into bed. |
| 04:49.51 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 04:50.19 | [TK]D-Fender | javb, pastebin "cat /proc/interrupts" zapata.conf, zaptel.conf, "dmesg" |
| 04:50.57 | javb | ok, just a min |
| 04:53.12 | *** join/#asterisk javb (n=javb@190.80.234.104) |
| 04:53.27 | *** join/#asterisk webman (n=adamg@124.246.8.196.static.nexnet.net.au) |
| 04:53.48 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 04:54.17 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 04:54.19 | javb | Installed Asterisk 1.4, and CentOS 5, did on zaptel and asterisk make config as the last cmd.. but now, everytime i reboot, i get the card unconfigured on zttool, and if i use ztcfg, it gets configured right there.. |
| 04:55.00 | javb | any ideas? |
| 04:55.59 | javb | guys? |
| 04:56.16 | [TK]D-Fender | javb, "make config" install the startup scripts.. it doesn't ENABLE THEM. Thats up to you. |
| 04:56.42 | javb | Mmm, [TK]D-Fender, how do i do ENABLE THEM... |
| 04:57.25 | [TK]D-Fender | javb, welcome to RH 101. "chkconfig zaptel on", "chkconfig asterisk on". These should load up the two of them on all standard runlevels. |
| 04:59.27 | javb | [TK]D-Fender: Ok, so, make config, will create the scripts, chkconfig will enable them, and keep the config and loads them everytime ? |
| 05:00.30 | [TK]D-Fender | javb, chkconfig tells init what daemons to start and in what order at boot time. This means on next boot they will start in the appropriate order. |
| 05:00.52 | Jeremy223 | Ok one last question before bed: our /var/log/asterisk/event_log file is 0 bytes, is there any way to get the asterisk console log to save to a file normally? |
| 05:01.13 | [TK]D-Fender | javb, once the scripts are installed you can execute them manually as well "service zaptel [start|stop|etc.....]" and the same for * |
| 05:01.52 | javb | [TK]D-Fender: i understand, what u ment with "welcome to RH 101" is that this is new in latest verions of RH and derivants? |
| 05:02.41 | [TK]D-Fender | javb, No RH 101 is just a school joke that this is one of the basic things you should know as a Redhat Linux administrator. |
| 05:03.33 | javb | [TK]D-Fender: Hehehe, well, thanks for the joke, and explaining it too. |
| 05:03.57 | javb | I cant get zap calls now, with the SAME config files i was using on asterisk 1.2 |
| 05:04.11 | [TK]D-Fender | javb, No please provide that pastebin I requested. |
| 05:04.19 | javb | this is the output: http://dpaste.com/22600/ |
| 05:04.48 | citats | Jeremy223: look at logger.conf, by default theres an entry for messages with what to log there. anything besides console can be used and that will be the filename to log to |
| 05:04.52 | [TK]D-Fender | javb, not wat I asked for..... |
| 05:05.03 | [TK]D-Fender | <[TK]D-Fender> javb, pastebin "cat /proc/interrupts" zapata.conf, zaptel.conf, "dmesg" |
| 05:05.15 | [TK]D-Fender | javb, and "ztcfg -vvvv" as well. |
| 05:05.28 | javb | cat /proc/interrups ---> http://dpaste.com/22601/ |
| 05:06.22 | [TK]D-Fender | javb, 9: 287624 XT-PIC acpi, wctdm, Allegro <- your TDM400P is sharing an IRQ. this is automatically a BAD thing. But continue.... |
| 05:06.22 | javb | zapata.conf --> http://dpaste.com/22602/ |
| 05:07.22 | javb | zaptel --> http://dpaste.com/22603/ |
| 05:08.16 | javb | dmesg --> http://dpaste.com/22604/ |
| 05:09.07 | javb | ztcfg -vvvv ---http://dpaste.com/22605/ |
| 05:11.22 | [TK]D-Fender | javb, Look at your zapata : http://dpaste.com/22602/ : -- Executing [8092201212@power-ca:1] Dial("SIP/102-090bde78", "ZAP/g0/8092201212|30|") in new stack |
| 05:11.29 | *** join/#asterisk bantu (n=Miranda@p54A32B38.dip0.t-ipconnect.de) |
| 05:11.32 | [TK]D-Fender | javb, You have no group 0! |
| 05:11.49 | [TK]D-Fender | javb, go get some coffee |
| 05:12.28 | Jeremy223 | thanks for all your help everyone! http://img233.imageshack.us/img233/5599/kekekemouse7od9vf7lrfo4.jpg |
| 05:12.46 | *** join/#asterisk javb (n=javb@190.80.234.104) |
| 05:13.00 | javb | [TK]D-Fender: sorry, hot disconnected. |
| 05:13.03 | javb | got |
| 05:13.11 | [TK]D-Fender | <[TK]D-Fender> javb, You have no group 0! |
| 05:13.11 | [TK]D-Fender | <[TK]D-Fender> javb, go get some coffee |
| 05:13.22 | [TK]D-Fender | <[TK]D-Fender> javb, Look at your zapata : http://dpaste.com/22602/ : -- Executing [8092201212@power-ca:1] Dial("SIP/102-090bde78", "ZAP/g0/8092201212|30|") in new stack |
| 05:13.31 | javb | :( |
| 05:13.47 | javb | didnt find the red bull i told later |
| 05:13.50 | javb | earlier |
| 05:14.10 | [TK]D-Fender | javb, See... you are clearly suffering from caffeine withdrawl |
| 05:15.10 | javb | how can we solve the IRQ sharing stuff? |
| 05:17.22 | [TK]D-Fender | javb, Check your BIOS and see if you can resever an IRQ for the slot its in. If no start shiffting it around |
| 05:18.04 | javb | Ok. Let me try that. |
| 05:18.25 | javb | Another thing is, in asterisk 1.2 i used to have mpg123 for mp3 moh... |
| 05:18.49 | javb | now i dont have mpg123, but installed moh in gsm format, in the menuselect of asterisk.. |
| 05:18.54 | javb | but, DONT have MOH |
| 05:19.44 | [TK]D-Fender | javb, go check your modes, and your files. |
| 05:19.53 | [TK]D-Fender | javb, but its late I've got to get some sleep |
| 05:19.59 | [TK]D-Fender | GL all. |
| 05:20.03 | javb | i het this warning: http://dpaste.com/22606/ |
| 05:20.05 | javb | : / |
| 05:22.01 | citats | javb: pastebin your musiconhold.conf |
| 05:23.13 | javb | citats: http://dpaste.com/22607/ |
| 05:24.13 | citats | i assume your just using the default moh class in whatever your calls are in? (zaptel/sip/etc?) |
| 05:25.09 | citats | if you just used the menuselect to pick gsm files change your default entry to: |
| 05:25.12 | citats | [default] |
| 05:25.14 | citats | mode=files |
| 05:25.19 | citats | directory=/var/lib/asterisk/moh |
| 05:25.29 | citats | that should be it |
| 05:25.59 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 05:26.56 | javb | let me see |
| 05:29.20 | serpent-fly | hta, problem te410p card , work onli 1,4. other ports (2,3) all time get errors " PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" and pri restart... im didsable my intel ethernet cards in bios and in kernel is dot work :( any idies? |
| 05:29.42 | *** join/#asterisk PepOSX (n=pepOSX@190.78.220.149) |
| 05:30.11 | javb | citats: same problem |
| 05:30.23 | citats | javb: did you restart? |
| 05:30.30 | javb | yes |
| 05:30.50 | citats | javb: the error message shouldnt be the same, since its trying to use files instead of mpg123 |
| 05:31.11 | javb | wait wait |
| 05:33.50 | javb | citats: here is .. http://dpaste.com/22609/ |
| 05:33.57 | javb | Same, output on the cli |
| 05:34.38 | citats | i doubt its compeltely the same output sinc e you've changed the directory. you still have [default] up there? |
| 05:35.17 | citats | did you verify the files are under /var/lib/asterisk/moh ? |
| 05:35.46 | javb | the files are there |
| 05:35.57 | javb | and the output is play / stop inmidiately |
| 05:36.55 | javb | this is the output: http://dpaste.com/22610/ |
| 05:37.32 | hmmhesays | bah that episode of earth final conflict was too good, now I have to watch another |
| 05:38.03 | citats | javb: how about pastebin'g the output of 'moh show classes' and 'moh show files' |
| 05:39.10 | javb | should i restart the system, or just asterisk? |
| 05:39.16 | citats | javb: just asterisk |
| 05:39.46 | javb | this is the out put http://dpaste.com/22611/ |
| 05:39.52 | javb | Seems nothing has changed |
| 05:40.34 | javb | dont understand now. |
| 05:40.53 | citats | javb: and you definitely editted musiconhold.conf? |
| 05:41.00 | citats | and restarted asterisk afterwards? |
| 05:41.52 | javb | OF COURSE |
| 05:41.55 | javb | :/ |
| 05:42.15 | javb | http://dpaste.com/22612/ |
| 05:42.16 | *** join/#asterisk ugenka (n=ugenka@86.57.151.154) |
| 05:43.31 | javb | look at this http://dpaste.com/22614/ |
| 05:44.08 | citats | javb: are you just doing a moh reload or have you tried a complete restart of asterisk? |
| 05:44.15 | javb | both |
| 05:44.24 | javb | now, i`m trying a complete system restart |
| 05:46.53 | citats | as far as asterisk is concerned there is no difference between restarting asterisk and a complete system restart |
| 05:47.31 | javb | citats: well, dont know if is the lack of coffee, but, now, after restarting system, evrything PERFECT :/ |
| 05:48.00 | citats | javb: and you did a 'restart now' on asterisk before? and not just a reload? |
| 05:48.35 | javb | reload.. |
| 05:48.50 | citats | javb: reload != restart |
| 05:49.12 | citats | <citats> and restarted asterisk afterwards? |
| 05:49.12 | citats | <javb> OF COURSE |
| 05:49.15 | citats | not true :) |
| 05:49.19 | javb | citats: :/.. new to asterisk, learning. Sorry. |
| 05:50.11 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:50.28 | citats | a restart is basically shutting down asterisk and restarting, so that will drop any active calls. a reload just reloads config files up to the point it can do it safely (ie not everything will be able to change with a reload) |
| 05:55.57 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net) |
| 05:56.41 | javb | citats: thanks. Have another question. |
| 05:57.22 | javb | Calling one exten, putting it on hold, an trying to make a new call, i get an weird error: "[Oct 16 01:55:38] NOTICE[2242]: chan_sip.c:13605 handle_request_invite: Failed to authenticate user "Joel Valdez" <sip:102@10.0.0.55>;tag=8B4B7F1A-B15200D9 |
| 05:57.22 | javb | " |
| 06:00.19 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:06.39 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 06:06.57 | adeel | is it possible to do group pickup on a zap channel from a sip phone? |
| 06:08.08 | hmmhesays | assuming you are ringing sip phones from the incoming zap call |
| 06:08.52 | adeel | well the call comes to one polycom phone (receptionist) and then i decide to pick it up directly from my phone instead of waiting for it transfer |
| 06:09.02 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:09.05 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:09.10 | hmmhesays | yes |
| 06:09.12 | hmmhesays | you can do that |
| 06:09.22 | adeel | through *8? |
| 06:09.29 | hmmhesays | through whatever you have configured |
| 06:09.45 | adeel | well i haven't been able to find anything on how to do it, cross technology |
| 06:10.25 | hmmhesays | put them in the same pickup group or whatever its called and make sure you have your features.conf configured properly |
| 06:11.04 | adeel | hmmhesays, interesting, i'll have to check it out |
| 06:22.49 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 06:23.11 | serpent-fly | hta, проблема Ñ ÐºÐ°Ñ€Ñ‚Ð¾Ð¹ te410p когда поток подключаетьÑÑ Ðº 1 или 4 каналу вÑе хорошо работет а к 2,3 поÑтоÑнно вбраÑывет ошику в конÑоль PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 и поток падает |
| 06:23.28 | JT | serpent-fly: can you please not do that |
| 06:23.50 | serpent-fly | sory |
| 06:25.34 | *** join/#asterisk af_ (n=getsmart@81-174-44-210.dynamic.ngi.it) |
| 06:26.35 | *** join/#asterisk neax (n=newdle@203-114-176-86.dsl.sta.inspire.net.nz) |
| 06:29.17 | hmmhesays | oh I get my sangoma a200 tomorrow I hope all goes well |
| 06:30.33 | hmmhesays | anyone run this with zaptel 1.4? |
| 06:31.53 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 06:34.09 | AJaymn | Anyone use ISDN BRI? |
| 06:37.47 | hmmhesays | crazy kids and their digital lines |
| 06:38.10 | AJaymn | :P |
| 06:38.20 | AJaymn | can you spoof caller id on a BRI? |
| 06:38.41 | hmmhesays | likely |
| 06:39.34 | AJaymn | cant find anything that says yes or no.. only on PRI (t1) |
| 06:39.56 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 06:39.58 | hmmhesays | could always try it? |
| 06:40.15 | hmmhesays | I know i'm not any help |
| 06:40.26 | AJaymn | lol |
| 06:40.42 | AJaymn | <PROTECTED> |
| 06:42.39 | citats | AJaymn: i've used BRI in the past and it was never possible to set callerid other than one of the 2 numbers assigned. of course it could be different depending on the telco |
| 06:43.27 | JT | it completely depends on the telco |
| 06:43.32 | hmmhesays | centos 5 installed and running sweetly |
| 06:43.44 | JT | telcos in north america let you set whatever rubbish you want with digital service |
| 06:43.48 | JT | generally |
| 06:44.08 | JT | AJaymn: a BRI is just a cutdown PRI |
| 06:44.16 | JT | signalling is the same sort of stuff |
| 06:44.42 | citats | JT: i've had BRI service from maybe 3-5 different LECs and none of them supported it, but on every PRI (except for one) it was possible |
| 06:46.37 | *** join/#asterisk Alex465 (n=Miranda@proxy.izhnt.ru) |
| 06:47.16 | AJaymn | What could I ask the telco? |
| 06:49.22 | *** join/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
| 06:50.12 | citats | AJaymn: i suspect the only people at the telco that would know anything about it would be the switch techs. |
| 06:50.27 | JT | citats: maybe the BRIs are setup properly :P |
| 06:50.54 | citats | AJaymn: i suppose you could ask them something like, if i had multiple BRIs could I send calls out any of the b channels with a callerid from any of the others |
| 06:51.10 | AJaymn | that works! :) |
| 06:51.27 | JT | yeah but not really |
| 06:51.40 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 06:51.51 | JT | since they're under the one account |
| 06:52.55 | AJaymn | i get my DIDs from my Voip carrier, but have alot of "local" calls and could use ISDN lines for local calls.. |
| 06:53.04 | AJaymn | but need caller id to be able to be set to any DID we have. |
| 06:54.19 | Alex465 | how in asterisk to register the client using database |
| 06:56.25 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 07:04.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:06.48 | Zipper_32 | What's the best way to have certain groups of phones use set groups of outbound lines to dial out? I have 3 groups using the same asterisk box, and they're all in the same context, but their analog lines have CID set by the telco, and I want them to use their telco designated lines. |
| 07:09.38 | orakle | put them in different contexts (include whatever internal extensions you have in each one), then write a different dial command for each context depending on which telco you're using |
| 07:10.01 | orakle | should work even if it's the same telco, just specify the extension that you've put it under in sip.conf |
| 07:11.13 | orakle | you might have to play with which context the telco accounts are in, i'm not sure |
| 07:16.09 | Zipper_32 | orakle: Thanks. |
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| 07:38.18 | orakle | does it work? |
| 07:43.02 | Zipper_32 | still working on it. Trying to make things a bit cleaner in my dialplan |
| 07:47.30 | *** part/#asterisk kodomo (n=memyself@emu.net.informatik.tu-muenchen.de) |
| 07:55.46 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
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| 08:02.14 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 08:02.19 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 08:13.17 | syle | anyone run asterisk on freebsd |
| 08:18.02 | Uatec | hi |
| 08:18.15 | Uatec | i'm getting loads of lines, similar to this: Oct 16 09:20:08 DEBUG[23049]: chan_sip.c:1411 __sip_ack: Stopping retransmission on '0f3bec5646b7681e7c246dc46c0126d1@192.168.232.52' of Request 102: Match Found |
| 08:18.19 | Uatec | in my CLI |
| 08:18.54 | Uatec | i don't care, frankly. How can i turn them off? I have console => notice,warning,error but they're still coming |
| 08:22.24 | Uatec | oh |
| 08:22.26 | Uatec | i've reloaded |
| 08:22.30 | Uatec | and restarted the cli |
| 08:22.58 | Uatec | but logger show channels still shows the console channel as notice,warning,error,debug,verbose |
| 08:24.22 | agx | Uatec remove debug from the console line OR modify the source to add if(option_debug>value) before or it OR comment it out |
| 08:27.18 | *** join/#asterisk gardo (n=gardo@121.97.251.62) |
| 08:28.48 | Uatec | i have removed the debug from the console line |
| 08:28.54 | Uatec | but it's not reloading the logger.conf |
| 08:30.43 | Uatec | ah |
| 08:30.48 | Uatec | a simple reload didn't work |
| 08:30.52 | Uatec | i had to do "reload logger" |
| 08:31.07 | *** join/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net) |
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| 08:52.18 | agx | Uatec, "reload" is an evil command :) |
| 08:52.55 | knarfly | module reload |
| 08:53.23 | knarfly | stop now is an evil command and my girlfriend keeps giving it to me... 8-) |
| 08:54.09 | syle | anyone run asterisk on freebsd? |
| 08:54.43 | knarfly | yep....but you already know that about me.... |
| 08:56.19 | *** join/#asterisk Zenton (n=vicente@212.166.192.134) |
| 08:57.05 | Uatec | agx, why? |
| 09:00.44 | Zenton | hi all |
| 09:00.56 | Zenton | is there any GUI available for asterisk? |
| 09:05.18 | Zenton | ops sorry |
| 09:10.35 | Uatec | do SIP and RTP use both TCP AND UDP or just the one? |
| 09:10.39 | Uatec | just UDP? |
| 09:14.29 | JT | just UDP |
| 09:14.43 | JT | the sip protocol can use tcp, but asterisk does not support it |
| 09:18.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:19.30 | *** join/#asterisk Pon`work (n=jamesm@ip-217.146.113.66.merula.net) |
| 09:21.10 | Uatec | ok |
| 09:21.13 | Pon`work | Is there anyway, even hacky, to after a caller hangsup to send the callee somewhere else? |
| 09:22.15 | JT | you could call them back, but i think that's about it |
| 09:23.46 | JT | rtp can never be over tcp btw, Uatec |
| 09:23.58 | Pon`work | I was thinking maybe doing something with a conferance, but not sure if it'll work or not |
| 09:27.02 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:31.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 09:32.43 | L|NUX | Hello every one |
| 09:32.57 | L|NUX | does any one used South Korea E1 ? |
| 09:35.33 | Uatec | JT, ok, ty |
| 09:35.33 | *** join/#asterisk syneus (n=syneus@host23-25-dynamic.180-80-r.retail.telecomitalia.it) |
| 09:35.58 | Uatec | when i turn on SIP debug, one of the lines i get looks like this: Retransmitting #3 (NAT) to XXX.XXX.XXX.XXX:2051: |
| 09:36.03 | Uatec | what does the 2051 mean? |
| 09:36.05 | Uatec | is it the port? |
| 09:36.23 | Uatec | becuase i was under the impression that sip used port 5060 and rtp use 10000-20000 |
| 09:36.26 | neax | Pon`work: just so I understand what you're asking; when person A has called person B, said what they wanted to say, and person A has hung up, you want person B to be redirected somewhere |
| 09:36.54 | neax | Pon`work: I'm guessing person A is using an asterisk extension? |
| 09:37.40 | Uatec | damn, the peer is using port 2051 |
| 09:37.46 | Uatec | why would it do that? |
| 09:39.00 | *** join/#asterisk kannan (n=kannan@121.246.26.150) |
| 09:41.12 | JT | Uatec: do you understand TCP/IP? |
| 09:41.14 | kannan | i am having trouble dialling on a E-1 line, it says "unable to ceate channel of type ZAP" in the * CLI. I have a digium sngle span card |
| 09:42.02 | kannan | zap status shows alarms blu/yell/red |
| 09:42.15 | JT | sip uses 5060 normally, and rtp can use any range, the default in asterisk is 10000 to 20000 udp |
| 09:45.36 | neax | cat rtp.conf | grep rtpstart |
| 09:46.20 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:47.11 | *** join/#asterisk sysadmin-leb (n=test@mail.splendor.net) |
| 09:47.15 | sysadmin-leb | hi all |
| 09:47.31 | sysadmin-leb | I have noticed that digium do provide codecs for around 10$ |
| 09:47.40 | sysadmin-leb | however I have also found open source binaries for the same codec |
| 09:47.51 | sysadmin-leb | can anyone tell me the difference between the two please ? |
| 09:48.09 | JT | yes, one is illegal to use, one is not |
| 09:48.16 | JT | also, one is maintained |
| 09:48.42 | sysadmin-leb | so the open source one is illegal to use ..? |
| 09:48.48 | neax | one also costs around $10 |
| 09:48.50 | neax | :) |
| 09:49.12 | defswork | sysadmin-leb: all open source is illegal - source S. Ballmer |
| 09:49.24 | sysadmin-leb | defswork ... yeah yeah :p |
| 09:49.34 | neax | open source is communism. pure and simple. |
| 09:49.53 | JT | sysadmin-leb: not paying patent royalties is illegal |
| 09:50.19 | sysadmin-leb | well I need a pro solution |
| 09:50.25 | sysadmin-leb | we have a class 5 swith |
| 09:50.44 | sysadmin-leb | through which we need to route calls done through soft voip phones on our LAN |
| 09:50.51 | JT | paying $10 is obviously more professional |
| 09:50.53 | sysadmin-leb | the LAN also has a trixbox |
| 09:51.14 | sysadmin-leb | all voip phones register to the trixbox |
| 09:51.25 | JT | sort of the wrong channel |
| 09:52.14 | Pon`work | neax: person A is from PSTN, person B is SIP |
| 09:52.33 | sysadmin-leb | and the trixbox in trun has a trunk to the C5 switch sends them to a carrier using codecs 729 and 723 |
| 09:52.50 | sysadmin-leb | can I conclude from the above that the free versions will not allow me to do that ..? |
| 09:53.24 | JT | it can transcode G.729 with appropriate licensing and cpu power |
| 09:53.34 | JT | asterisk cannot transcode G.723 |
| 09:53.40 | JT | no codecs available |
| 09:53.54 | sysadmin-leb | not even commerically ? |
| 09:54.12 | JT | no. |
| 09:54.15 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 09:54.26 | JT | well there's a TC400B card, but that's relatively untested |
| 09:54.35 | JT | what's the attraction? G.723 audio sounds like trash |
| 09:54.42 | sysadmin-leb | so you suggest purchasing the 729 codec ? |
| 09:54.43 | agx | JT, indeed! :) |
| 09:55.18 | JT | yes |
| 09:56.12 | sysadmin-leb | but even if I wanted to test the open source version it wont work because the carrier wont accept it ? because it is unlicensed ? |
| 09:56.32 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 09:56.52 | JT | you could try |
| 09:56.58 | JT | but seriously |
| 09:56.59 | JT | $10 |
| 09:57.27 | sysadmin-leb | it is not about the money I have a scenario that I need to test first if it works I will purchase all necessary licenses |
| 09:58.07 | neax | if it were myself implementing this, and it was a commercial solution (not just my mucking-around-at-home system), I wouldn't steer to the wrong side of the law |
| 09:58.20 | JT | it's clearly about the money |
| 09:58.23 | JT | it's 10 bucks. |
| 09:58.51 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 09:58.57 | sysadmin-leb | well guys thanks for your help |
| 09:59.02 | sysadmin-leb | have an nice day |
| 09:59.51 | b1ch0 | hi i nned to handle from 5 to 8 external lines (fxo ports), wich is better, an internal PCI card with fxo modules, or an external gateway (ATA) like the AudioCodes ones (or from any other vendor) ? |
| 10:00.04 | Uatec | JT, yes, i understand TCP/IP. The point is. Why would one phone randomly decide to use port 2051 (yes, udp) for SIP, instead of port 5060 like every other sip device i've ever heard of |
| 10:01.29 | JT | Uatec: because your server is on 5060. |
| 10:03.33 | penguinFunk | b1ch0: stay well away from audicodes |
| 10:03.56 | penguinFunk | the quality is hopeless and they are a nightmare to configure |
| 10:04.28 | penguinFunk | your much better off going for a sangoma/digium card and using asterisk |
| 10:04.55 | penguinFunk | make sure to get hardware echo cancelling if you can afford it |
| 10:05.08 | agx | b1ch0 OR if you find on ebay a channelbank you can use a PRI so you can expand it in future |
| 10:05.31 | JT | channel banks don't work with PRIs |
| 10:08.47 | agx | JT, uhm that's new.... |
| 10:09.15 | JT | agx: have you used a channel bank? |
| 10:09.28 | b1ch0 | ok, thanks ... i have already an asterisk box running with TDM400P, but i was just asking to experts or someone that have already did it before wich is the diference between those 2 ways to interconnect * to tradicional pstn lines |
| 10:09.53 | Uatec | JT, my server is on 5060, all my phones are on 5060 |
| 10:10.00 | Uatec | except this single snom 190, which is on 2051 |
| 10:10.55 | b1ch0 | <PROTECTED> |
| 10:11.20 | JT | Uatec: symmetrical port usage as you've described is actually very rare in the tcp/ip world |
| 10:11.43 | JT | usually the client port is a random high numbered port, and the server port is the one that matters |
| 10:15.11 | Uatec | it's not the client port |
| 10:15.14 | Uatec | it's the server port |
| 10:15.21 | Uatec | i'm trying to route data through a firewall |
| 10:15.28 | Uatec | but if the port is something random, that gets rather hard |
| 10:16.11 | b1ch0 | here i have tested an ATA (similar to Sipura 3000), but it is not a good solution, call the ata extension to recently have external line tone .... you cant register in the cdr the real external line called .... the better way i know until now is 9|. in the external trunk in * .... so this is why i was asking for if there are good ata that work similar ton TDM PCI cards |
| 10:16.18 | JT | Uatec: that's strange, you said it was from a sip debug that you saw that |
| 10:16.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:21.19 | agx | Uatec, what are you going to do? give access to the PBX to a phone outside the office or trying to use a VOIP provider account? |
| 10:24.10 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:25.27 | b1ch0 | well, any good tip over my "problem" ? |
| 10:25.53 | b1ch0 | does anyone had experience on that before? |
| 10:26.28 | stmaher | Hello everyone.. I need some help with a transfer problem.. I have a third party external IVR that I want to link with an asterisk box, cisco VOIP gateway and a sip phone.. |
| 10:26.57 | stmaher | Call flow.. Xlite -> * -> IVR -> * -> CiscoVG |
| 10:27.19 | stmaher | When I try getting the IVR to do a blind transfer i get a 603 declined. |
| 10:27.53 | stmaher | No wait.. Think i just fixed it |
| 10:30.53 | DRTHM | kannan: run zttool, ztcfg -vv check the errors you get |
| 10:31.11 | agx | b1ch0, i never found a decent FXO ATA so far |
| 10:38.05 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 10:41.21 | *** join/#asterisk syneus (n=syneus@host23-25-dynamic.180-80-r.retail.telecomitalia.it) |
| 10:41.52 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.143.224) |
| 10:41.56 | CBU[^_^]M`` | hello |
| 10:42.36 | CBU[^_^]M`` | i am behind a router... do i only need to forward port 5060 to my asterisk pbx? |
| 10:42.45 | JT | to do what |
| 10:42.51 | JT | to connect to a sip server outside |
| 10:43.01 | JT | or to act as a sip server to people on the Internet? |
| 10:43.21 | CBU[^_^]M`` | connect my someone via internet to my asterisk server |
| 10:43.34 | CBU[^_^]M`` | <JT> or to act as a sip server to people on the Internet? <= this one :) |
| 10:43.59 | JT | well you'll need to forward rtp as well |
| 10:44.23 | CBU[^_^]M`` | hmm |
| 10:44.28 | CBU[^_^]M`` | how do i do that? |
| 10:44.45 | CBU[^_^]M`` | what port number do i need to forward? |
| 10:45.34 | JT | 10000 to 20000 udp by default |
| 10:49.07 | *** join/#asterisk i3inary (i=i3inary@ip72-207-113-253.sd.sd.cox.net) |
| 10:50.01 | ai-a | CBU[^_^]M``: If you are only doing this for yourself and work colleague's then it might be more secure to use vpn. |
| 10:50.14 | i3inary | wow if anyone wants to see why video cards were put on this planet go try out sabayon live cd |
| 10:51.05 | ai-a | i3inary: gfx card's are for games. |
| 10:52.03 | i3inary | lol no sir...games should not cost you more than your computers processor to play....there is no reason you shouldnt benefit from a 3d card all the time |
| 10:52.21 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 10:52.35 | ai-a | does 3d gfx increase your productivity ? |
| 10:52.42 | i3inary | fuck yes |
| 10:52.47 | penguinFunk | lol |
| 10:53.15 | ai-a | unless your drawing, 3d gfx is not going to write your c code faster. |
| 10:53.25 | ai-a | what language required 3d rendering ? |
| 10:53.38 | stmaher | Hello everyone.. I have a problem with Consultation transfers.. Blind transfers work perfectly.. The error that is shown on the CLI # |
| 10:53.38 | stmaher | Oct 16 13:03:25 NOTICE[12737]: chan_sip.c:6932 get_refer_info: Supervised transfer requested, but unable to find callid '2a9da68-0-13c4-372238-6e04d88e-372238@10.0.0.204'. Both legs must reside on Asterisk box to transfer at this time. |
| 10:53.49 | orakle | it depends what you do |
| 10:53.52 | i3inary | heh..then dont use it...you wont hurt my feelings party pooper |
| 10:53.53 | stmaher | Could someone please explain this to me |
| 10:54.00 | orakle | if you're a 3d designer then it helps your productivity to have a fast video card |
| 10:54.19 | orakle | like CATIA for example is an amazingly powerful 3d app, but it needs a beastly computer otherwise it doesn't run that fast |
| 10:54.31 | stmaher | my call flow is Xlite -> * -> IVR -> * -> CiscoVG |
| 10:54.32 | i3inary | since when does the virtual world you work in have to be 3d...thats just stale thinking ai-a |
| 10:54.59 | ai-a | i3inary: do you have a 'wow' video of this sabayon ? |
| 10:55.08 | i3inary | kramer said it best ~"level jerry levels" |
| 10:55.23 | ai-a | as sabayonlinux.org is the worse website ive seen since the 80s. |
| 10:55.26 | stmaher | My full config with CLI debug output http://pastebin.com/d49db8521 thanks |
| 10:55.38 | i3inary | there are plenty of wow videos on the boobtube |
| 10:56.00 | i3inary | i tried it for myself...and i can tell you that you dont know what your talking about |
| 10:56.13 | i3inary | but you keep thinking like that and ill keep thinking like i do |
| 10:56.16 | ai-a | I have never used it. I never claimed i did. |
| 10:56.35 | i3inary | you think you cant benefit from it...cause you write C |
| 10:56.45 | ai-a | All im saying is how does the gfx card increase your brain / typing / designing? |
| 10:57.32 | ai-a | and as you seem to know 'boobtube' i can imagine your brain decreasing. |
| 10:57.42 | b1ch0 | hey can anyone kick 131nary please ?, this is a serious asterisk channel, no zabbaione or 3dfx shit ... please |
| 10:57.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 10:58.29 | i3inary | i just made a comment and ai-a wanted to spit his dutch about it.. |
| 10:58.38 | i3inary | i dont need to say anything else about it |
| 11:00.00 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 11:00.00 | i3inary | since when does a lower case i look like a 1 |
| 11:03.11 | k31th | Guys, I have my pbx setup like this --- Internet -> NAT -> PBX (on a public IP) the NAT router also routes public IP and private 192.168.2.* range. |
| 11:04.49 | orakle | so the PBX has a routable ip? |
| 11:05.50 | k31th | orakle: yeah |
| 11:05.56 | k31th | you could ping it directly from there |
| 11:06.20 | orakle | okay |
| 11:06.41 | orakle | so your gateway is acting as a nat for the people on 192.168.2.x, but a regular router for the public ips |
| 11:06.49 | orakle | i understand. what's wrong? |
| 11:07.20 | k31th | ahh |
| 11:07.52 | k31th | of corse if i ping it from a private ip the router wont route it out and back in again. |
| 11:08.06 | k31th | so i wont have to deal with nat hell ofr portfowarding |
| 11:08.24 | k31th | does each handset have to have a different sip port ? |
| 11:08.30 | k31th | or all on 5060 ? |
| 11:08.34 | orakle | usually 5060 |
| 11:08.39 | orakle | lets take this to pm |
| 11:09.36 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 11:10.50 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.150) |
| 11:11.02 | CBU[^_^]M`` | JT.. is still now working :( |
| 11:12.39 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.57) |
| 11:14.37 | JT | ~sipnat |
| 11:14.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 11:18.35 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
| 11:20.39 | *** join/#asterisk Whisk (n=whisk@82-44-47-95.cable.ubr04.croy.blueyonder.co.uk) |
| 11:29.41 | *** join/#asterisk mLx (n=mlx@217.151.231.18) |
| 11:29.51 | mLx | Hello. Does anybody can help me with call transfer, please? |
| 11:31.25 | mLx | ????? |
| 11:32.40 | stmaher | Hi guys.. is there anyway in asterisk that in the event of a busy or missing SIP end that the user isnt waiting 10 seconds for a engaged tone? |
| 11:33.16 | Uatec | hey, i have two groups of phones |
| 11:33.26 | Uatec | sip phones on the lan, which connect to the internal IP |
| 11:33.33 | JT | stmaher: qualify=yes |
| 11:33.42 | Uatec | and sip phones over the internet, which connect to the external IP |
| 11:34.02 | Uatec | i've taken a sip trace on the phone itself |
| 11:34.11 | Uatec | and the SIP packets coming from asterisk contain the internal IP |
| 11:34.19 | mLx | I see that nobody doesn't know about call transferring |
| 11:34.44 | Uatec | mLx, nobody Doesn't know about call transferring? surely that means everybody does? |
| 11:34.54 | *** join/#asterisk gardo (n=gardo@121.97.251.62) |
| 11:35.13 | Uatec | how can i persuade asterisk to send a different IP to the different sip phones? |
| 11:35.19 | stmaher | JT, thanks :-). |
| 11:35.29 | *** join/#asterisk basty (n=basty123@212.218.65.236) |
| 11:35.31 | basty | Hi |
| 11:36.06 | basty | anyone familar with connecting a nortel cs1k with asterisk pbx ? (over SIP) |
| 11:36.12 | stmaher | JT, Herm.. that didnt work. |
| 11:36.21 | JT | Uatec: what's wrong with the ip? |
| 11:36.25 | JT | stmaher: what did you do? |
| 11:36.52 | stmaher | added qualify to the relavent sip phone in sip.conf ? |
| 11:37.25 | JT | then what did you do? |
| 11:37.33 | stmaher | restarted asterisk |
| 11:37.36 | stmaher | made a test call. . |
| 11:37.41 | Alex465 | HELP asterisk + h323 |
| 11:37.42 | stmaher | still takes 10-30 seconds |
| 11:37.52 | JT | and this sip phone was the one that's offline? |
| 11:37.56 | stmaher | im trying to get it down to two |
| 11:38.02 | stmaher | Yep |
| 11:38.14 | JT | Alex465: that's got to be the laziest help request i've ever seen |
| 11:38.18 | stmaher | there is nothing listening on 5060 on the IP of the sip phone |
| 11:38.37 | JT | try setting it in the general section, stmaher |
| 11:38.46 | stmaher | OK.. thanks JT |
| 11:39.04 | JT | and waiting half a minute after reloading sip |
| 11:39.07 | JT | or more |
| 11:39.27 | stmaher | I kill the asterisk server completely and restart it |
| 11:39.33 | JT | pointless really |
| 11:39.36 | JT | sip reload is fine |
| 11:39.42 | stmaher | mkay :-) |
| 11:39.52 | Alex465 | with AquaGK on h323 goes bell on àñòåðèñê and occurs unset |
| 11:40.08 | Alex465 | on asterisk |
| 11:40.28 | JT | Alex465: no-one's going to be able to understand that |
| 11:40.50 | Alex465 | ))) |
| 11:40.54 | Alex465 | mmm |
| 11:41.44 | Alex465 | who speaks in russian |
| 11:42.36 | stmaher | jt,WARNING[12948]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'default' |
| 11:42.37 | stmaher | Scheduling destruction of call 'MWUxMDEwNDRjOGZhMDlmNDMwYjU2NGU0ZWI0ZjBjMTk.' in 32000 ms |
| 11:43.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:43.14 | stmaher | how can i get that destruction down to 2000ms |
| 11:44.19 | JT | make a t extension |
| 11:44.49 | Uatec | JT, what's wrong? well you can't access the internal IP from outside, and there's no point accessing the external IP from inside |
| 11:45.14 | Uatec | but i found the externip and localnet variables in sip.conf |
| 11:47.08 | Alex465 | 2stmaher set in extension.conf exten => t, prority, action |
| 11:47.28 | JT | Uatec: yes, as stated in |
| 11:47.30 | JT | ~sipnat |
| 11:47.30 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 11:47.33 | stmaher | exten => t,4,Busy(10) |
| 11:47.54 | stmaher | Alex465, didnt work.. still ~10-30 seconds |
| 11:48.01 | JT | stmaher: well of course it didn't |
| 11:48.12 | JT | all extensions must start at 1 unless you're priority jumping |
| 11:48.45 | stmaher | changed it to 1 |
| 11:48.45 | Alex465 | 2stmaher show text |
| 11:49.07 | Alex465 | from context 'default' |
| 11:50.01 | stmaher | Alex465, http://pastebin.com/d397cff24 |
| 11:52.41 | *** join/#asterisk dijungal (n=kdaniel@205.244.148.37) |
| 11:53.04 | dijungal | hello... how do i move a recording after the call has completed? |
| 11:53.13 | Alex465 | 2stmaher this exactly used code |
| 11:53.21 | stmaher | yes |
| 11:55.13 | Alex465 | 2stmaher try use timeout in "dial(sip,TIMEOUT,opt) |
| 11:55.26 | dijungal | timeout? |
| 11:56.54 | stmaher | Alex465, Ill give that a go.. Does asterisk have any built in Load Balancing Capabilities or Failover without external tools? |
| 11:57.11 | stmaher | So i have 2 external IVRS.. |
| 11:57.19 | stmaher | Can asterisk load between the two and possibly failover |
| 11:58.14 | *** join/#asterisk guillote_GNU (n=bancaria@host9.201-253-17.telecom.net.ar) |
| 11:59.04 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:59.04 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:01.38 | Alex465 | 2stmaher what meaning exten => t,1,Busy(10) in your context "default" , delete it line ))) |
| 12:02.19 | blitzrage | 't' is a built in extension for a timeout |
| 12:02.28 | blitzrage | set with the TIMEOUT() dialplan function |
| 12:02.48 | stmaher | blitzrage, can you be a little more verbose please? |
| 12:03.19 | blitzrage | 'show function TIMEOUT' on the asterisk console, and it should make sense |
| 12:04.12 | blitzrage | most commonly used in an auto-attendant situation |
| 12:04.30 | blitzrage | I just woke up 2 mins ago... so I'm not very verbose yet :) |
| 12:04.38 | stmaher | blitzrage, LOL :-) |
| 12:05.26 | slima | hi, I have a problem with TIMEOUT(digit) when I typing '100' asterisk dont wait for and search a '1' exten my config: http://pastebin.com/f33d1f924 (sorry for my english) |
| 12:05.44 | slima | whats wrong? |
| 12:06.15 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:06.15 | blitzrage | slima: you need WaitExten() after Background() |
| 12:06.32 | blitzrage | looks like you've got autofallthrough=yes turned on at the top of the file (on by default in 1.4) |
| 12:06.39 | blitzrage | which is a good thing |
| 12:07.58 | slima | hm, thx i try |
| 12:09.55 | sysadmin-leb | Hey All I am trying to create an inbound route for asterisk I have assigned the DID, and the call is reaching asterisk but I am getting an error "Proxy Authentication Required" |
| 12:10.47 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.232.254) |
| 12:12.47 | blitzrage | sysadmin-leb: that is normal. With SIP, you get the INVITE coming in, then Asterisk sense a 407 Proxy Authentication Required, which includes some things the other end needs to generate authentication parameters. Then the other end should send another INVITE, this time with authentication information, and Asterisk will either reply with a 200 OK if alright, or 401 Unauthorized if not alright |
| 12:13.16 | blitzrage | s/sense/sends |
| 12:13.38 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.39 | blitzrage | if the other end isn't trying to send another INVITE, then it's probably not getting the 407 |
| 12:13.53 | blitzrage | oh [TK]D-Fender is in the house... I can leave now |
| 12:15.16 | sysadmin-leb | Where can I set the authenication info |
| 12:15.23 | blitzrage | in sip.conf for the peer |
| 12:15.37 | sysadmin-leb | I have done this before but the only authenicatino info I used to set was for the end point |
| 12:15.55 | sysadmin-leb | for example a linksys device would take username password domain etc and a DID and that is it |
| 12:16.05 | sysadmin-leb | I did the same for asterisk and could dial towards any phone |
| 12:16.17 | blitzrage | are you connecting to an ITSP or a phone? |
| 12:16.18 | sysadmin-leb | but I am facing probs when I try to dial the DID from a normal phone |
| 12:16.29 | blitzrage | you still have to authenticate the ITSP |
| 12:16.44 | sysadmin-leb | I am calling a phone |
| 12:16.52 | blitzrage | you said you are calling a DID |
| 12:17.02 | blitzrage | the DID and the phone are two separate channels |
| 12:17.10 | blitzrage | incoming from the ITSP, and out to the phone |
| 12:17.15 | blitzrage | entirely separate connections |
| 12:17.19 | blitzrage | http://downloads.oreilly.com/books/9780596510480.pdf |
| 12:17.24 | Alex465 | why through oh323 is not sent dtmf |
| 12:18.13 | slima | blitzrage: I added WaitExten() after Background() and set autofallthrough to 'no' but stil dont works ;( |
| 12:18.25 | sysadmin-leb | I do outgoing calls from Asterisk to phones and I am trying to receive calls on Asterisk from a phone using the DID as the number that I dial on the phone |
| 12:18.29 | blitzrage | you shouldn't have changed the autofallthrough -- it was already right |
| 12:18.56 | blitzrage | slima: are you dialing fast enough? |
| 12:19.01 | slima | yes |
| 12:19.03 | blitzrage | there is nothing else wrong in the dialplan |
| 12:19.24 | blitzrage | other than maybe DTMF not getting through -- add 'dtmf' to the 'console =>' line of logger.conf and do 'logger reload' at the CLI |
| 12:19.32 | blitzrage | make sure you're seeing the DTMF |
| 12:19.38 | blitzrage | other than that -- I'm out -- going to the gym. |
| 12:19.53 | slima | thx |
| 12:20.06 | slima | bye |
| 12:21.52 | [TK]D-Fender | slima ; pastebin your dialplan and the CLI output of your failed call at verbose 10 |
| 12:21.53 | [TK]D-Fender | ~pb |
| 12:21.54 | jbot | A Pastebin is a place to paste your stuff 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 |
| 12:21.55 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 12:22.04 | blitzrage | [TK]D-Fender: he already did |
| 12:22.07 | blitzrage | but you weren't here |
| 12:22.43 | [TK]D-Fender | blitzrage: Fat load of good that does me NOW eh? |
| 12:22.55 | blitzrage | well... I was mostly commenting on the auto ~pb |
| 12:23.03 | BBHoss | anyone know any per minute speex outbound only providers? |
| 12:23.30 | slima | [TK]D-Fender: http://pastebin.com/f33d1f924 |
| 12:23.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:25.20 | [TK]D-Fender | slima: :now the output of "dialplan show mainmenu" |
| 12:26.43 | slima | http://pastebin.com/m6a8215aa |
| 12:27.20 | *** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60) |
| 12:28.05 | [TK]D-Fender | slima: [Oct 16 13:57:25] WARNING[28929]: pbx.c:2494 __ast_pbx_run: Invalid extension '1', but no rule 'i' in context 'mainmenu' |
| 12:28.30 | [TK]D-Fender | slima: this means you dialed a "1" but not the rest of 100 or 101 in time |
| 12:28.56 | [TK]D-Fender | slima: And because you have no "i" (Invalid) handler you don't get to try again and it hangs up |
| 12:29.17 | slima | bah, |
| 12:29.17 | slima | [14:03] slima >> hi, I have a problem with TIMEOUT(digit) when I typing '100' asterisk dont wait for and search a '1' exten my config: http://pastebin.com/f33d1f924 (sorry for my english) |
| 12:30.03 | [TK]D-Fender | slimyou need to type faster |
| 12:31.18 | [TK]D-Fender | slima: And because you have no "i" your caller had better get his choice right the first time. |
| 12:32.20 | sysadmin-leb | I am trying to make a call to my TuxBox from a normal phone I am getting a 401 Unauthorized error...I did go to sip.conf in the tuxbox |
| 12:32.44 | sysadmin-leb | and changed context=from-trunk but that did not change anything |
| 12:32.53 | sysadmin-leb | can anyone guide me into the right directin please ? |
| 12:33.49 | [TK]D-Fender | sysadmin-leb: 401 isn't a context issue, its user/pass <-- |
| 12:35.03 | sysadmin-leb | but the thing is that I have syslink device on which I can accept calls on the smae DID without settingt a user /pass |
| 12:35.13 | sysadmin-leb | well I do set a user/pass but I thought that was to make outgoing calls |
| 12:35.22 | sysadmin-leb | and I set the same user/pass on the trixbox |
| 12:35.32 | sysadmin-leb | and I can make outgoing calls |
| 12:36.11 | *** join/#asterisk bintut (n=bintut@cm2.gamma181.maxonline.com.sg) |
| 12:36.43 | BBHoss | you shoulndt have said trixbox :) |
| 12:36.53 | [TK]D-Fender | ~trixbox |
| 12:36.53 | jbot | hmm... trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 12:37.19 | [TK]D-Fender | sysadmin-leb: GUI's are not supported here. |
| 12:37.41 | [TK]D-Fender | sysadmin-leb: So I've explained what the 401 is. Go fix it. |
| 12:38.12 | sysadmin-leb | I already knew what it was ..I needed a hint in the right dir..thx for the help |
| 12:40.42 | BBHoss | http://www.liveleak.com/view?i=ba8_1190226087&p=1 |
| 12:40.42 | BBHoss | start your day off great |
| 12:41.02 | [TK]D-Fender | BBHoss: OLD..... |
| 12:41.29 | stmaher | blitzrage, timeout didnt work :-( exten => 1234,1,Dial(SIP/1961@fester,TIMEOUT,2000) |
| 12:41.51 | blitzrage | you're using it totally wrong |
| 12:42.03 | blitzrage | TIMEOUT is a dialplan function |
| 12:42.07 | blitzrage | you'd use it like: |
| 12:42.17 | blitzrage | exten => s,n,Set(TIMEOUT(digit)=2) |
| 12:42.42 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:43.07 | [TK]D-Fender | stmaher: What do you THINK you're doing in that Dial line? |
| 12:43.19 | slima | [TK]D-Fender: why Set(TIMEOUT(digit)=5) don`t wait 5secs when I dialing '1' '0' '0'? |
| 12:44.10 | stmaher | [TK]D-Fender, in the event of a busy or unavailable recipient to cut the timeout down from 10-30 seconds to 2 seconds |
| 12:45.02 | [TK]D-Fender | stmaher: If they are busy or unavailable, Dial with quit IMMEDIATELY |
| 12:45.13 | [TK]D-Fender | stmaher: exten => 1234,1,Dial(SIP/1961@fester) |
| 12:45.21 | stmaher | yeah |
| 12:45.38 | stmaher | Sorry.. |
| 12:45.41 | [TK]D-Fender | slima: enable core & dtmf debug and let me see shent he event is registered. |
| 12:46.02 | puzzled | hi |
| 12:46.32 | [TK]D-Fender | s/shent th/when the |
| 12:46.33 | stmaher | [TK]D-Fender, call flow xlite dials 1234@* -> *(1234@sipivr) -> third party sipIVR |
| 12:47.04 | stmaher | [TK]D-Fender, if the sipIVR is unavailable.. i need it to play a busy tone to the xlite user |
| 12:47.07 | stmaher | within 2 seconds |
| 12:47.30 | [TK]D-Fender | stmaher: Yeah I rememebmber your setup... only had *1* exten when we left but I understood you were expanding and you wanted to be able to transfer to other agents |
| 12:47.45 | stmaher | [TK]D-Fender, you have a good memory :_)) |
| 12:47.48 | [TK]D-Fender | stmaher: show me what its doing NOW. |
| 12:48.31 | slima | [TK]D-Fender: http://pastebin.com/m43960c5f |
| 12:48.33 | stmaher | [TK]D-Fender, http://pastebin.com/d558feffa |
| 12:48.50 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 12:49.09 | Uatec | lol, i just had to explain the weighted companion cube to my MD |
| 12:49.11 | *** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 12:49.23 | blitzrage | Uatec: eh? |
| 12:52.00 | [TK]D-Fender | stmaher: if the other side is "busy" dialplan will continue on priority 2. if they may not ANSWER then you'll want to set the maximum timout in seconds in your Dial line as a limit. That is the SECOND parameter. |
| 12:52.31 | stmaher | [TK]D-Fender, can you please give me an example? |
| 12:52.34 | [TK]D-Fender | slima: What DTMFMODE is your channel using? |
| 12:53.01 | slima | inband |
| 12:53.03 | [TK]D-Fender | stmaher: exten => 1234,1,Dial(SIP/1961@fester,10) <- after 10s withoutanswer go on to do other stuff |
| 12:53.11 | [TK]D-Fender | slima: What codec? |
| 12:53.15 | stmaher | thanks |
| 12:53.36 | slima | G711a |
| 12:55.02 | [TK]D-Fender | slima: Is this what your provider forces you to use? They should offer rfc2833.... I'm not sure if this parameter is applicable there (I think so), but you may be able to add "relaxdtmf=yes" to your sip.conf entry to have it ease up on DTMF detection. |
| 12:55.56 | *** join/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk) |
| 12:56.01 | [TK]D-Fender | slima: I can see you trying to dial 100 pretty fast there and I think its dropping the "00" for coming in to unsteady |
| 12:56.30 | stmaher | [TK]D-Fender, Hi tk.. made that change and its still taking 10 seconds.. when i set it to 2 |
| 12:57.17 | [TK]D-Fender | stmaher: Show me. |
| 12:57.52 | *** part/#asterisk BBHoss (n=hoss@146.229.183.84) |
| 12:57.57 | *** join/#asterisk BBHoss (n=hoss@146.229.183.84) |
| 12:58.11 | stmaher | [TK]D-Fender, config is the same as the last pastbin.. except with exten => 1234,1,Dial(SIP/1961@fester,2) .. |
| 12:58.25 | stmaher | [TK]D-Fender, would you like to see the SIP Debug from the CLI? |
| 12:58.27 | [TK]D-Fender | stmaher: pastebin the CALL showing me whats happening. |
| 12:58.38 | *** part/#asterisk BBHoss (n=hoss@146.229.183.84) |
| 12:58.39 | [TK]D-Fender | stmaher: CLI + sip debug. |
| 12:59.01 | _x86_ | when an analog station hangs up on a bridged call, and then tries to pick up the phone to dial someone, it gives a stutter tone, and they can hang up and pick up again and get the original bridged call back |
| 12:59.18 | _x86_ | how can i tell asterisk that when they hang up, actually hang up the call? |
| 12:59.24 | stmaher | [TK]D-Fender, http://pastebin.com/d7c2bd92a |
| 12:59.47 | [TK]D-Fender | stmaher: Complete waste. |
| 13:00.04 | [TK]D-Fender | stmaher: Forget sip debug. You didn't even have your CALL in there. |
| 13:00.25 | stmaher | [TK]D-Fender, the call is being made from an external xlite phone |
| 13:00.30 | [TK]D-Fender | _x86_: What phone? |
| 13:00.44 | [TK]D-Fender | stmaher: Well I don't see it REACHING * at all, do YOU? |
| 13:00.50 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.254) |
| 13:00.58 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:01.08 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:01.10 | stmaher | [TK]D-Fender, yeah i do # |
| 13:01.11 | stmaher | To: <sip:10.0.0.151> |
| 13:01.11 | stmaher | # |
| 13:01.11 | stmaher | Contact: <sip:asterisk@10.0.0.34> |
| 13:01.12 | [TK]D-Fender | doh |
| 13:01.34 | [TK]D-Fender | stmaher: those aren't CALLS |
| 13:01.46 | [TK]D-Fender | stmaher: Thats friggen QUALIFY=YES junk |
| 13:01.57 | [TK]D-Fender | stmaher: No INVITE, and no DIALPLAN EXECUTION. |
| 13:02.21 | stmaher | [TK]D-Fender, can you please specify the debug command you wish me to use for the CLI |
| 13:02.55 | [TK]D-Fender | stmaher: in the time frame that that pastebin covers, no call landed on * and nothing happened at all. |
| 13:02.57 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 13:03.37 | stmaher | [TK]D-Fender, the External IVR is unavailable.. ie nothign running on 5060 on the External IVR |
| 13:03.59 | stmaher | [TK]D-Fender, again call flow is xlite -> * -> IVR |
| 13:03.59 | _x86_ | [TK]D-Fender: some crappy analog VXi headset phone |
| 13:04.12 | _x86_ | [TK]D-Fender: basically just a dial pad and a headset |
| 13:04.14 | [TK]D-Fender | stmaher: That isn't our problem. ASTERISK is not getting called PERIOD, |
| 13:04.23 | stmaher | [TK]D-Fender, Yes it is.. |
| 13:04.31 | [TK]D-Fender | stmaher: Sip debug begs to differ |
| 13:05.00 | stmaher | [TK]D-Fender, With xlite i call 1234@10.0.0.34 (which is the Asterisk box) |
| 13:05.16 | [TK]D-Fender | stmaher: So either you have some nasty networking issues or what you are showing me isn't everything. |
| 13:05.23 | [TK]D-Fender | stmaher: because there is no invite in there. |
| 13:05.25 | *** join/#asterisk vargran (n=naquad@oman.Te.NeT.UA) |
| 13:05.33 | [TK]D-Fender | stmaher: thats jsut how it is. |
| 13:05.34 | vargran | hi everyone! |
| 13:05.47 | _x86_ | [TK]D-Fender: busydetect perhaps? right now it's off |
| 13:06.05 | _x86_ | or what's that other one... callprogress? but that's for PRI's only eh? |
| 13:06.11 | vargran | I want to set up some voip server for me & 10 my friends. where should I start? are there any ready solutions? |
| 13:06.27 | [TK]D-Fender | _x86_: lost your issues with my accidental disconnect |
| 13:06.28 | _x86_ | vargran: AsteriskNOW |
| 13:06.36 | vargran | ? |
| 13:06.39 | [TK]D-Fender | vargran: here : |
| 13:06.41 | [TK]D-Fender | ~book |
| 13:06.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 13:06.45 | [TK]D-Fender | _x86_: EW! |
| 13:07.01 | _x86_ | 07:59 < _x86_> when an analog station hangs up on a bridged call, and then tries to pick up the phone to dial someone, it gives a stutter tone, and they can hang up and pick up again and get the original bridged call back |
| 13:07.02 | Pon`work | Question: after the caller has hungup - how would I take control of the callee and transfer him somewhere else (caller comes from PSTN, callee is SIP) ? |
| 13:07.05 | _x86_ | 07:59 < _x86_> how can i tell asterisk that when they hang up, actually hang up the call? |
| 13:07.08 | sysadmin-leb | Just one note I was able to fix the problem using the Asterisk Book from Oreilly by setting the correct paramters in sip.conf...the idea is that it does not matter if I have a GUI or not in the end Trixbox provides Asterisk with a web interface with a page where I can overview all of my .conf files to edit |
| 13:07.09 | _x86_ | 08:00 < [TK]D-Fender> _x86_: What phone? |
| 13:07.12 | sysadmin-leb | anyhow I am glad it worked |
| 13:07.15 | _x86_ | 08:03 < _x86_> [TK]D-Fender: some crappy analog VXi headset phone |
| 13:07.19 | sysadmin-leb | thanks for your help folks |
| 13:07.22 | Katty | mew. |
| 13:07.30 | _x86_ | [TK]D-Fender: eh? |
| 13:07.38 | _x86_ | Katty: mornin |
| 13:07.38 | vargran | oh christ... I want simple functionality and I need to read a book and a lots of docs? is there any simpler way? |
| 13:07.51 | _x86_ | vargran: AsteriskNOW (2nd time) |
| 13:07.59 | [TK]D-Fender | sysadmin-leb: And the second you hit "commit changes" on your GUI next those changes get blown to BITS |
| 13:07.59 | Katty | _x86_: morning |
| 13:08.05 | javb | I`m getting this weird error: [Oct 16 09:06:33] NOTICE[2242]: chan_sip.c:13605 handle_request_invite: Failed to authenticate user "Recepcion" <sip:105@10.0.0.55>;tag=D481078B-4E22DF54 |
| 13:08.07 | vargran | asteriskNOW helped thnx [TK]D-Fender :) |
| 13:08.10 | javb | Any ideas? |
| 13:08.20 | javb | This is new after upgrading to 1.4 |
| 13:08.23 | JT | vargran: do you have a good understanding of telephony, networking, and linux? |
| 13:08.31 | vargran | [TK]D-Fender distribution??? |
| 13:08.35 | vargran | no f way |
| 13:08.55 | _x86_ | vargran: Asterisk@home / FreePBX |
| 13:08.56 | javb | And not able to transfer.. :/ |
| 13:08.59 | stmaher | [TK]D-Fender, http://pastebin.com/d2a9a724c sorry .. missed the sip debug and verbose command |
| 13:09.00 | [TK]D-Fender | _x86_: Memor is faded this morning and I have no scroll-back, refresh my memory.... |
| 13:09.04 | JT | vargran: ? |
| 13:09.14 | javb | [TK]D-Fender: Zup.. |
| 13:09.44 | _x86_ | [TK]D-Fender: what did you miss? |
| 13:09.49 | vargran | I got setted up router and I don't want to install some another distribution. |
| 13:10.01 | [TK]D-Fender | stmaher: -- Executing Dial("SIP/10.0.0.151-b5c11500", "SIP/1961@fester") in new stack |
| 13:10.03 | JT | vargran: do you have an answer to my question? |
| 13:10.07 | _x86_ | [TK]D-Fender: when a user hangs up a bridged call, it takes asterisk a while to actually tear the call down... why? |
| 13:10.30 | stmaher | [TK]D-Fender, OK........... so now what do you need? |
| 13:10.32 | [TK]D-Fender | stmaher: Oct 16 15:20:06 NOTICE[13754]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination |
| 13:10.40 | vargran | JT: linux, networking, not telephony |
| 13:10.54 | JT | vargran: then reading a book would help |
| 13:11.12 | [TK]D-Fender | stmaher: You are not setting a timeout and your call is (bit itself) being rejected IMMEDIATLY, and you have no 2nd priority to continue processing in. What are you expecting here? |
| 13:11.16 | stmaher | [TK]D-Fender, thats fine.. I just need to reduce the time now that it takes to give a busy signal |
| 13:11.18 | javb | Upgrade to Asterisk 1.4, now, get this notice: --> [Oct 16 09:06:33] NOTICE[2242]: chan_sip.c:13605 handle_request_invite: Failed to authenticate user "Recepcion" <sip:105@10.0.0.55>;tag=D481078B-4E22DF54 |
| 13:11.29 | javb | Cant tranfer |
| 13:11.31 | vargran | JT: hf*ck :( I had a hope that it wount take a week :( |
| 13:11.37 | [TK]D-Fender | vargran: Go download the BOOK, and install * and get learning |
| 13:11.49 | stmaher | [TK]D-Fender, ok where and how do i set the timeout |
| 13:12.35 | [TK]D-Fender | stmaher: I just gave that to you like TWICE!!!!!!! |
| 13:12.35 | [TK]D-Fender | stmaher: -- Executing Dial("SIP/10.0.0.151-b5c11500", "SIP/1961@fester") <--- wheres the timout? |
| 13:12.35 | Katty | and the sploding begins... |
| 13:12.35 | stmaher | [TK]D-Fender, its in the config alright |
| 13:12.36 | javb | [TK]D-Fender: any idea ? :/ |
| 13:12.39 | stmaher | [TK]D-Fender, http://pastebin.com/d2a9a724c |
| 13:12.45 | [TK]D-Fender | stmaher: Dial(SIP/1961@fester,10) <-------- waits 10s if no answer |
| 13:12.47 | stmaher | [TK]D-Fender, SORRY IGNORE PASTEBIN!! |
| 13:12.56 | stmaher | [TK]D-Fender, exten => 1234,1,Dial(SIP/1961@fester,2) ; permit transfer |
| 13:13.17 | [TK]D-Fender | stmaher: Where do you see the "," in that pastebin? |
| 13:13.31 | [TK]D-Fender | stmaher: Where do you see the ",2" in that pastebin? |
| 13:13.42 | stmaher | Sorry.. thats an old pastebin please ignore |
| 13:13.54 | stmaher | exten => 1234,1,Dial(SIP/1961@fester,2) ; permit transfer is in the config |
| 13:14.19 | [TK]D-Fender | stmaher: and it wouldn't matter because the call FAILS. That isn't "waited for the other side to answer", this was "I don't even know how to call them so I give up!" |
| 13:14.39 | [TK]D-Fender | stmaher: Wake up and show me what you've been requested to provide. |
| 13:15.00 | [TK]D-Fender | stmaher: because if thats in your config you may want to APPLY your changes. |
| 13:15.11 | [TK]D-Fender | *grumble* |
| 13:15.49 | [TK]D-Fender | javb: those little bits you show aren't enough to help with your problem. pastebin entire calls please |
| 13:16.02 | [TK]D-Fender | Katty: Mew..... |
| 13:16.12 | Katty | [TK]D-Fender: morning sunshine. |
| 13:16.20 | Katty | [TK]D-Fender: you appear to be nicely caffeinated. |
| 13:16.27 | stmaher | [TK]D-Fender, I have restarted asterisk... and the problem still persists.. let me explain.. I have an external IVR that is not currently running.. IE NOTHING listening on 5060.. If i use xlite to call it.. i get an unavailable tone instantly.. with xlite -> asterisk -> IVR i get a 10 second delay before the unavailable tone |
| 13:16.28 | [TK]D-Fender | Katty: sunshine BURNS PEOPLE. |
| 13:16.38 | javb | [TK]D-Fender: http://dpaste.com/22630/ |
| 13:16.41 | Katty | [TK]D-Fender: if the shoe fits.. ;) |
| 13:17.11 | _x86_ | [TK]D-Fender: did you get disco again? :P |
| 13:17.22 | _x86_ | [TK]D-Fender: or are you simply ignoring me hehehe |
| 13:17.27 | [TK]D-Fender | javb: I don't see sip debug for 105's attempted call... |
| 13:17.58 | stmaher | [TK]D-Fender, I am trying to reduce the time it takes for asterisk to realise the IVR is down and send an unavailble to the xlite in under 2 seconds |
| 13:18.03 | *** join/#asterisk MrParity (n=patrick@dslb-088-077-010-174.pools.arcor-ip.net) |
| 13:18.07 | MrParity | hi ho :-) |
| 13:18.08 | [TK]D-Fender | _x86_: Yes, I'm missing everything. Pastebin the whole deal & debug,e tc and I'll pick up on it. |
| 13:18.23 | Katty | _x86_: i get to kick everyone off the email this server this morning... all in the name of manual backups! |
| 13:18.36 | [TK]D-Fender | stmaher: its already failing instantly and you call should end as fast. |
| 13:18.56 | stmaher | [TK]D-Fender, that sentence does not make sense! |
| 13:18.57 | _x86_ | [TK]D-Fender: i don't have a debug of the event, and it's hard for me to re-produce, since i'm ~100 miles from the location |
| 13:19.20 | stmaher | [TK]D-Fender, why is it taking 10 seconds with asterisk |
| 13:19.21 | [TK]D-Fender | _x86_: that should account for a few ms of SSH :p |
| 13:19.46 | krdian_ | hi |
| 13:19.55 | [TK]D-Fender | stmaher:maybe, just MAYBE you should make the next pririty in your extens a HANGUP or a BUSY <---- |
| 13:19.57 | Katty | i gave at the office. |
| 13:19.59 | _x86_ | [TK]D-Fender: right, but i don't have an analog phone to play with here, that's connected to that phone system |
| 13:20.38 | javb | http://dpaste.com/22632/ <--- 103 is talking, i wall 103, and does not have call waiting.. and IT USED to have. |
| 13:22.00 | krdian_ | what ip phones can you recommend for * ? |
| 13:22.12 | stmaher | [TK]D-Fender, i put a exten => s,2,Busy and it still doesnt work.. is there something wrong with the busy line? |
| 13:22.13 | _x86_ | krdian_: anything with the "Polycom" name on it ;) |
| 13:22.19 | javb | krdian_: Polycom |
| 13:22.40 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 13:22.44 | krdian_ | _x86_: you think they are better than grandstream? |
| 13:22.59 | [TK]D-Fender | javb: well it gets called for 20s without answer. Guess your PHON has an issue |
| 13:23.04 | Qwell | anything is better than grandstream |
| 13:23.10 | [TK]D-Fender | stmaher: Show said anything about "s"?! |
| 13:23.13 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 13:23.22 | [TK]D-Fender | stmaher: what >NUMBER< are you dialing? |
| 13:23.29 | [TK]D-Fender | s/show/who" |
| 13:23.39 | MrParity | i have a problem with a voip account. i usually use internvoip and isdn to call outside. now i want to try to use voip outside to an voip provider, but someting doesnt work. " Executing Dial("SIP/gxp2000-0820bee0", "SIP/016334445334@arcor|30|tr") in new stack" is okay -i think, but then i get th following message: "Forbidden - wrong password on authentication for INVITE to '"Patrick" <sip:gxp2000@88.87.10.174>;tag=as2bf550e2'" |
| 13:23.44 | krdian_ | thank you :) |
| 13:23.55 | [TK]D-Fender | krdian_: Where are you located? |
| 13:24.05 | krdian_ | [TK]D-Fender: Poland |
| 13:24.07 | stmaher | [TK]D-Fender, Thank you .. it works now :-) |
| 13:24.16 | MrParity | can anyone push me into the right direction? |
| 13:24.22 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 13:24.28 | [TK]D-Fender | krdian_: Linksys is probably a good bet for you. |
| 13:24.45 | Sci_05 | morning all |
| 13:25.07 | javb | [TK]D-Fender: http://dpaste.com/22633/ <--- i called 105, told 105 5o transfer a call, this sounds busy... |
| 13:25.16 | _x86_ | krdian_: anything is better than grandstream |
| 13:25.19 | _x86_ | ~grandstream |
| 13:25.20 | jbot | grandstream is, like, the Yugo of VoIP hardware. Run. Run away now. |
| 13:25.26 | _x86_ | see ;) |
| 13:25.28 | jfitzgibbon | MrParity: do a 'sip debug peer <peername>' for your ITSP and double check that the password you're using is your auth PW, not your register PW (assuming your provider makes you register) |
| 13:25.41 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:25.54 | [TK]D-Fender | ~gs |
| 13:25.55 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:25.57 | [TK]D-Fender | ^^^^^^^^^ |
| 13:27.17 | javb | This is happening with all the 15 phones here, Polycom 330. and they were ok with Asterisk 1.2 lastnight! C |
| 13:27.21 | Katty | they brought me a dirty windows 98 machine to fix. |
| 13:27.30 | Katty | and by dirty, i mean it's been in a warehouse |
| 13:27.44 | javb | [TK]D-Fender: http://dpaste.com/22633/ <--- i called 105, told 105 5o transfer a call, this sounds busy... |
| 13:27.46 | javb | This is happening with all the 15 phones here, Polycom 330. and they were ok with Asterisk 1.2 lastnight! C |
| 13:27.49 | Katty | this thing belongs in a dumpster :< |
| 13:27.56 | javb | (think i got disconnected) |
| 13:28.13 | MrParity | jfitzgibbon: itsp? |
| 13:28.20 | jfitzgibbon | ~itsp |
| 13:28.21 | jbot | itsp is probably an Internet Telephony Service Provider, or a "VoIP Phone Company". |
| 13:28.23 | jfitzgibbon | damn |
| 13:28.29 | jfitzgibbon | no, there it is |
| 13:28.41 | [TK]D-Fender | javb: next one with sip debug please. |
| 13:28.55 | MrParity | jfitzgibbon: *g* ok :-) |
| 13:29.36 | defswork | can you monitor a ring group with blf |
| 13:29.39 | krdian_ | [TK]D-Fender: i sse, so i have to exchange my gxp2000 |
| 13:29.45 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 13:29.55 | defswork | quit |
| 13:30.05 | MrParity | jfitzgibbon: 02151.sip.arcor.de:5060 [username] 1785 Registered |
| 13:30.18 | MrParity | jfitzgibbon: isn't i ok? |
| 13:30.24 | MrParity | it |
| 13:30.31 | jfitzgibbon | MrParity: do the sip debug for the call with verbose set to at least 3 and pastebin it |
| 13:30.33 | [TK]D-Fender | defswork: you can monitor a group of DEVICES. And NEVER use the term "ring group" again. EVER |
| 13:30.52 | jfitzgibbon | MrParity: also include the dialplan you're using to call |
| 13:31.03 | MrParity | jfitzgibbon: ok, ill do it, thanks :-) |
| 13:31.29 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 13:31.32 | defswork | [TK]D-Fender: :o |
| 13:32.45 | defswork | [TK]D-Fender: I set a blf on an aastra 9133i to group 600 but doesn't show when it's ringing |
| 13:32.57 | defswork | should that work ? |
| 13:33.36 | javb | [TK]D-Fender: http://dpaste.com/22634/ |
| 13:34.00 | [TK]D-Fender | defswork: Gee I dunno... did you se up the HINT for it properly? |
| 13:34.04 | MrParity | jfitzgibbon: http://pastebin.com/m3b0dd463 |
| 13:34.39 | javb | <PROTECTED> |
| 13:36.26 | b1ch0 | hi everybody again, does anybody worked with stun server on the same asterisk machine before ? .... it is to avoid transversal nat problem (without VPN between remote branch) |
| 13:36.54 | [TK]D-Fender | b1ch0: * doesn't currently support STUN, nor does it need it |
| 13:37.01 | defswork | [TK]D-Fender: you ask as if I have a clue what I am doing :) |
| 13:37.17 | [TK]D-Fender | defswork: * doesn't do anything that you don't configure it to do. |
| 13:37.28 | [TK]D-Fender | defswork: Go read up on setting up dialplan hints. |
| 13:37.51 | defswork | [TK]D-Fender: I'm using freepbx so have some constraints :) |
| 13:38.11 | [TK]D-Fender | defswork: Sorry... You're stuck with the IQ you've got, make the most of it :p |
| 13:38.28 | [TK]D-Fender | defswork: Oh... and GTFO ;) |
| 13:38.43 | defswork | get the furry orange ? |
| 13:38.44 | [TK]D-Fender | Merci, salut la visite! |
| 13:39.02 | [TK]D-Fender | Suivent, NEXT!@!@! |
| 13:39.26 | b1ch0 | <[TK]D-Fender>: i mean as externa aplication running on the same machine .... or any other solution that can help to avoid that problem |
| 13:39.58 | [TK]D-Fender | b1ch0: You haven't actually DESCRIBED the problem and scenario..... |
| 13:40.33 | javb | [TK]D-Fender :/ |
| 13:40.40 | syle | what OS's you guys run? |
| 13:41.01 | b1ch0 | i have and * with 50 IP phones on the same network , firewalled |
| 13:41.09 | [TK]D-Fender | syle: Popular : CentOs, FC, Debian, Gentoo, Slackware, and jsut about everything else under the sun |
| 13:41.15 | b1ch0 | and 6 remote branch offices |
| 13:41.37 | b1ch0 | each one behin firewall |
| 13:41.49 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 13:41.58 | keith4 | how's asterisk's H323 support? |
| 13:42.08 | [TK]D-Fender | b1ch0: Here : |
| 13:42.09 | [TK]D-Fender | ~sipnat |
| 13:42.10 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:42.11 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 13:42.18 | [TK]D-Fender | keith4: Mew |
| 13:42.21 | [TK]D-Fender | keith4: Meh |
| 13:42.30 | b1ch0 | like 192.x.x.x - FIREWALL - INTERNET - FIREWALL- 10..x.x.x |
| 13:43.21 | [TK]D-Fender | b1ch0: please read the guide I linked for you |
| 13:43.26 | badcfe | whats the canonical way of allowing a user to access the asterisk cli? (asterisk run as asterisk) |
| 13:43.31 | [TK]D-Fender | b1ch0: it will tell you what you need to do. |
| 13:43.47 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:43.47 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:44.13 | defswork | [TK]D-Fender: just for bonus IQ points I think I got it working :) |
| 13:44.20 | javb | everytime i want to transfer i get this : [Oct 16 09:43:40] NOTICE[2242]: chan_sip.c:13605 handle_request_invite: Failed to authenticate user "Joel Valdez" <sip:102@10.0.0.55>;tag=B14A7CA0-BEDDA39... where "102" is the exten wanting to DO the trasnfer. |
| 13:44.38 | [TK]D-Fender | defswork: Success doesn't make you smarter, but hopefully QUIETER :p |
| 13:44.41 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.240) |
| 13:44.46 | defswork | :) |
| 13:44.47 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 13:45.07 | badcfe | whats the canonical way of allowing a user to access the asterisk cli? |
| 13:45.13 | [TK]D-Fender | javb: When you feel like following what we ask you to do you might eventually get somewhere... |
| 13:46.06 | javb | [TK]D-Fender: Ok, i`m sorry, but, can you explain WHAT is what you want me to do? |
| 13:46.23 | [TK]D-Fender | javb: I asked for SIP DEBUG for your call. |
| 13:46.31 | keith4 | javb: he's in a good mood today, you're lucky |
| 13:46.41 | [TK]D-Fender | keith4: No, decidedly not. |
| 13:47.07 | javb | how can i use the SIP DEBUG cmd for an specific call |
| 13:47.54 | [TK]D-Fender | javb: Don't do it for a specific call, do ti in general so I can see everything. |
| 13:49.14 | javb | [TK]D-Fender: http://dpaste.com/22635/ |
| 13:51.10 | [TK]D-Fender | javb: And where's the error in there? |
| 13:51.34 | [TK]D-Fender | javb: Calls a zap channel, looks fine... |
| 13:53.44 | b1ch0 | <[TK]D-Fender> : Asterisk as a SIP server behind nat, clients on the same network (always behind NAT), clients on the outside behind a second NAT connecting to Asterisk |
| 13:53.46 | javb | it is just when i need the phone to use two lines |
| 13:54.06 | [TK]D-Fender | b1ch0: Thats fine, and this scenatio is described in there... |
| 13:54.25 | [TK]D-Fender | javb: Whatver problems you are having is not in that pastebin. |
| 13:55.00 | syle | some sick people trolling channels today |
| 13:55.01 | javb | Here http://dpaste.com/22636/ <--- i make a call to voicemail, then, try to transfer it. |
| 13:55.03 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 13:55.18 | syle | one guy is describing what his cat looks like after every 30 sec its in the microwave |
| 13:57.30 | [TK]D-Fender | syle: is it ON? |
| 13:58.06 | syle | yep, talking about its eyes melting and crap |
| 13:58.12 | syle | thankgod he got kicked out of channel |
| 13:58.17 | stimpie | I have a line 'exten => _X!,8,Set(CDR(userfield)=${CDR(userfield)} Hangupcause:${HANGUPCAUSE})' which sometimes adds 'set, CDR(userfield)= Hangupcause:38' to the cdr |
| 13:58.20 | CBU[^_^]M`` | hellppp |
| 13:58.51 | CBU[^_^]M`` | X lite ===> Internet ===> My PABX.... error 404 |
| 13:59.22 | [TK]D-Fender | javb: To: <sip:10@10.0.0.55;user=phone>;tag=as25d62700 - SIP/2.0 484 Address Incomplete |
| 13:59.34 | CBU[^_^]M`` | X lite soft phone ===> Internet ===> My PABX.... error 404 on the softphone |
| 13:59.38 | [TK]D-Fender | javb: Who is 10? That number is no food. |
| 13:59.41 | [TK]D-Fender | good* |
| 14:00.03 | javb | and user "phone" does not exist. |
| 14:00.12 | [TK]D-Fender | CBU[^_^]M``: pastebin the call attempt with sip debug enabled |
| 14:00.14 | [TK]D-Fender | ~pb |
| 14:00.14 | jbot | A Pastebin is a place to paste your stuff 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 |
| 14:00.15 | javb | i dont understand that. |
| 14:00.16 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:00.32 | CBU[^_^]M`` | thanks ill try that |
| 14:00.38 | [TK]D-Fender | javb: You are dialing "10" and that exten is no good. |
| 14:01.08 | *** join/#asterisk ming_zym (n=ming_zym@124.254.57.51) |
| 14:01.50 | javb | but i am not dialing 10, and the dial plan y exact the same as inthe asterisk 1.2 |
| 14:02.25 | *** join/#asterisk gardo (n=gardo@121.97.251.62) |
| 14:02.39 | [TK]D-Fender | javb: your phone begs to differ. |
| 14:02.52 | [TK]D-Fender | javb: [TK]D-Fender>javb: To: <sip:10@10.0.0.55;user=phone>;tag=as25d62700 - SIP/2.0 484 Address Incomplete |
| 14:03.31 | [TK]D-Fender | javb: Something tells me you have not correctly configured your POLYCOM's dialplan <------ |
| 14:04.30 | b1ch0 | <[TK]D-Fender> : any other good site that can help to solve NAT problem ? ... sorry about bothering you so much |
| 14:05.09 | [TK]D-Fender | b1ch0: well you haven't SHOWN me what you've done so far, nor described the actual problem, so there really isn't anything to say for it. |
| 14:05.20 | [TK]D-Fender | b1ch0: PASTEBIN is your friend. |
| 14:05.22 | [TK]D-Fender | ~pb |
| 14:05.23 | jbot | A Pastebin is a place to paste your stuff 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 |
| 14:05.24 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:05.34 | javb | [TK]D-Fender: i havent touch the 10 polycoms dialplan, but, may that problem be noticed now in asterisk 1.4 |
| 14:05.59 | [TK]D-Fender | javb: go look at its dialplan |
| 14:06.12 | sysadmin-leb | Hi All i have running asterisk with multiple voip phones ...I have added a DID and I can accept incoming calls from a PSTN however the same softphone always rings how can I change the default phone that will ring when I receive an incomin call |
| 14:07.20 | [TK]D-Fender | sysadmin-leb: its your dialplan, it does whatever you TELL IT TO. |
| 14:07.38 | CBU[^_^]M`` | X lite soft phone ===> Internet ===> My PABX.... error 404 on the softphone still |
| 14:07.42 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 14:07.50 | [TK]D-Fender | CBU[^_^]M``: And you've shown us exactly NOTHING. |
| 14:09.30 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 14:11.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:11.22 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 14:11.33 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-7be70ff1c3de00de) |
| 14:11.33 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:11.46 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:11.47 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:13.43 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:15.33 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 14:19.59 | MrParity | it works :-) |
| 14:20.10 | MrParity | jfitzgibbon: thanks :-) |
| 14:21.13 | *** join/#asterisk ylon (n=ylon@cpe-76-181-182-10.columbus.res.rr.com) |
| 14:22.01 | ylon | I've got an emergency, someone set up an asterisk adhearsion system for me and they are not available for help and I'm scratching around trying to figure out what is going on by the logs, etc. |
| 14:22.24 | ylon | can anyone help me with some basic troubleshooting to isolate the issue? |
| 14:22.29 | [TK]D-Fender | ~ask |
| 14:22.30 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:23.06 | peanut- | whos here against their will? |
| 14:23.22 | nestAr | ME! |
| 14:23.26 | stimpie | ME! |
| 14:23.27 | nestAr | ;) |
| 14:23.44 | ylon | so is that a no? That is my question... |
| 14:23.55 | [TK]D-Fender | "you can check out anyt ime you like, but you can never leave..." |
| 14:23.58 | ylon | I need some basic steps to move forward here in isolating the issue |
| 14:24.00 | *** join/#asterisk synthetiq (n=tampon@193.79.224.62) |
| 14:24.14 | stimpie | ylon, you havent even told us what the issue is |
| 14:24.20 | [TK]D-Fender | ylon: thats a "ask a SPECIFIC question and maybe you'll get a SPECIFIC answer |
| 14:24.26 | ylon | that's correct, I don't know the issue |
| 14:24.30 | [TK]D-Fender | ylon: PASTEBIN is your friend |
| 14:24.32 | [TK]D-Fender | ~pb |
| 14:24.33 | jbot | A Pastebin is a place to paste your stuff 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 |
| 14:24.34 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:24.41 | ylon | there is dialtone, but the calls don't go out |
| 14:24.44 | ylon | nor can it receive calls |
| 14:24.50 | [TK]D-Fender | ylon: You'd better know SOMTHING of use for us because we're not psychic |
| 14:24.59 | ylon | I'd like to know where to look in order to find answers in the logs |
| 14:25.19 | synthetiq | if i do a call to synthetic@my.asterisk.box .... is there someway to prevent the striping of the "s" ? this happens when i use voicemail... exten => _[0-9a-zA-Z].,1,Voicemail(synthetic@context) |
| 14:25.26 | ylon | okay, lsof is my friend |
| 14:25.34 | [TK]D-Fender | ylon: what technologies are involved? What is talking to what? How far are calls getting? What are the symptoms? |
| 14:25.53 | synthetiq | or i can see a problem with "u" too |
| 14:26.29 | ylon | Okay, thanks: I have SIP phones connected to a poe switch which in turn is connected to a server with asterisk and adhearsion running (on a ruby web interface) |
| 14:26.30 | MrParity | wow. now i have an other problem. after 1:20 the call hangs up (SIP/2.0 487 Request Terminated) |
| 14:26.42 | [TK]D-Fender | synthetiq: You should not be using NAMES at box #'s |
| 14:26.46 | MrParity | does anyone have an idea that could be wrong? |
| 14:26.49 | *** join/#asterisk BBHoss (n=hoss@146.229.183.84) |
| 14:26.51 | ylon | Calls are not ringing into the building |
| 14:27.02 | [TK]D-Fender | synthetiq: but you may be able to get around this by using the SECOND parameter for Voicemail explicitly. |
| 14:27.08 | ylon | Nor can calls go out, there is a dial tone, however there is no outbound traffic from the server |
| 14:27.29 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 14:27.39 | [TK]D-Fender | ylon: Phone is registered to * directly? |
| 14:28.09 | ylon | so, I don't know, the fellow set it up so that registration occurs through adhearsion via the web interface |
| 14:28.21 | synthetiq | "SECOND" parameter? |
| 14:28.35 | [TK]D-Fender | synthetiq: "show application voicemail" <- RTFM :) |
| 14:28.39 | ylon | (whoops, "so" was a mistake, was going to type something else) |
| 14:29.00 | ylon | I'm peaking into the event_log right now |
| 14:29.08 | ylon | but nothing really telling appears to be in there |
| 14:29.09 | [TK]D-Fender | ylon: pastebin a failed call attempt at * CLI with SIP DEBUG enabled |
| 14:29.22 | ylon | can you tell me how to do that? |
| 14:29.23 | [TK]D-Fender | ylon: and you likely won't see anything of sue in logs... |
| 14:29.31 | [TK]D-Fender | ylouse* |
| 14:29.57 | stimpie | ylon, connect to asterisk cli by 'asterisk -r' |
| 14:29.59 | [TK]D-Fender | ylon: get to * CLI and type in "sip debug". Then place a call and pastebin the complete CLI output. |
| 14:30.42 | ylon | how do you place a call once there |
| 14:30.49 | stimpie | use a phone |
| 14:31.04 | ylon | alright, I'm not onsite, doing this via ssh stimpie, sorry |
| 14:31.07 | [TK]D-Fender | ylon: use your phone like nomal and place a call |
| 14:31.09 | ylon | I need to do this virtually |
| 14:31.13 | stimpie | for dialing from asterisk cli use the originate command |
| 14:31.21 | [TK]D-Fender | ylon: You need to SHOW us the problem. |
| 14:31.40 | [TK]D-Fender | ylon: So if you can't replicate it then you aren't going to be able to identify the problem and fix it. |
| 14:31.58 | ylon | sure, lets see if this does not show us the problem, right? |
| 14:32.09 | [TK]D-Fender | stimpie: thats worthless if its an auth issue, bad contexts, etc... and he has NO clue about his system |
| 14:32.11 | ylon | I'm going to try to place a virtual sip call to myself |
| 14:32.18 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:33.45 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.182) |
| 14:35.07 | CBU[^_^]M`` | 5030, 10000-20000 <= are these the only port that i need to set for the port forwarding? |
| 14:35.28 | [TK]D-Fender | CBU[^_^]M``: Here : |
| 14:35.31 | [TK]D-Fender | ~sipnat |
| 14:35.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:35.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 14:35.51 | CBU[^_^]M`` | hmmm.. thanks :) |
| 14:37.23 | CBU[^_^]M`` | question again :) |
| 14:37.37 | CBU[^_^]M`` | where can i find the NAT = yes in asterisk? |
| 14:38.35 | De_Mon | CBU[^_^]M`` keep reading |
| 14:38.45 | lirakis | does anyone know how to correlate a entry from the queue_log file that asterisk generates, to a cdr?? .. |
| 14:39.51 | CBU[^_^]M`` | De_Mon.. can it be accessed through the GUI or do i need to go to the CPU w/ the asterisk? my CPU with the asterisk dont have a monitor hehehe |
| 14:40.12 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:40.13 | *** part/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:41.00 | [TK]D-Fender | CBU[^_^]M``: /etc/asterisk <- folder with your config files. |
| 14:41.02 | *** join/#asterisk toot (n=toot@84.19.254.50) |
| 14:41.04 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 14:41.21 | [TK]D-Fender | CBU if you don't even know where they are or how to get to them then we can't help you |
| 14:42.06 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:42.10 | [TK]D-Fender | lirakis: before sending a call to queue, set the accountcode to the uniqueID. That will map the the uniqueid in our queue log |
| 14:42.19 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-6788265e7022a3ca) |
| 14:42.41 | lirakis | [TK]D-Fender: hmm .. okay |
| 14:43.12 | CBU[^_^]M`` | :) |
| 14:44.06 | ylon | okay, chatted with someone for a moment and it turns out that I'm using an agi script to dial, can anyone help with that in terms of taking this a step further in figuring out how to properly troubleshoot this remotely? |
| 14:44.41 | [TK]D-Fender | ylon: You are in the territory of |
| 14:44.42 | [TK]D-Fender | ~hafc |
| 14:44.43 | jbot | i heard hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 14:45.05 | [TK]D-Fender | ylon: You are no doubt running in a highly complex solution |
| 14:45.54 | [TK]D-Fender | ylon: www.voip-info.org <- go check out the consultants list |
| 14:46.18 | orakle | heh. hafc. |
| 14:49.15 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 14:50.30 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 14:50.53 | javb | if i have installed asterisk 1.4, how can i totally uninstall it... have my system clean, and install asterisk 1.2 ? |
| 14:52.29 | orakle | Heh, I did this once |
| 14:52.30 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 14:52.39 | javb | is it posible? |
| 14:52.45 | orakle | yeah |
| 14:52.57 | orakle | i used locate and kept deleting until it couldn't find anything called asterisk anymore |
| 14:53.06 | javb | :/ |
| 14:53.18 | orakle | i don't think there's a nice uninstall command sadly |
| 14:53.19 | JT | it's quite easy |
| 14:53.25 | JT | deleted the modules binaries |
| 14:53.30 | JT | maybe change some config files |
| 14:53.32 | JT | recompile |
| 14:54.20 | orakle | well yeah that takes care of the main stuff |
| 14:54.31 | orakle | but if you want to make your system totally clean before goign to 1.2.. |
| 14:54.35 | orakle | it's a bit more work |
| 14:54.43 | JT | nonsense |
| 14:54.49 | JT | what i said is sufficient |
| 14:54.54 | [TK]D-Fender | agreed |
| 14:54.55 | JT | wipe the configs if you want |
| 14:54.57 | orakle | i don't disagree |
| 14:55.00 | orakle | it'll work |
| 14:55.08 | orakle | he was asking how to "clean" his system that's all |
| 14:55.14 | javb | [TK]D-Fender: i have the 501 here, working great, i have checked the digit map, and is THE SAME as the 330... NOTE: 330 were working normal 10 hours ago.. just when asterisk 1.2 was working |
| 14:55.22 | javb | This is very very very weird. |
| 14:55.32 | javb | And no, i dont need coffee anymore |
| 14:55.37 | orakle | mmm coffee |
| 14:55.44 | [TK]D-Fender | javb: We both see that the phone is sending 10. that is not *'s fault. It is your PHONE or the user behind it |
| 14:56.16 | javb | [TK]D-Fender: ok. |
| 14:56.28 | [TK]D-Fender | javb: Pick one, because either way this is not *'s fault |
| 14:59.23 | syle | anyone recommend a good colocation? was with servepath for 1k a year and dudes after 4 years being there double the price on me |
| 14:59.44 | JT | $1k a year got you what? |
| 14:59.45 | javb | [TK]D-Fender: Do you know where can i trubleshoot this o Polycoms web based config |
| 14:59.47 | javb | ? |
| 15:01.13 | syle | i had 300GB a month and big ass pipe |
| 15:02.17 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 15:02.50 | JT | syle: how many RUs? |
| 15:03.46 | syle | i can;t even remember was a 2u or 5u |
| 15:04.15 | syle | big ass dell server one of those 2650's i beleive |
| 15:04.34 | JT | well do you own the server or not? |
| 15:04.39 | syle | yep |
| 15:04.40 | JT | and a 2650 is 2RU |
| 15:04.44 | JT | so not that big |
| 15:05.02 | syle | i like to think it is has 6 scsi drives lol |
| 15:05.10 | Dan0maN_Work | yes |
| 15:05.19 | Dan0maN_Work | it has room for 5 scsi's |
| 15:05.25 | Dan0maN_Work | (got one in the other room) |
| 15:05.30 | Dan0maN_Work | but that's not all that much space |
| 15:05.32 | JT | 6 probably |
| 15:05.51 | Dan0maN_Work | (ours has 5 with a cd/floopy) |
| 15:06.02 | syle | i had 6 put in there, i have never seen it, ordered it in states and shipped it right to a colo |
| 15:06.05 | orakle | floopy :) |
| 15:06.23 | syle | has about 8 gigs of ram |
| 15:06.31 | syle | dual cpus 3.2 ghz |
| 15:06.35 | JT | what did you need so many disks for? |
| 15:06.36 | khronos | I've used www.alphared.com and www.sagonet.com before. |
| 15:07.09 | syle | i was increasing I/O efficiency for raid5 |
| 15:07.21 | syle | more drives faster disk speed |
| 15:07.28 | JT | high drive failure rate |
| 15:07.52 | syle | haven;t had one die yet |
| 15:07.53 | JT | also with RAID5, faster speed with lots of drives really depends on having a very good RAID5 controller |
| 15:08.08 | JT | that doesn't mean one won't |
| 15:08.11 | syle | its the megaraid2 driver |
| 15:08.18 | syle | acraid i beleive |
| 15:08.27 | JT | controller |
| 15:08.29 | JT | not driver |
| 15:08.36 | Dan0maN_Work | 04:08.1 RAID bus controller: Dell PowerEdge Expandable RAID Controller 3/Di (rev 01) |
| 15:08.38 | toot | is raid5 faster? i thought raid1 was faster :) |
| 15:08.39 | [TK]D-Fender | javb: You should only be configuring these phones via provisioning. |
| 15:08.43 | syle | i don;t remember i bought the server for about 10k 3 years ago |
| 15:08.54 | [TK]D-Fender | javb: And No, I am not certain which screen the dialplan is in. |
| 15:08.58 | toot | slower write, faster read |
| 15:09.06 | JT | toot: RAID5 can be fast for reads, very slow to write |
| 15:09.12 | javb | [TK]D-Fender: what do you mean by "provisioning"? |
| 15:09.23 | syle | either was i need a colo suggestion |
| 15:09.30 | syle | s/was/way |
| 15:09.40 | JT | where? |
| 15:09.52 | [TK]D-Fender | javb: config files picked up via ftp/http/tftp, etc |
| 15:10.05 | orakle | you write the config file on the server and the phone just goes and picks it up |
| 15:10.10 | [TK]D-Fender | javb: Go download the admin guide and get the firmware from your reseller |
| 15:10.11 | orakle | i do it with my ciscos |
| 15:10.12 | syle | any good spot in the US is fine |
| 15:10.16 | syle | good backbone |
| 15:10.20 | s0ck | are you uk syle |
| 15:10.28 | JT | i'm guessing it's not for voip then |
| 15:10.52 | Dan0maN_Work | javb: http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf |
| 15:10.59 | syle | nope, i do develop in asterisk tree time to time, but this server is for database only stuff |
| 15:11.14 | *** join/#asterisk ai-a (n=jake2@megan.healthnet.co.uk) |
| 15:11.15 | JT | he.net is fairly decent from what i can tell |
| 15:11.27 | s0ck | vi.net i was looking at the other day, seems alright |
| 15:11.27 | syle | hurricane? |
| 15:11.52 | toot | can i ask - are most people using asterisk for 1-5 or bigger installs? - reason being we are going to release a freeware version of tigercube shortly.. |
| 15:11.57 | syle | yeah thats same backbone i have now, i think they probably doubling rates to if same area |
| 15:12.23 | syle | california can go to hell lol, i just want to move it |
| 15:13.13 | syle | i currently living in central canada, so central US prob be good |
| 15:13.26 | JT | syle: yes, HE is one of the biggest players around |
| 15:13.36 | JT | latency to australia is excellent :) |
| 15:13.40 | orakle | HE is awesome |
| 15:13.44 | orakle | used to have a box there |
| 15:14.52 | JT | syle: you don't want california? |
| 15:14.55 | [TK]D-Fender | toot: Yes, plenty of people. Do you guys have a commercial license with Digium already? |
| 15:14.58 | JT | it's very well connected |
| 15:15.27 | s0ck | anyone running 6.5.12 on a snom360 |
| 15:15.55 | [TK]D-Fender | toot: And since you already seem to have made this solution, shouldn't you already know the answer to this? |
| 15:16.16 | javb | Mmm, if my phone sends 10 with Asterisk 1.4 and sends 102 with asterisk 1.2 (both cases dialing 102) ... isnt this out of the question? |
| 15:16.53 | syle | it don;t matter to me where it is |
| 15:17.00 | syle | just somewhere in US |
| 15:17.19 | syle | low hop count preferably |
| 15:17.24 | mildk | s0ck: yes |
| 15:17.28 | toot | heh - we have a good idea, but just wondered what people in the channel were mostly doing |
| 15:17.57 | syle | yeah he is like 1 hop from me right now |
| 15:18.02 | syle | he.net |
| 15:18.25 | [TK]D-Fender | toot: the answer is "not using or making GUI's and trying to sell their solutions". |
| 15:18.28 | [TK]D-Fender | :) |
| 15:18.47 | JT | syle: that puts them pretty close by, like in your building |
| 15:19.07 | syle | think one of their rackspaces is a building away |
| 15:19.10 | [TK]D-Fender | toot: For which I certainly hope you either abide the Digium commercial license or are providing the full source with your solution... |
| 15:19.44 | JT | syle: you live in california? |
| 15:19.45 | toot | heh - we are in close and very friendly ongoing discussions with Digium |
| 15:19.52 | syle | nope |
| 15:19.54 | toot | we are also good and active open source people :P |
| 15:20.34 | ai-a | PABX or PBX ? |
| 15:20.40 | syle | in canada, which is alright, US dollar sucks ass anyways |
| 15:21.21 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:21.21 | ai-a | "Private Automatic Branch eXchange" or "Private Bbranch eXchange" ? |
| 15:21.36 | s0ck | mildk: blf working fine? |
| 15:21.43 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:21.52 | syle | which sucks for me, their president keeps spending millions a day on crap in iraq, dude has 50% shares in oil stocks not hard to see the scam hes pullin |
| 15:22.07 | [TK]D-Fender | ai-a: http://acronyms.thefreedictionary.com/PABX |
| 15:22.09 | JT | ai-a: a matter of location really |
| 15:22.13 | [TK]D-Fender | ai-a: "All of the above" :p |
| 15:22.21 | JT | syle: they have a datacentre in canada? |
| 15:22.37 | [TK]D-Fender | ai-a: the "A" is a WORTHLESS addition to "PBX" |
| 15:22.47 | ai-a | A as in Asterisk ? |
| 15:22.48 | syle | problem is i cater to http requests for US people |
| 15:22.49 | ai-a | lol. |
| 15:22.54 | badcfe | i use the Read application and it only receives the first few (randomly) digits i type |
| 15:23.13 | syle | thats why its there and generally cheaper colos in the US then canada, they want an arm and a leg for bandwidth etc here |
| 15:23.17 | JT | syle: the only datacentres i knew of were fremont, CA and San Jose, CA |
| 15:24.27 | badcfe | formulated as a question: "i use the Read application and it only receives the first few (randomly) digits i type, anyone has a hint about why it doesnt get all the digits upto the # i type ? |
| 15:24.52 | syle | theres tons, people who own the US pipe line are from texas, nice pipes around there |
| 15:25.23 | [TK]D-Fender | badcfe: pastebin the CLI output of your attempt at verbose 10 with DTMF debug |
| 15:25.36 | mildk | s0ck: yes, but i think version 7 is better |
| 15:25.40 | JT | syle: he.net |
| 15:25.44 | badcfe | [TK]D-Fender: how do i enable dtmf debug? |
| 15:25.45 | mildk | s0ck: it handles resubscriptions if asterisk is restarted |
| 15:27.49 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) |
| 15:27.51 | [TK]D-Fender | badcfe: "set debug 10" oughtta do |
| 15:30.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:30.57 | badcfe | [TK]D-Fender: history is ive tried rfc2833, info (ofcourse in the sender gw in parallell), and i monitor with wireshark, ngrep, asterisk cli and the gw log. arg! (thanks for helping, ill pastebin the cli output of an attempt now) |
| 15:32.58 | khronos | Anybody have suggestions for compnies I can call to customize a few servers? |
| 15:34.05 | badcfe | [TK]D-Fender: i dont get dtmf debug on the cli |
| 15:34.51 | [TK]D-Fender | khronos: www.ibm.com , www.dell.com , www.hp.com Have fun! |
| 15:35.14 | [TK]D-Fender | badcfe: pastebin what you DO get and make a small IVR to validate your DTMF. |
| 15:35.20 | badcfe | ok |
| 15:37.55 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 15:39.10 | badcfe | [TK]D-Fender: ok, here i have my extensions.conf sip.conf and the cli output with debug 10: http://pastebin.ca/738690 |
| 15:40.14 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 15:42.21 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:42.53 | badcfe | [TK]D-Fender: hmm, i have an interesting log done with logger to get the dtmf |
| 15:43.27 | badcfe | [TK]D-Fender: check this out .. http://pastebin.ca/738700 |
| 15:44.45 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 15:45.01 | duki | hello, I have just suscribed to iptel.org and got my very new sip address. Is there any test services like FWD? How can I do some testing with iptel? My sip account is now configured in asterisk. |
| 15:45.50 | TrentCreek | try looking on their web site for a tets |
| 15:47.30 | duki | TrentCreek: Yes I were there but cannot find what I am looking for. I go there once again ... |
| 15:50.09 | [TK]D-Fender | badcfe: very interesting... |
| 15:50.44 | [TK]D-Fender | duki: If you don't know how to test it, what are theys upposed to be doing for you in the first place? |
| 15:51.21 | badcfe | [TK]D-Fender: for info its asterisk 1.4.2 |
| 15:51.55 | [TK]D-Fender | badcfe: That already doesn't bode well for you... what version? |
| 15:52.07 | [TK]D-Fender | badcfe: nvm... |
| 15:52.19 | [TK]D-Fender | badcfe: Ok, well you might want to consider upgrading... |
| 15:53.22 | badcfe | [TK]D-Fender: hmm, but i think im doing something nasty. i think i had the same problem with another version of asterisk. with wireshark i see some RTP flowing there right before i tap numbers |
| 15:53.32 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.182) |
| 15:53.46 | [TK]D-Fender | badcfe: RTP will flow... taht jsut audio... |
| 15:53.53 | [TK]D-Fender | badcfe: * doesn't stop that for read... |
| 15:54.22 | duki | [TK]D-Fender: I was there and found that it is possible to search for users online/offline, but nothing like echo test for example or time/weather service |
| 15:54.39 | [TK]D-Fender | badcfe: So yuo can ChanSpy them and here them say "What's my friggen PIN again.. I think I'm gonna have to call the BOFH again dammit...." |
| 15:55.09 | Nugget | http://macnugget.org/photos/2007c2s/bofhcar2 |
| 15:55.30 | [TK]D-Fender | :D |
| 15:56.02 | badcfe | [TK]D-Fender: heh, yeah all this is just so i can collect social security pin |
| 15:56.20 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-214-164.dsl.irvnca.pacbell.net) |
| 15:56.35 | UnixDog | ok |
| 15:56.38 | UnixDog | morning |
| 15:56.58 | UnixDog | anyone here know how to convert a mp3 to a wav |
| 15:57.21 | badcfe | [TK]D-Fender: thing is that on my production asterisk installations dtmf _relay_ works perfect (i can see all them security payments authentified actually). this makes me think its the Read app that teases me badly |
| 15:57.51 | badcfe | [TK]D-Fender: Read works for half the times i call it |
| 15:58.36 | badcfe | [TK]D-Fender: on my test computer here (for the Read app test) i have asterisk debian lenny package. maybe its buggy |
| 15:59.07 | nestAr | UnixDog: sox should do it |
| 15:59.21 | UnixDog | hmmok |
| 15:59.26 | UnixDog | I will try |
| 16:00.08 | _x86_ | what would cause asterisk to not hang up a bridged call when an analog user hangs up their leg of the call for over 15 seconds? |
| 16:00.26 | _x86_ | callprogress? busydetect? |
| 16:01.48 | _x86_ | PSTN --(sangoma A20002D-x / POTS line)--> asterisk --(sangoma A102D-x / T1)--> Rhino channel bank --> user |
| 16:03.10 | *** join/#asterisk kmchen (n=kmchen@gar13-4-82-240-99-84.fbx.proxad.net) |
| 16:03.32 | kmchen | hi everybody |
| 16:04.12 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 16:05.44 | kmchen | I have an Asterisk/Ekiga/Debian install. Works fine but the sound is jerky. Is it a pb of codecs ? |
| 16:06.48 | kmchen | anybody there ? |
| 16:06.48 | *** join/#asterisk Kernel_Core (n=I@83.217.236.227) |
| 16:06.55 | [TK]D-Fender | _x86_: how exactly is hanging up? |
| 16:07.08 | [TK]D-Fender | *who |
| 16:07.57 | badcfe | kmchen: yeah. ive tried that. i dont know what it is. what sound does ekiga give you when you try it towards something else |
| 16:08.23 | _x86_ | [TK]D-Fender: analog user |
| 16:08.37 | _x86_ | [TK]D-Fender: off the rhino channel bank |
| 16:09.03 | badcfe | [TK]D-Fender: hmm. is there some workaround of Read. like programming it by hand using extentions with goto and so? |
| 16:09.16 | tzafrir | kmchen, what codecs do you use? |
| 16:09.16 | kmchen | badcfe: never tried to towards anything else |
| 16:09.39 | tzafrir | It could also be an issue with the sound card (yeah, blame the hardware) |
| 16:09.49 | badcfe | [TK]D-Fender: WaitExten is ok for the purpose of receiving exactly 9 digits ? |
| 16:10.23 | [TK]D-Fender | badcfe: No, you could make a small IVR for this however. |
| 16:10.43 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:11.10 | badcfe | tzafrir, kmchen: i tried asterisk with good sound, then (on same computer) i used ekiga to do the call. it was awfull! |
| 16:11.14 | kmchen | tzafrir: I tried alaw, ulaw, speex, gsm and lpc10. Can it be hardware if sound looks good when play on local speekers |
| 16:12.02 | badcfe | [TK]D-Fender: can you give me an example. it would spare me for some frustration -- if you have an example that is |
| 16:12.34 | tzafrir | kmchen, is it on the same LAN? if so: no reason to use any compressed codec (speex, gsm, lpc10) |
| 16:13.13 | [TK]D-Fender | badcfe: Don't have one handy. Initialize a var , collect each digit until a timeout is reached maxlen is reached, or a terminating char is reached. |
| 16:13.33 | tzafrir | kmchen, it does extra work and has lower sound quality |
| 16:13.43 | tzafrir | Asterisk and Ekiga are on the same system? |
| 16:13.47 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:13.48 | kmchen | tzafrir: yes on same LAN but I call out |
| 16:14.13 | UnixDog | ok I cant find a doc on how to convert from mp3 to wav with sox |
| 16:14.17 | tzafrir | what about a test call on your LAN? from ekiga to an echo test on asterisk? |
| 16:14.18 | kmchen | tzafrir: and asterisk and ekiga on same system |
| 16:14.22 | UnixDog | and i am ok the sox site |
| 16:15.05 | kmchen | tzafrir: did not test echo |
| 16:15.31 | kmchen | tzafrir: gonna do it |
| 16:15.45 | tzafrir | or any similar test call |
| 16:19.59 | *** join/#asterisk MacReady (i=efc@ip-62-69-198-115.globalconnect.pl) |
| 16:20.17 | *** part/#asterisk MacReady (i=efc@ip-62-69-198-115.globalconnect.pl) |
| 16:20.49 | *** join/#asterisk MacReady (i=efc@ip-62-69-198-115.globalconnect.pl) |
| 16:22.20 | *** join/#asterisk pepse (n=pepse@71-223-117-66.phnx.qwest.net) |
| 16:26.05 | UnixDog | got it |
| 16:26.57 | *** part/#asterisk psy65535 (n=psy65535@24-205-53-78.dhcp.gldl.ca.charter.com) |
| 16:27.23 | dijungal | hello |
| 16:27.51 | dijungal | how do I rename a recording after an agent is finished with the call? |
| 16:29.24 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 16:30.43 | penguinFunk | anyone care about this: http://www.securityfocus.com/infocus/1862 ? |
| 16:31.23 | *** join/#asterisk syneus (n=syneus@host23-25-dynamic.180-80-r.retail.telecomitalia.it) |
| 16:33.02 | penguinFunk | This attack can be successful even if the remote SIP proxy server requires authentication of user registration, because the SIP messages are transmitted in the clear |
| 16:34.55 | kmchen | tzafrir: do you have simple exemple cause I'm afraid to pass lot of time to make this work. (I'm a real newbie) |
| 16:35.47 | tzafrir | kmchen, there's an echo test extension in the example extensions.conf IVR |
| 16:36.03 | tzafrir | Basically, Echo() does the trick |
| 16:36.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:37.06 | kmchen | tzafrir: tried that but nothing happens: exten => 12,1,echo() and compose 12 |
| 16:37.32 | jcanfield | Does polycom support or have plans to support LLDP? |
| 16:37.47 | *** join/#asterisk Remenic (n=Richard@cc1222307-a.frane1.fr.home.nl) |
| 16:37.57 | Remenic | hi, I have an odd question |
| 16:39.06 | CBU[^_^]M`` | hello... i have read some articles in the internet about connecting Skype to asterisk.. is it true? |
| 16:39.30 | Remenic | is it possible, in the dialplan, to add a short delay before sending the client "180 Ringing" ? |
| 16:39.40 | Remenic | I need to to catch a little bug in a softphone |
| 16:41.18 | *** join/#asterisk kmchen (n=kmchen@gar13-4-82-240-99-84.fbx.proxad.net) |
| 16:41.44 | [TK]D-Fender | jcanfield: LLDP? |
| 16:41.50 | keith4 | Remenic: does Wait() not work? |
| 16:43.00 | jcanfield | [TK]D-Fender: http://en.wikipedia.org/wiki/LLDP-MED |
| 16:43.02 | Remenic | ahhh now why the hell didn't I try that first :P |
| 16:43.02 | [TK]D-Fender | CBU[^_^]M``: All of the ways that exist are non-free and SHIT. Skype is a proprietary protocol so don't expect to see much support for it |
| 16:43.07 | Remenic | keith4: thanks, it does :) |
| 16:43.09 | keith4 | CBU[^_^]M``: http://www.chanskype.com/ |
| 16:43.18 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:44.22 | Qwell | chan_skype is a joke, at best |
| 16:44.22 | keith4 | CBU[^_^]M``: what [TK]D-Fender said still applies, however |
| 16:44.30 | jcanfield | [TK]D-Fender: Basicly, you plug in the phone, it tell the network what it is, based on that it is assigned VLAN, IP info. (Layer 1 discovery) |
| 16:44.53 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:44.58 | GreggB | penguinFunk: yep, and the same thing can happen if someone taps an analog line... I personally wouldn't approach it from the perspective that VoIP is insecure, but more that the protocol could be improved - or maybe you should run your SIP traffic over a VPN. |
| 16:45.06 | [TK]D-Fender | jcanfield: Well Polycom does support VLANs... |
| 16:45.52 | jcanfield | [TK]D-Fender: Yep, but that has to assigned/provisioned. |
| 16:45.59 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 16:46.33 | [TK]D-Fender | jcanfield: Go download the bootrom admin guide and check it out |
| 16:47.08 | jcanfield | [TK]D-Fender: Doing it....LLDP will solve some security issues with bootrom. |
| 16:47.26 | [TK]D-Fender | jcanfield: like? |
| 16:47.50 | jcanfield | [TK]D-Fender: Like anyone can easy get IP and MAC of phone and plug in. |
| 16:48.07 | jcanfield | [TK]D-Fender: As if they were the phone |
| 16:48.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:49.15 | slima | hello, it`s me again, I still have a problem with picking 3 digit interial number. Asterisk stops at the first digit and search for exten '1', the same configuration works well with other operator, but i`d really like it to work with this operator as well, my config and debug: http://pastebin.com/m7fcd042b any suggestions? |
| 16:49.49 | GreggB | penguinFunk: you might also check out RFC 3261 (basically SIP over TLS), I believe many of the LinkSys SIP devices support this. |
| 16:52.29 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:54.55 | _x86_ | grr |
| 16:54.57 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 16:55.18 | _x86_ | if I Read() or Authenticate() it sees the button i press twice |
| 16:55.27 | _x86_ | why would it be doubling the DTMF? |
| 16:56.23 | [TK]D-Fender | _x86_: maybe * is picking up OOB & IB simultaneously. |
| 16:57.37 | De_Mon | slima the phones are using different extension pattern matching |
| 16:57.47 | *** join/#asterisk marcan (i=1337@host214-134.cvd.fit.edu) |
| 16:57.50 | De_Mon | slima fix the phone |
| 16:58.11 | GreggB | penguinFunk: Looks like SIP over TLS is under development in * right now too: http://bugs.digium.com/view.php?id=4903 |
| 16:59.02 | _x86_ | [TK]D-Fender: hmm... i have dtmfmode=auto |
| 16:59.12 | [TK]D-Fender | _x86_: PICK ONE :p |
| 16:59.25 | _x86_ | [TK]D-Fender: i want OOB |
| 16:59.45 | _x86_ | [TK]D-Fender: 2833 it is? :P |
| 16:59.47 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 16:59.53 | [TK]D-Fender | _x86_: I want a million dollars... nobody's leaving happy it seems..... |
| 17:01.11 | _x86_ | [TK]D-Fender: dtmfmode=rfc2833 did not solve the issue... |
| 17:01.21 | _x86_ | will try dtmfmode=inband |
| 17:01.42 | slima | De_Mon: what phone? i called mobile phone -> my sip number (sip operator) -> asterisk [mainmenu] and i try to dial '100' |
| 17:01.49 | slima | called by* |
| 17:01.55 | linagee | [TK]D-Fender: i want that too! |
| 17:02.23 | mosty | is it considered valid to want to be able to dial 1800-SOMEHUGELONGPIECEOFTEXT and expect a PBX to only use the first N digits? |
| 17:02.32 | linagee | [TK]D-Fender: although you have to admit, a million dollars in 2007 money is not much. it will just buy you a large, adequate house. |
| 17:02.51 | De_Mon | slima ok, and under what conditions does it work as desired? |
| 17:02.53 | [TK]D-Fender | linagee : Only an idiot would go and buy a house with it.... |
| 17:03.04 | _x86_ | [TK]D-Fender: does not matter if i set dtmfmode=rfc2833 or dtmfmode=inband, same results as dtmfmode=auto |
| 17:03.16 | _x86_ | [TK]D-Fender: i tried relaxdtmf=no as well as =yes, no difference |
| 17:03.18 | hmmhesays | a million bucks, I could live on that for a long ass time |
| 17:03.27 | _x86_ | [TK]D-Fender: progressinband is no |
| 17:03.27 | [TK]D-Fender | _x86_: verify where you are setting thi and that its in the appropriate place.... |
| 17:03.28 | linagee | hmmhesays: on ramen noodles? :) |
| 17:03.40 | linagee | hmmhesays: buy like a million noodle cans |
| 17:03.42 | _x86_ | [TK]D-Fender: iax.conf and sip.conf in [general] |
| 17:03.45 | [TK]D-Fender | hmmhesays: I could make it last a lifetime. |
| 17:04.14 | _x86_ | you could easily live off the interest of a million dollars |
| 17:04.19 | jcanfield | hmmm...or forever if you are smart. http://en.wikipedia.org/wiki/Time_value_of_money |
| 17:04.21 | [TK]D-Fender | _x86_: exactly |
| 17:04.24 | mosty | do any businesses add unused digits to the end of their advertised numbers? |
| 17:04.34 | _x86_ | standard savings account interest would be well over 50,000$/yr |
| 17:05.00 | hmmhesays | I get 4% on my savings account |
| 17:05.03 | [TK]D-Fender | _x86_: wHAT SAVINGS ACCOUNT GIVE YOU 5%? |
| 17:05.09 | _x86_ | but with that kind of money, you can do money market and all kinds of crazy shit, where you can get triple digit returns every year |
| 17:05.15 | _x86_ | [TK]D-Fender: MINE DOES |
| 17:05.17 | hmmhesays | well 3.8723423 something |
| 17:05.22 | slima | De_Mon: when I call: mobile phone -> *other* sip operator -> asterisk [mainmenu] |
| 17:05.29 | [TK]D-Fender | _x86_: good stuff... |
| 17:05.39 | linagee | [TK]D-Fender: i could make $160,000 last a lifetime |
| 17:05.40 | mosty | [TK]D-Fender, not everyday savings accounts. online banking accounts (from major banks) are often above 5%pa |
| 17:05.50 | _x86_ | [TK]D-Fender: any ideas on my DTMF handling issue? |
| 17:05.55 | [TK]D-Fender | _x86_: nope |
| 17:06.01 | linagee | [TK]D-Fender: i could life off the interest of $160,000 |
| 17:06.11 | linagee | s/life/live/ |
| 17:06.18 | linagee | jbot: shh |
| 17:06.18 | jbot | QUIET! |
| 17:06.44 | kmchen | tzafrir: Tested with xlite from another PC on LAN. Sounds ok. So it seems to be an Ekiga problem |
| 17:06.46 | *** join/#asterisk matsk (n=mk@host-217-213-138-53.mobileonline.telia.com) |
| 17:06.57 | linagee | mosty: try 22%, prosper. ;) |
| 17:07.18 | mosty | linagee, that's quite a bit riskier than a traditional bank |
| 17:07.23 | linagee | mosty: true |
| 17:07.48 | *** join/#asterisk captiancrash (n=jmoore@70.159.118.70) |
| 17:07.51 | linagee | mosty: what if i just put money in there that i wouldn't mind losing anyway? like money that i might have otherwise taken to vegas and lost anyway? |
| 17:09.14 | kmchen | Does anyone know how to get correct sound with Ekiga ? |
| 17:09.15 | mosty | linagee, you go to vegas for fun, not to make money. prosper isn't as fun |
| 17:09.22 | linagee | mosty: it's fun to me. :) |
| 17:09.30 | mosty | well enjoy |
| 17:09.37 | linagee | mosty: vegas is actually dumb fun |
| 17:09.49 | hmmhesays | vegas is a lot more fun if you are winning |
| 17:09.59 | linagee | mosty: put money into a computer, computer picks a random number, deposits money when random number hits other number |
| 17:10.04 | hmmhesays | I get my A200 today |
| 17:10.06 | mosty | linagee, i don't find losing money fun |
| 17:10.09 | hmmhesays | I dn't play computer games |
| 17:10.14 | hmmhesays | cards and dice only |
| 17:10.24 | linagee | hmmhesays: every game at vegas is being replaced with a computer game |
| 17:10.28 | mosty | poker machines are just plain retarded |
| 17:10.42 | hmmhesays | live dealers will never be replaced completely |
| 17:10.58 | linagee | hmmhesays: depends on the cost/benefit ratio to the large business owner at the top. ;) |
| 17:11.24 | *** join/#asterisk BrokenNoze (n=root@host81-149-254-218.in-addr.btopenworld.com) |
| 17:11.32 | *** join/#asterisk dez71 (i=dez@216.83.0.172) |
| 17:12.41 | BrokenNoze | Hi, I've hagin an issue with SIPGate, inbound calls seem to work fine, then for no apparent reason they stop working. the next time i get a SRV mapped to host sipgate.co.uk, port 5060 event, they work again, anyone any ideas? |
| 17:13.16 | UnixDog | polycoms rule |
| 17:13.24 | linagee | UnixDog: yes |
| 17:13.41 | linagee | UnixDog: just make 2.2.0 boot up faster and i will be even more happy. ;) |
| 17:13.58 | UnixDog | I have 501/601+sidecar and a 650 in route |
| 17:14.35 | UnixDog | I just wish they would have back light the units |
| 17:14.39 | hmmhesays | I just provisioned my first 601 with a sidecar |
| 17:14.44 | [TK]D-Fender | linagee : Boots plenty fast for me, and since it doesn't NEED rebooting, I could not care less... |
| 17:14.45 | hmmhesays | yeah backlights are the only thing I can see missing |
| 17:14.59 | [TK]D-Fender | IP 550/650 have backlight |
| 17:15.08 | linagee | [TK]D-Fender: boots slower than previous version though, which i find mysterious. (did they leave debugging symbols on or something?) |
| 17:15.24 | UnixDog | well update the firm ware |
| 17:15.24 | [TK]D-Fender | linagee : my IP 501 takes 1:45 to boot SIP 2.2.0 |
| 17:15.28 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 17:15.30 | UnixDog | 2.2.0 is out |
| 17:15.31 | slima | [TK]D-Fender: its me agan.. I still have a problem with picking 3 digit interial number. Asterisk stops at the first digit and search for exten '1', the same configuration works well with other sip operator, but i`d really like it to work with this sip operator as well, my config and debug: http://pastebin.com/m7fcd042b |
| 17:15.36 | linagee | [TK]D-Fender: my 320 takes about 3-5 minutes |
| 17:16.12 | [TK]D-Fender | slima: I have nothing more I can add on this. Did you try "relaxdtmf=yes" for your channel's entry? |
| 17:16.12 | UnixDog | 3.2.3 boot rom and 2.2.0 firmware works great |
| 17:16.20 | UnixDog | takes less then a min to boot |
| 17:16.32 | linagee | [TK]D-Fender: maybe a 320 is a lower end model, but still, it booted 2.1.1 just plenty fast. like 1-2 minutes |
| 17:16.42 | UnixDog | I need a 330 next |
| 17:16.51 | UnixDog | I want 1 of each major model |
| 17:16.57 | UnixDog | to better support them |
| 17:17.15 | linagee | UnixDog: i have not upgraded the boot rom. maybe that's why it boots slow. (can't find a source) |
| 17:17.30 | UnixDog | hold on |
| 17:17.41 | slima | [TK]D-Fender: yes, and i`m changing dtmfmode= noting work... |
| 17:17.46 | linagee | UnixDog: i'm running 3.2.3 boot rom |
| 17:17.51 | linagee | UnixDog: there's a newer one out afaik |
| 17:18.17 | slima | nothing works* |
| 17:18.53 | [TK]D-Fender | slima: Ok, then i'm out of ideas |
| 17:18.58 | UnixDog | 3.2.3 rev b is the latest I know of |
| 17:18.58 | jcanfield | [TK]D-Fender: Just talked to polycom SE, LLDP is in beta should be a firmware upgrade. |
| 17:19.24 | [TK]D-Fender | UnixDog: BootrROM 4.0 <- |
| 17:19.28 | UnixDog | where |
| 17:19.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:19.40 | [TK]D-Fender | UnixDog: Go ask your reseller for it |
| 17:20.05 | UnixDog | lol but all mine fall off the back of truks |
| 17:20.08 | UnixDog | lol joking |
| 17:20.48 | linagee | UnixDog: 4.0.0 is latest. |
| 17:20.50 | linagee | UnixDog: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html |
| 17:21.25 | linagee | weird. the 4.0.0 release document is for 3.2.3 |
| 17:22.16 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 17:22.51 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.182.176) |
| 17:22.57 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 17:28.42 | UnixDog | ok well now I know soomething is wrong for a week nowe i can not reach ipphone-warehouse.com no one is answering the phones |
| 17:28.49 | UnixDog | I want the new bootrooom |
| 17:28.55 | Qwell | sure sounds like they died |
| 17:28.58 | *** join/#asterisk synthetiq (i=walletje@53516DE0.cable.casema.nl) |
| 17:29.00 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 17:29.09 | Qwell | and with such a great name like "ipphone-warehouse", I can't imagine why |
| 17:29.17 | UnixDog | I want the new bootrom and I want to see about a clients phone thats on order I have not recieved |
| 17:29.33 | BrokenNoze | why would a vip provider work one minute, then fail the next for an Inbound call? I have quality=yes, shouldn't that fix the problem? |
| 17:29.34 | UnixDog | they have been around for 3 years |
| 17:29.47 | UnixDog | nat issues |
| 17:29.53 | UnixDog | network issues |
| 17:29.53 | Qwell | UnixDog: 3 year old companies can't abandon their customers, and take your money and run? |
| 17:29.55 | UnixDog | routing |
| 17:30.14 | UnixDog | no they cant its not permited |
| 17:30.22 | UnixDog | lol |
| 17:30.38 | BrokenNoze | UnixDog: would putting in DMZ fix the problems then? |
| 17:31.00 | UnixDog | no that would make it more hacable |
| 17:31.07 | UnixDog | hackable |
| 17:31.24 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:31.43 | UnixDog | just open the ports 5060 - 5066 and ports 10000-20000 udp |
| 17:32.06 | BrokenNoze | ok |
| 17:32.21 | keith4 | 5066? |
| 17:32.41 | UnixDog | and set localnet=x.x.x.x/x.x.x.x extenip extenrnal ip and nat = yes ip sip.cfg |
| 17:32.53 | keith4 | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:32.54 | UnixDog | ok just 5060 |
| 17:33.17 | BrokenNoze | mm, i have done that |
| 17:33.38 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 17:34.13 | UnixDog | ok now I have to find where to get the bootrom since I cnat reah ipphone-warehouse |
| 17:35.28 | UnixDog | brb |
| 17:37.12 | *** join/#asterisk mikealeonetti (n=mikel@ool-457b736e.dyn.optonline.net) |
| 17:38.00 | mikealeonetti | let's say I want to set up Asterisk in my business environment, do I use analog lines or can I sign up with a Telephone company that will give me digital phone lines? I'm not sure how it works. |
| 17:38.29 | Qwell | mikealeonetti: how many lines do you have? what country are you in? |
| 17:38.31 | mosty | mikealeonetti, you can do either, or both |
| 17:38.34 | khronos | It would depend on the type of calls you'll be making. |
| 17:38.35 | Qwell | s/have/need/ |
| 17:38.54 | khronos | If you do primarily local calls land lines might be better. |
| 17:39.05 | khronos | Lots of ld calling internet based would probably be better. |
| 17:39.18 | mikealeonetti | I have four lines and and I'm in north america |
| 17:40.14 | UnixDog | get a tdm400 |
| 17:40.18 | UnixDog | plugh it in |
| 17:40.31 | UnixDog | and connect your phone lines to it |
| 17:40.38 | mikealeonetti | what if I wanted to expand into about 16 lines? |
| 17:40.45 | UnixDog | get 4 voip phones like polycoms |
| 17:41.00 | UnixDog | then you get a pri card |
| 17:41.07 | UnixDog | and go that route |
| 17:41.22 | khronos | Either that or get a sip gay to plug the analog lines in to. |
| 17:41.23 | mikealeonetti | and who do I register with to get my phone numbers? |
| 17:41.44 | khronos | Ah, anything over 8 you'll need a pri. |
| 17:41.51 | UnixDog | there is alot you need to learn youn man |
| 17:41.54 | UnixDog | ?book |
| 17:42.14 | UnixDog | you need the book |
| 17:42.21 | UnixDog | !thebook |
| 17:42.22 | mikealeonetti | the book? |
| 17:42.36 | [TK]D-Fender | ~book |
| 17:42.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:42.38 | UnixDog | asterisk the future of telepohny |
| 17:43.08 | UnixDog | there is alot you need to learn grasshoper |
| 17:43.16 | mikealeonetti | well of cousre |
| 17:43.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:45.27 | mikealeonetti | this book will tell me everything I need to know? |
| 17:45.32 | khronos | Yes. |
| 17:45.41 | mikealeonetti | it better, or I'll be back |
| 17:46.05 | khronos | Evne if you have questions you'll be in a much better position of what to askand you'll have a good understanding of how things work. |
| 17:46.19 | mikealeonetti | it's not a really long book is it? |
| 17:46.42 | khronos | Depends on what you classify as long. |
| 17:46.57 | mikealeonetti | ~30 pages |
| 17:47.07 | khronos | Will you learn asterisk in a day? No, it does take some time to learn, but it is well worth learning about. |
| 17:47.23 | khronos | Longer than 30 yes. |
| 17:47.35 | mikealeonetti | man that's HUGE |
| 17:49.28 | *** join/#asterisk iratik (n=itariki@adsl-70-248-216-14.dsl.spfdmo.swbell.net) |
| 17:49.29 | [TK]D-Fender | mikealeonetti: Here is a quick guide to help you start : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 17:49.31 | iratik | I'd like to setup the gizmo sipphone with AsteriskNow ... I have an sipphone.com account -- I'm at the asterisk setup in the add voip phone section ... what do I do? |
| 17:49.44 | mikealeonetti | [TK]D-Fender: thanks much |
| 17:50.08 | [TK]D-Fender | iratik: You go to #asterisknow for support, because this isn't the channel for that. |
| 17:50.22 | iratik | i've been there for a while .. thanks tho |
| 17:50.47 | [TK]D-Fender | mikealeonetti: Not 100% applicable to you but gives you an idea how little it could take to set up a system. You can then expand on it to let you do all sorts of other stuff as you go along. |
| 17:51.21 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 17:56.18 | stimpie | does someone have a method to store the ${HANGUPCAUSE} in a cdr? |
| 17:57.15 | stimpie | I use Set(CDR(hangupcause)=$HANGUPCAUSE) but this creates strange entries in the userfield |
| 17:57.30 | *** join/#asterisk angom (n=angom@201.143.89.82) |
| 17:57.54 | *** join/#asterisk soulfreshner (n=Derick@dsl-243-57-187.telkomadsl.co.za) |
| 17:59.09 | soulfreshner | my flash operator panel has all the lights flashing... the configs are ok as far as I can tell |
| 17:59.17 | soulfreshner | what else do I need to check |
| 17:59.30 | soulfreshner | ? |
| 18:00.41 | [TK]D-Fender | soulfreshner: Could your description and backup be any more vague? |
| 18:01.37 | soulfreshner | [TK]D-Fender, I'm sure it could - but I thought you'd like it as descriptive as possible :P |
| 18:01.48 | [TK]D-Fender | soulfreshner: you = failure |
| 18:02.03 | soulfreshner | the FOP has all the lights flashing |
| 18:02.12 | soulfreshner | the little led thingies |
| 18:02.24 | soulfreshner | TK - that's not very nice... |
| 18:02.44 | [TK]D-Fender | soulfreshner: We can't see ANY of your configs so we have no idea what "all the lights flashing" represents exactly. |
| 18:02.48 | [TK]D-Fender | soulfreshner: PASTEBIN it all. |
| 18:02.50 | [TK]D-Fender | ~pb |
| 18:02.51 | jbot | A Pastebin is a place to paste your stuff 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 |
| 18:02.52 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:02.54 | soulfreshner | the configs are set like the defaults... |
| 18:03.11 | [TK]D-Fender | soulfreshner: You know what... I'd bet that if everything was right... it would WORK. |
| 18:03.20 | soulfreshner | i didn't configure the layout yet - so the lights are the default ones |
| 18:03.32 | [TK]D-Fender | soulfreshner: But clearly it isn't working and we should have absolutely no faith in their correctness. |
| 18:04.29 | soulfreshner | ok |
| 18:04.36 | soulfreshner | pasting the stuff... |
| 18:06.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:06.28 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:07.37 | *** part/#asterisk ylon (n=ylon@cpe-76-181-182-10.columbus.res.rr.com) |
| 18:10.17 | *** join/#asterisk ACiDV (n=acidv@97-147.dr.cgocable.ca) |
| 18:10.36 | *** join/#asterisk lemanal (n=lemanal@ip68-14-106-198.no.no.cox.net) |
| 18:11.09 | ACiDV | not sure if this is an asterisk-dev or not question, but if someone can help me =) Does exist a way to confirm that an Asterisk has been build w/ DEBUG_THREAD and DONT_OPTIMIZE ? |
| 18:12.02 | hmmhesays | anyone know of a telnet client that will rotate log files once the file gets too big? |
| 18:12.40 | Nugget | "telnet client"? |
| 18:12.44 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 18:12.46 | soulfreshner | here is the paste : http://pastebin.com/m507a36fe |
| 18:13.00 | mosty | telnet clients do telnet, not logging, generally |
| 18:13.41 | Nugget | do you mean "command line app" when you say "telnet client"? |
| 18:13.46 | kratzers | dumb problem... * stops playing MoH when the destination channel is ringing... |
| 18:14.03 | [TK]D-Fender | soulfreshner: guess what... nothing of any valoue in there. |
| 18:14.12 | *** join/#asterisk syneus (n=syneus@host23-25-dynamic.180-80-r.retail.telecomitalia.it) |
| 18:15.24 | Qwell | Nugget: where's your telnet trigger? |
| 18:15.57 | [TK]D-Fender | Qwell: its on a 24h repeat timer |
| 18:16.03 | Qwell | lame |
| 18:16.51 | kratzers | any ideas? |
| 18:18.51 | kratzers | we have MoH between calls to Dial(), but not when the destination is actually ringing |
| 18:19.39 | [TK]D-Fender | kratzers: pastebin it at verbose 10 |
| 18:19.46 | [TK]D-Fender | ~pb |
| 18:19.55 | jbot | A Pastebin is a place to paste your stuff 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 |
| 18:20.25 | kratzers | heh, we've got so much crap that a 10 second pastebin would probably be a few hundred lines |
| 18:23.31 | [TK]D-Fender | kratzers: thena few hunderd lines it is... |
| 18:23.37 | kratzers | eesh, ~5 seconds of output is > 550 lines |
| 18:23.39 | kratzers | here goes |
| 18:24.39 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 18:26.30 | stimpie | how do make sure an extension is reached when the dial command finishes? |
| 18:29.11 | [TK]D-Fender | stimpie: ....huh?! |
| 18:29.20 | *** join/#asterisk pkwong (n=chatzill@68.195.200.20) |
| 18:30.06 | pkwong | hi all.. quick question.. is there a reason why the 7970s don't do call transfer? |
| 18:30.20 | Qwell | pkwong: chan_skinny? |
| 18:30.23 | pkwong | sip. |
| 18:30.26 | stimpie | might be me being stupid, lets say I have: exten => s,2,Dial(100@test) |
| 18:30.31 | Qwell | then nope |
| 18:30.39 | pkwong | lovely |
| 18:30.41 | [TK]D-Fender | stimpie: You're right... that Dial IS completely stupid... |
| 18:30.55 | stimpie | just an example |
| 18:31.07 | kratzers | probably nothing relevant but the last two lines between which MoH stops |
| 18:31.11 | [TK]D-Fender | stimpie: A bad one and a warning of things to come I'm sure... continue :p |
| 18:31.12 | kratzers | http://pastebin.com/d78bf275 |
| 18:31.42 | synthetiq | are there any good sites on debug asterisk voicemail realtime? I dont think asterisk knows about using odbc |
| 18:31.47 | stimpie | I mean how do I make sure exten => s,3,set(CDR) is reached always |
| 18:32.00 | synthetiq | err well the odbc conenction is there but the statments are not being sent out |
| 18:32.08 | [TK]D-Fender | kratzers: Waitasec.. are these Queue members? |
| 18:32.13 | kratzers | yeah |
| 18:32.22 | [TK]D-Fender | kratzers: you should let the QUEUE mange MoH... |
| 18:33.20 | [TK]D-Fender | stimpie: It will unless the call is ANSWERED |
| 18:33.49 | kratzers | [TK]D-Fender: Why not? It seems to be designed to do so. |
| 18:34.09 | [TK]D-Fender | stimpie: thent he call will only continue the next priority if you use the "g" option and the CALLEE hangs up. For the other case, yuo'll need the "h" Standard Extension. |
| 18:34.13 | kratzers | given the configuration options available in queues.conf |
| 18:34.26 | [TK]D-Fender | kratzers: For the exact reasons I wrote above, the answer is NO. |
| 18:34.37 | [TK]D-Fender | kratzers: This has been the way * was designed from the start. |
| 18:34.55 | [TK]D-Fender | kratzers: nix that, bad aim |
| 18:34.59 | [TK]D-Fender | :/ |
| 18:35.04 | pkwong | so the 7970 won't do call transfers with sip, huh? via the transfer button? |
| 18:35.09 | kratzers | oh, sorry... we Are letting the queues manage it |
| 18:35.10 | kratzers | ok |
| 18:35.14 | pkwong | it was working with 802SR1. |
| 18:35.31 | [TK]D-Fender | kratzers: remove the "m" from your dials and let the Queue handle MoH. |
| 18:35.43 | [TK]D-Fender | kratzers: they are probably fighting. |
| 18:35.44 | kratzers | no m in the dials |
| 18:35.51 | kratzers | M for Macro, not no m for music on hold |
| 18:36.25 | [TK]D-Fender | kratzers: -- Executing [423@agents:21] Dial("Local/423@agents-53ce,2", "SIP/423||mM(setup_chaninfo^423)") in new stack |
| 18:36.25 | [TK]D-Fender | YES, BOTH |
| 18:36.25 | kratzers | blah |
| 18:36.25 | pkwong | does anyone know if there are plans to supportit? |
| 18:36.25 | [TK]D-Fender | :p |
| 18:36.27 | pkwong | errr.. make it work? |
| 18:36.40 | kratzers | I wonder when that snuck in |
| 18:37.21 | kratzers | ah, it's the same both ways |
| 18:37.32 | kratzers | that was just added 10 minutes ago to see if it changed anything |
| 18:38.12 | kratzers | I'm thinking it's an * bug since it worked before we upgraded to 2.4.11 and then 2.4.13 |
| 18:40.10 | tzafrir | 2.6.11, 2.6.13, right? |
| 18:41.13 | kratzers | yeah, sorry |
| 18:41.32 | kratzers | actually 1.4.11 and 1.4.13 |
| 18:41.36 | kratzers | asterisk version, not Linux |
| 18:41.37 | [TK]D-Fender | .... |
| 18:43.03 | jarrod | anyone know how to permit a polycom to receive more than 2 calls without giving busy when using asterisknow? |
| 18:46.02 | s0ck | bloody snom ;/ |
| 18:46.21 | kratzers | jarrod: increase call-limit in sip.conf? |
| 18:47.29 | pkwong | so no plans to make the 7970 work with the transfer button? |
| 18:47.52 | Kwakwa | jarrod: which polycom phone do u have? |
| 18:47.56 | *** join/#asterisk Uatec (n=Uatec@77.241.176.36) |
| 18:48.11 | Uatec | Evening |
| 18:48.14 | Kwakwa | jarrod: also, have you set a call-limit? |
| 18:48.25 | pkwong | anyone wanna take on a bounty then? |
| 18:48.37 | [TK]D-Fender | jarrod: You're in the wrong channel, please read the topic. |
| 18:49.31 | Uatec | when i try to dial a number from my sip phone i get: Unable to lookup host in c= line, 'IN IP4 xxx.241.176.3607' |
| 18:49.33 | Uatec | and nothign else happens |
| 18:49.38 | Uatec | why is it mangling my IP address? |
| 18:49.59 | kratzers | looks like the whole A record |
| 18:50.55 | kratzers | or not |
| 18:51.00 | Uatec | you talking to me? |
| 18:51.04 | Uatec | becuase i'm not using DNS |
| 18:51.45 | hmmhesays | sorry, I meant a telnet client that can log the output of the telnet session |
| 18:51.52 | kratzers | yeah, I'm being stupid |
| 18:52.27 | mosty | hmmhesays, script |
| 18:52.39 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-92-213-14.dsl.hstntx.swbell.net) |
| 18:52.48 | kratzers | Uatec: what is the correct value for the final octet? |
| 18:52.52 | Uatec | 36 |
| 18:53.18 | [TK]D-Fender | Uatec: Guess your phone has a bug. What is it exactly? |
| 18:53.33 | Uatec | a snom 190 |
| 18:53.46 | Uatec | i've used the phone on a lan and it worked fine |
| 18:53.47 | [TK]D-Fender | Uatec: You could say I'm less than surprised. |
| 18:53.49 | Uatec | earlier today |
| 18:54.33 | Uatec | now, i'm using it on my home lan, connection across the internet |
| 18:54.37 | Uatec | there IS nat involved |
| 18:55.26 | Uatec | but this is just bizarre, beyond anything i know and have read about with nat |
| 18:55.35 | [TK]D-Fender | Uatec: Any chance you told it your wan IP and typo'd? |
| 18:56.49 | Uatec | i've not told it my wan ip |
| 19:01.24 | soulfreshner | in /var/log/op-panel/error.log I get the error: Failed to open PID file /var/run/op-panel/op-panel.pid for writing. at /usr/sbin/op_server line 325. |
| 19:01.32 | soulfreshner | and the server doesn't start |
| 19:02.08 | soulfreshner | I'm starting the server as root - there shouldn't be permission problems, right? |
| 19:02.18 | Uatec | if i turn on sip debug and that shows the correct WAN IPs of the server and client |
| 19:02.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:03.44 | kratzers | soulfreshner: maybe make sure /var/run/op-panel exists and has proper permissions |
| 19:04.25 | soulfreshner | wouldn't it only exist if the server is running? |
| 19:04.55 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 19:04.56 | Uatec | the pid file would |
| 19:05.02 | Uatec | but the directory might not |
| 19:05.02 | soulfreshner | running the server should create the file, I think - and that's probably the problem - it can't |
| 19:05.13 | hmmhesays | I guess I'll just split the file after I log everything |
| 19:05.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:05.17 | kratzers | it can't if the directory that it's trying to create it in doesn't exist |
| 19:05.21 | kratzers | or has the wrong permissions |
| 19:05.55 | soulfreshner | the directory doesn't exist... |
| 19:06.00 | kratzers | that could be a problem |
| 19:06.27 | Mercestes | http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root |
| 19:07.28 | Mercestes | It is also helpful to temporarily setup the asterisk user to be able to access a shell and run asterisk -cvvvvvvv as asterisk to see exactly where your error is occuring. |
| 19:08.07 | synthetiq | Anyone know a reason why asterisk will not send out sql statements, such as querying a voicemail box? |
| 19:08.13 | Mercestes | asterisk *should* be able to create the /var/run/asterisk directory but if not, mkdir it, chown it to asterisk:asterisk and chmod the 660 permissions. |
| 19:08.20 | synthetiq | yes it says its connected via odbc |
| 19:08.26 | synthetiq | yes=yet |
| 19:08.44 | *** join/#asterisk SparFux (n=raoul@e182025048.adsl.alicedsl.de) |
| 19:08.50 | SparFux | Hi all! |
| 19:08.56 | Mercestes | synthetiq: Is asterisk configured to use res_odbc for voicemail? |
| 19:09.42 | synthetiq | mercestes what do you mean? i have res_odbc.conf set right and extconfig.conf |
| 19:10.21 | Mercestes | synthetiq, http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail |
| 19:10.26 | synthetiq | yes i read that |
| 19:10.28 | SparFux | I have a QUESTION. I use sipgate and on incoming dial it only gives me "everyone is busy / congested at the time" I try to dial out to a capi device. Nobody is hanging on any line. Nobody is busy! How can this be? |
| 19:10.35 | soulfreshner | i created the dir and set the permissions - now I get a new error: |
| 19:10.36 | soulfreshner | Filehandle STDIN reopened as VARIABLES only for output at /usr/sbin/op_server line 1305. |
| 19:10.49 | *** join/#asterisk grandpapa (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 19:10.55 | synthetiq | but there are no debugging steps to find out why its not sending otu the sql even though i configured everythign properly (suppsoedly) |
| 19:11.30 | Mercestes | soulfreshner, I thought you were having problems with asterisk, not FOP. |
| 19:11.38 | grandpapa | Are hints context specific? i.e., if I have a hint for extension 805 in context called MyContext, will only watchers in that context get the hint? |
| 19:12.16 | Mercestes | grandpapa, I believe hints are sip user specific. |
| 19:12.19 | soulfreshner | nope - asterisk is running fine... this is the right place to ask, though - or not? |
| 19:12.34 | Mercestes | soulfreshner, What user are you running FOP as? |
| 19:12.51 | grandpapa | hmm.. ok, thanks, Mercestes. |
| 19:13.20 | soulfreshner | Mercestes, I start it up as root |
| 19:13.38 | Mercestes | soulfreshner, Then my hypothesis is that it is not a permissions issue. |
| 19:14.04 | Mercestes | That's just a guess tho. |
| 19:14.32 | kratzers | soulfreshner: did you create the directory? |
| 19:14.50 | soulfreshner | Mercestes, but I don't think it runs as root for some reason - after changing the permissions on the /var/run/op-panel directory to the asterisk group it moved on to the new error |
| 19:14.56 | soulfreshner | kratzers, yep |
| 19:15.06 | soulfreshner | now I have the new error |
| 19:15.16 | kratzers | sorry, missed that |
| 19:15.38 | soulfreshner | Filehandle STDIN reopened as VARIABLES only for output at /usr/sbin/op_server line 1305. |
| 19:15.40 | Mercestes | how are you executing the server? |
| 19:16.08 | soulfreshner | /etc/init.d/op-server restart |
| 19:16.16 | kratzers | soulfreshner: that sounds like a software bug |
| 19:17.21 | hmmhesays | installing wanpipe da, dad da dada |
| 19:17.50 | soulfreshner | it looks like that error is not fatal - it seems to be running and connecting now |
| 19:18.44 | SparFux | sipgate dialout Dial() causes this message: Everyone is busy/congested at this time (1:0/0/1) |
| 19:18.44 | SparFux | <PROTECTED> |
| 19:18.52 | SparFux | What does the (1:0/0/1) mean? |
| 19:20.00 | *** join/#asterisk dijungal (n=kdaniel@205.244.148.37) |
| 19:20.02 | dijungal | hi |
| 19:20.11 | dijungal | how do I uninstall asterisk |
| 19:20.15 | _x86_ | hahaha |
| 19:20.29 | Mercestes | lol. Nice. |
| 19:20.36 | dijungal | i'm on version 1.4 i want get rid of that and go to 1.2 |
| 19:20.43 | Mercestes | Much better. |
| 19:20.50 | dijungal | :) |
| 19:20.51 | Mercestes | try a make clean in /usr/src/asterisk |
| 19:20.58 | dijungal | yea... i know i was scaring u guys for a moment there |
| 19:21.06 | hmmhesays | that was really easy |
| 19:21.22 | Mercestes | Nah, not at all. I was just contemplating telling you to dd your /dev/urandom onto your root mount point. That's all. |
| 19:21.33 | dijungal | don't understand... |
| 19:21.40 | Mercestes | Oh good, it would've worked then. |
| 19:21.51 | dijungal | lol |
| 19:22.11 | Mercestes | Let's just say feeding your harddrive to a rottweiler would have been less destructive. |
| 19:22.18 | Mercestes | but it would have uninstalled asterisk. |
| 19:22.23 | dijungal | true |
| 19:22.33 | dijungal | so what's the process to downgrade? |
| 19:22.37 | Mercestes | soulfreshner, where did you get a /dev/init.d/op-server script? |
| 19:22.49 | Mercestes | dijungal, make clean && make distclean in /usr/src/asterisk. |
| 19:23.10 | soulfreshner | Mercestes, it's part of the ubuntu package |
| 19:23.22 | dijungal | k |
| 19:23.37 | Mercestes | and if your really wanna get in depth you can rm -dvfr /etc/asterisk /var/lib/asterisk /var/spool/asterisk /var/run/asterisk |
| 19:23.47 | Mercestes | and anywhere else a locate asterisk returns a directory. |
| 19:23.54 | Mercestes | *warning* rm -dvfr is dangerous. use catiously. |
| 19:24.29 | Mercestes | soulfreshner, Yea, I've had some experience with Ubuntu. |
| 19:24.30 | dijungal | i'll use the make clean :) |
| 19:24.40 | Mercestes | dijungal, good man. |
| 19:24.53 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
| 19:25.02 | Mercestes | soulfreshner, well, since it's working we'll just...chalk it up as a victory and be happy. |
| 19:25.03 | dijungal | what about zaptel and lib pri? |
| 19:25.10 | dijungal | libpri |
| 19:25.20 | soulfreshner | Mercestes, that's what I think too |
| 19:25.29 | Mercestes | dijungal, well, if you were in gentoo I would suggest a emerge -Ca asterisk libpri zaptel asterisk-addons asterisk-sounds. |
| 19:25.47 | dijungal | i'm in fedora core 6 |
| 19:25.48 | Mercestes | dijungal, but I'm rarely that lucky so, make clean in /usr/src/zaptel and /usr/src/libpri as well. |
| 19:25.54 | Mercestes | I'm so sorry. |
| 19:26.32 | soulfreshner | I'm not so sure I'll be using FOP - but I thought I'd try it out - didn't expect the package to be a bit wonky, though |
| 19:27.12 | soulfreshner | I already spent more time on it than I really should have |
| 19:27.28 | soulfreshner | you guys all use a graphical interface for clients? |
| 19:28.04 | Mercestes | only custom ones. |
| 19:28.10 | Mercestes | FOP is nice tho. |
| 19:28.28 | dijungal | ok done...so now i should be able to download the 1.2 and install from there |
| 19:28.32 | Mercestes | It's the ubuntu part that is wonky. |
| 19:28.45 | dijungal | and the order is.. zaptel, libpri then asterisk |
| 19:28.46 | Mercestes | dijungal, theoretically. |
| 19:28.48 | dijungal | right? |
| 19:29.01 | Mercestes | dijungal, Umm....yea. |
| 19:29.25 | Mercestes | dijungal, I would read the directions on that but it makes sense. |
| 19:29.38 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:29.42 | dijungal | k |
| 19:29.50 | soulfreshner | Mercestes, you write your own manager? |
| 19:30.08 | Mercestes | soulfreshner, I could but...no one's offered me enough $$$ to do it. |
| 19:30.34 | soulfreshner | Mercestes, hehe - but what did you mean by custom gui? |
| 19:30.42 | Mercestes | I was referring to portals to configure phones, actually, find me/follow me, phone forwarding, etc. etc. things of that nature. |
| 19:30.55 | Mercestes | managers..... |
| 19:30.58 | Mercestes | meh. Not my thing. |
| 19:31.04 | Mercestes | I just use fop |
| 19:31.27 | soulfreshner | Mercestes, how's customer feedback been on fop? |
| 19:31.41 | soulfreshner | I'm still debating wether I even want to install it... |
| 19:31.48 | Mercestes | people like it |
| 19:32.08 | Mercestes | I think it could be done alot better. |
| 19:32.39 | Mercestes | Hell, * could kick CCM ass if someone would program the right interfaces for it. |
| 19:32.49 | Mercestes | the things I could do with .net and some touch screens. woohoo |
| 19:33.11 | Mercestes | but, alas....money calls and I must stay the dream. |
| 19:33.57 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 19:34.51 | soulfreshner | stay the dream... sad |
| 19:34.52 | *** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.109) |
| 19:34.55 | vitaminmoo | Hello |
| 19:35.00 | soulfreshner | ...but true |
| 19:35.26 | hmmhesays | so should I be using users.conf now to create peers? |
| 19:36.05 | jarrod | tk shuttup |
| 19:36.53 | [TK]D-Fender | jarrod: You really should so something about that lag :) |
| 19:36.58 | vitaminmoo | I'm getting Spawn extension (blah,s,2) exited non-zero on 'Zap/8-1', where blah,s,2 is a very simple Dial command |
| 19:37.14 | jarrod | i was eating lunch :( |
| 19:37.17 | vitaminmoo | When this happens, it seems to give the caller a fast busy and drop them, but I can't reproduce it reliably |
| 19:37.18 | rpm | ICBC is the Plague. |
| 19:38.24 | vitaminmoo | If a Dial() cmd has a goto directly after it with the 'n' priority, if the dial fails for any reason, it should hit the goto(), shouldn't it? |
| 19:38.58 | soulfreshner | vitaminmoo, sounds about right |
| 19:39.18 | vitaminmoo | soulfreshner: Any known reason why it would error out and drop the call instead? |
| 19:39.34 | Mercestes | vitaminmoo, what version of asterisk are you running and do you have priority jumping enabled? |
| 19:39.34 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 19:39.58 | vitaminmoo | 1.4.12.1, and I do not have priority jumping enabled |
| 19:39.58 | Mercestes | with priorityjumping=yes, then your next priority would be n+101, not n+1. |
| 19:40.19 | Mercestes | vitaminmoo, I would pastebin a copy of your dialplan and the cli output of your error on verbosity 37. |
| 19:40.44 | Mercestes | core set verbose 37 |
| 19:40.45 | vitaminmoo | verbosity is inclusive so 999 will work, yes? |
| 19:41.01 | Mercestes | I don't want that much output, 37 will be fine. |
| 19:41.04 | hmmhesays | users.conf anyone is there any documentation on it? |
| 19:41.13 | [TK]D-Fender | Mercestes: because 38 would be SILLY :p |
| 19:41.23 | Mercestes | [TK]D-Fender, Precisely. |
| 19:41.25 | soulfreshner | Mercestes, why 37? I thought it only made a difference up to 10? |
| 19:41.27 | vitaminmoo | Mercestes: This has been very difficult to reproduce, and I've got saved output from 999 |
| 19:41.39 | Mercestes | vitaminmoo, ok ok, you can pastebin that then. |
| 19:41.41 | [TK]D-Fender | soulfreshner: Careful... I'm sensing some synapses firing ;P |
| 19:41.48 | Mercestes | soulfreshner, actually, it only makes a difference up to 3. |
| 19:41.56 | vitaminmoo | One moment |
| 19:42.10 | Mercestes | It's documented up to 10 but......people only used 1, 2, and 3. |
| 19:42.42 | soulfreshner | [TK]D-Fender, be gentle - I know not what I donot know yet. It's still early days with asteris for me |
| 19:42.43 | Mercestes | soulfreshner, same reason I tell people to use $callerid(numanumadance) |
| 19:43.09 | Mercestes | It works.....and only a few people know why. |
| 19:44.34 | hmmhesays | [TK]D-Fender: can you make the poly's monitor more than one mailbox? |
| 19:45.41 | [TK]D-Fender | hmmhesays: yup |
| 19:45.51 | soulfreshner | Mercestes, that's just cruel (: |
| 19:46.04 | Mercestes | soulfreshner, I'm well known for that. |
| 19:46.27 | hmmhesays | how do you do that? |
| 19:48.21 | Mercestes | I'm pretty sure it's at the bottom of phone.cfg |
| 19:48.35 | Mercestes | It's got like six entries for mailbox. Just.....make them different. |
| 19:49.46 | Mercestes | I think you have to have your lines 2-6 configured for your mailboxes 2-6 to work tho. |
| 19:50.20 | vitaminmoo | Mercestes: http://pastebin.com/d6611d537 |
| 19:51.55 | Mercestes | couldn't you grep -v ; that before you posted it? |
| 19:52.39 | vitaminmoo | Didn't to avoid just mashing it all together, I will next time. |
| 19:52.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:53.12 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 19:53.15 | ZaVoid | morning |
| 19:53.50 | Mercestes | Morning ZaVoid. |
| 19:54.02 | ZaVoid | hey Mercestes wassap |
| 19:54.04 | Mercestes | vitaminmoo, I think sip/23 is broken. Zap/8-1 appears to be working just fine. |
| 19:54.48 | vitaminmoo | Hmm, that's just an unused Polycom, should just be ringing for 12 seconds |
| 19:54.50 | Mercestes | vitaminmoo, You might want to sip debug it to see if it's a transcoding issue. |
| 19:55.21 | ZaVoid | i think rtptimeout is broken |
| 19:55.31 | Mercestes | ZaVoid: only on cisco phones. |
| 19:55.40 | ZaVoid | really? |
| 19:55.43 | Mercestes | Yea. |
| 19:55.43 | jer | anybody know of some nice * accounting packages? |
| 19:55.45 | ZaVoid | i'm testing with a pap2 |
| 19:55.50 | Mercestes | you mute a cisco...and the cisco stops putting out rtp. |
| 19:55.53 | ZaVoid | i can't get the call to disconnect |
| 19:56.01 | ZaVoid | i pull the ethernet cable from the pap2 |
| 19:56.09 | ZaVoid | and the pap2 account in iasterisk is a peer type |
| 19:56.38 | Mercestes | vitaminmoo, what are your disallow/allows on zap and sip? |
| 19:56.41 | vitaminmoo | Don't ciscos only stop sending rtp if CNG is off, or is that only a PBX option with skinny? |
| 19:56.51 | *** part/#asterisk synthetiq (i=walletje@53516DE0.cable.casema.nl) |
| 19:57.00 | Mercestes | ... |
| 19:57.13 | ZaVoid | this looks different |
| 19:57.13 | Mercestes | I dunno. I hit mute and the call hangs up after hte rtptimeout on ciscos but not polycoms |
| 19:57.21 | ZaVoid | rtptimeout = seconds : Terminate call if x seconds of no RTP activity when we're not on hold. Valid only in [general] section and type=peer. |
| 19:57.28 | ZaVoid | iot never used to say in [general] only |
| 19:58.00 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 19:58.02 | Mercestes | I think you have to define rtptimeout= under [general |
| 19:59.09 | ZaVoid | and it would apply to others as well? |
| 19:59.22 | Mercestes | ZaVoid: in theory. |
| 19:59.23 | ZaVoid | i got it set to 30 there |
| 19:59.28 | ZaVoid | but its not killing calls |
| 19:59.33 | ZaVoid | i use realtime too |
| 19:59.40 | Mercestes | hrm. |
| 19:59.44 | ZaVoid | but my entrys in the db are set to peer |
| 20:00.00 | nestAr | hahah.. motherfuckin colin |
| 20:00.01 | Mercestes | sounds like rtptimeout is broken then. |
| 20:00.12 | Mercestes | nestAr: wishing you'd gone straight? |
| 20:00.48 | nestAr | once you go black, you never go back. |
| 20:01.12 | Mercestes | yea, but i'm not the one bitching about my colon in #asterisk either. |
| 20:01.26 | nestAr | except i didn't say colon. |
| 20:01.37 | Mercestes | technically, you didn't say anything.... |
| 20:01.41 | Mercestes | unless you self-narrate. |
| 20:01.59 | Mercestes | or you have one of those voice-text deals because your a quardiparalegic. |
| 20:02.02 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:02.12 | Mercestes | ...quadriparalegic.....even. |
| 20:02.15 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 20:02.16 | [hC] | So... 1.2 isnt available for download off the website anymore? :) |
| 20:02.28 | nestAr | freud was right? |
| 20:02.49 | Mercestes | that you have penis envy? yea, I think so. |
| 20:03.13 | Mercestes | ... |
| 20:03.22 | Mercestes | sorry, that was just unnecessary. >.> |
| 20:03.26 | vitaminmoo | Eep, I've got to work on other things, thanks for the help Mercestes |
| 20:03.31 | Mercestes | So what about Colin? |
| 20:03.41 | Mercestes | vitaminmoo, laters. :D |
| 20:04.17 | *** join/#asterisk macog (n=dklima@200.195.161.164) |
| 20:04.44 | *** join/#asterisk Bl0w_M0nk (n=gy@66-168-56-207.dhcp.mdsn.wi.charter.com) |
| 20:06.02 | [hC] | Digium guys: you know the ftp site is handing out connection refused? |
| 20:06.25 | *** join/#asterisk Zenith77 (n=moose@c-76-110-200-130.hsd1.fl.comcast.net) |
| 20:06.29 | Zenith77 | Hi. |
| 20:07.32 | Zenith77 | Is any available for help, I don't understand what the following means: |
| 20:07.49 | _x86_ | [hC]: i think that's desired, as they are moving to http://downloads.digium.com/ |
| 20:07.50 | Mercestes | Zenith77, it's a colon. |
| 20:07.54 | file | [hC]: the FTP site was taken down a few months ago |
| 20:07.59 | Zenith77 | ha Mercestes =p |
| 20:08.02 | Zenith77 | Here's the url |
| 20:08.03 | Zenith77 | http://zenith.ampaste.net/110254 |
| 20:08.08 | Zenith77 | sorry had to grab it real quick ^^ |
| 20:08.26 | ZaVoid | anyone else use rtptimeout successfuly? |
| 20:08.43 | Mercestes | Yea, I don't know what it means either. |
| 20:08.49 | Mercestes | is there something broken about it? |
| 20:09.02 | Mercestes | ZaVoid: I only got it to drop my calls when I muted a cisco. That was all the luck I had with it. |
| 20:09.34 | [hC] | the problem with the http://downloads.digium.com site is that the url's are not direct, they are all reference links which (presumably) track downloads. This makes it difficult to download to a server using something like wget |
| 20:10.00 | ZaVoid | yeah i'm trying ot make it workoutside mute though |
| 20:10.14 | file | [hC]: you should be able to wget http://downloads.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz |
| 20:10.27 | _x86_ | [hC]: quote it... the only plausible issue with using wget is the ? in the URL getting expanded by the shll |
| 20:10.31 | _x86_ | shell* |
| 20:10.49 | [hC] | _x86 not true, wget downloads this: |
| 20:10.50 | [hC] | 13:10:37 (50.93 MB/s) - `elqRedir.htm?ref=http:%2F%2Fdownloads.digium.com%2Fpub%2Fasterisk%2Fold-releases%2Fasterisk-1.2.24.tar.gz' saved [2403/2403] |
| 20:10.51 | _x86_ | [hC]: throw quotes on the whole URL, and you wont have any problems |
| 20:10.54 | [hC] | and it is quoted. |
| 20:11.02 | fujin_ | man wget |
| 20:11.05 | Zenith77 | Don't mean to be a push over, but is there anyone here that could just point me in the right direction for my error? |
| 20:11.06 | fujin_ | you can tell it what to save the file to |
| 20:11.21 | [hC] | fujin_: thats not hte real file, its html containing a redirect. |
| 20:11.29 | fujin_ | then use the ftp directory like everyone else |
| 20:11.40 | [hC] | ........................ |
| 20:12.21 | [hC] | fujin_: so that would be the logical step... especially since its listed on asterisk.org that ftp is available. but all the ftp sites have been shut down. :) |
| 20:12.33 | [hC] | which is how i started this conversation. |
| 20:12.33 | Dan0maN_Work | i always follow that link in my web browser, and right click the "if you are having difficulty downloading this file, please click here" link |
| 20:12.38 | Mercestes | Zenith77, what's broken? |
| 20:12.53 | Zenith77 | I...I don't know lol |
| 20:13.02 | Zenith77 | I'm using X-Lite as a softphone |
| 20:13.09 | Zenith77 | I have two computers, both connected through an ethernet hub |
| 20:13.10 | Mercestes | Zenith77, is it doing something you don't wish it to do or not doing something you want it to do? |
| 20:13.22 | Zenith77 | Mercestes, did you read the link up there? |
| 20:13.24 | Zenith77 | http://zenith.ampaste.net/110254 |
| 20:13.30 | [hC] | I'll just hand-edit the url to take out the referring page. just a pain in the ass. |
| 20:13.42 | Mercestes | Zenith77, yes........ |
| 20:13.53 | Mercestes | what's broken? |
| 20:14.03 | Zenith77 | Err, Asterisk Win32? |
| 20:14.23 | Mercestes | Zenith77, how does behavior deviate from expectations? |
| 20:14.35 | Zenith77 | I place a call on ext 77 |
| 20:14.37 | Zenith77 | This works |
| 20:14.41 | Zenith77 | it calls, then it gives that in the console |
| 20:14.43 | Zenith77 | and hangs up |
| 20:14.49 | Zenith77 | And X-Lite gives me "call failed" |
| 20:15.02 | Mercestes | Do you have sip call limits set? |
| 20:15.17 | Zenith77 | err, don't know what that is... |
| 20:15.34 | Zenith77 | I'm only about half way through the manual atm, so... |
| 20:15.48 | *** join/#asterisk Dovid (n=Dovid@bzq-79-182-99-49.red.bezeqint.net) |
| 20:16.18 | Mercestes | Zenith77, well, first, I would suggest running asterisk in linux. |
| 20:16.26 | Zenith77 | not an option |
| 20:16.33 | Dovid | ${CALLERID(ani)} should show me the ANI on the line ? I am running it how ever I get nothing andmy carrier says they are sending it over SIP. does asterisk s upport this ? |
| 20:16.39 | Zenith77 | Our boxes will be running WinXP SP2 |
| 20:16.47 | fujin_ | serious? |
| 20:16.49 | Mercestes | Zenith77, Then your screwed. |
| 20:16.52 | fujin_ | asterisk on windows, well, that's a stupid idea |
| 20:16.56 | Zenith77 | ... |
| 20:16.57 | Zenith77 | -.- |
| 20:17.02 | Zenith77 | Hey. |
| 20:17.05 | fujin_ | isn't that entirely unsupported? |
| 20:17.05 | Zenith77 | Not my choice :) |
| 20:17.08 | Dovid | Zenith77: LEARN LINUX !!!!! |
| 20:17.09 | Zenith77 | I don't know. |
| 20:17.11 | fujin_ | your choicemaker is stupid |
| 20:17.14 | Zenith77 | Dovid, I know it. |
| 20:17.15 | fujin_ | fire them |
| 20:17.32 | Zenith77 | lol we didn't have an option I believe |
| 20:17.51 | [TK]D-Fender | ZenSure you did... Sedition :) |
| 20:17.57 | Dovid | Zenith77: There is a windows option. It is not the most reliable option |
| 20:18.12 | fujin_ | I didn't even know that, it's so far fetched and silly |
| 20:18.24 | orakle | whoa |
| 20:18.27 | orakle | there's asterisk for windows |
| 20:18.28 | orakle | hahahahah |
| 20:18.36 | [TK]D-Fender | No, not really |
| 20:19.08 | [TK]D-Fender | there is on that runs under cygwin which emulates a *nix environment under windows, but I wouldn't count that. |
| 20:19.09 | *** join/#asterisk mvanbaak (i=michiel@vanbaak.xs4all.nl) |
| 20:19.25 | [TK]D-Fender | Pathetic illusions for pathetic admins. |
| 20:20.02 | Mercestes | Windows Vista unixtools > cygwin |
| 20:20.33 | Mercestes | Zenith77, Just setup one silly linux box. how hard is that? |
| 20:20.52 | Mercestes | That's like demanding a manual transmission and proclaiming that using hte clutch is not an option. |
| 20:21.06 | Dovid | TK: Is there ANI support over SIP in asterisk ? |
| 20:21.22 | nestAr | which is a viable option if you're driving a Ferrari |
| 20:21.49 | Mercestes | nestAr: You've obviously never driven a ferrari. |
| 20:22.22 | nestAr | i haven't, but i have driven a maserati |
| 20:22.29 | nestAr | with a ferrari engine and transmission |
| 20:22.51 | [TK]D-Fender | Dovid: Dunno |
| 20:23.05 | Dovid | TK: ARgh !!!!! thanks |
| 20:24.33 | orakle | anyone good with cisco phones? i have a CP-7912G |
| 20:24.44 | orakle | you plug it in, it lights up for a second and then the lights turn off and it just sits there |
| 20:24.48 | orakle | nothing on the display |
| 20:25.09 | orakle | if you press the "world" button though, the green LED turns on, and if you press it again it turns off |
| 20:25.40 | orakle | i'm thinking it's like a corrupt flash but i don't know how to fix it |
| 20:25.54 | [TK]D-Fender | ok, BBIAB |
| 20:26.35 | Zenith77 | sigh Mercestes, nm. |
| 20:26.56 | Mercestes | Zenith77, surely you have a spare machine somewhere... |
| 20:28.41 | nestAr | heh |
| 20:31.53 | Mercestes | why do ppl make it difficult? |
| 20:32.37 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 20:33.28 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 20:34.04 | Alan_Hicks | Howdy folks. I may have a project in installing Asterisk for a small business with four incoming PSTN lines, six or seven stations, and a remote telecommuting user. |
| 20:34.43 | Mercestes | Alan_Hicks, Congratz. |
| 20:34.45 | Alan_Hicks | I'm trying to keep costs to a minimum. Can anyone recommend inexpensive phones (preferably with a good warranty) that don't have to be packed with all the latest and greatest features? |
| 20:35.00 | Mercestes | Alan_Hicks, Polycom 501s. |
| 20:35.04 | Alan_Hicks | Thanks. |
| 20:35.07 | Mercestes | ~phones |
| 20:35.07 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 20:35.42 | Alan_Hicks | Oh, the bot talks. Excellent. Thanks. |
| 20:35.49 | Mercestes | No problem. good luck. |
| 20:36.05 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 20:36.10 | Mercestes | brb |
| 20:36.11 | SparFux | On an incoming sipgate call I try to immediately dial to a capi line and get "Everyone is busy/congested at this time (1:0/0/1)". But no line is busy! What could cause this problem? |
| 20:36.27 | Zenith77 | TJNII, you there? |
| 20:37.28 | nestAr | lol. Grandstream.. BudgetTel, or as I like to call them GhettoTels |
| 20:37.37 | Dovid | ~gs |
| 20:37.38 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:37.42 | nestAr | i gotta couple of them, they are sweet. |
| 20:37.58 | Dovid | nestAr: I would have to disagree |
| 20:39.12 | nestAr | and I quote Tommy Boy "I'm picking up on your sarcasm", "Well, I should hope so, because I'm laying it on pretty thick" |
| 20:40.01 | Dovid | hehe |
| 20:40.21 | Alan_Hicks | Perhaps a dumb question, but I'm reading up on those Polycom 501 phones, and it says they support 3 lines. Does this mean 3 PSTN lines? |
| 20:40.28 | nestAr | no |
| 20:40.38 | Alan_Hicks | What does it mean then? |
| 20:40.43 | Strom_M | Alan_Hicks: three line appearances |
| 20:40.46 | *** join/#asterisk eldon (i=eldon@nat/digium/x-71acd1282bfda836) |
| 20:40.52 | Alan_Hicks | OH! |
| 20:40.56 | Alan_Hicks | For the LCD screen. |
| 20:40.59 | Strom_M | yes |
| 20:41.04 | Dovid | Alan_Hicks: as strom said or 3 sip accounts |
| 20:41.05 | Alan_Hicks | Thank you. |
| 20:41.22 | Strom_M | you have to realize that the number of line appearances behind your pbx bears little relation to the number of circuits you have in front of your PBX |
| 20:41.41 | Alan_Hicks | Is there any particular merchant that the channel recommends for voip hardware? |
| 20:41.59 | Strom_M | Alan_Hicks: are you in the US? |
| 20:42.13 | Alan_Hicks | Strom_M: Yes I understand, but I was thinking that it might have three buttons to specify an outbound number when making a call outside the service set. |
| 20:42.16 | Alan_Hicks | Yes. |
| 20:42.21 | Dovid | Alan_Hikcs: In the US I like voipsupply.com and telephonydepot.com |
| 20:42.22 | Strom_M | telephonydepot.com |
| 20:42.31 | Strom_M | i'm not fond of voipsupply.com |
| 20:42.42 | Dovid | telephonydepot.com seems to have lower prices |
| 20:42.46 | Alan_Hicks | Thanks. I'm window shopping voipsupply.com right now. I'll check the other. |
| 20:42.47 | Dovid | Strom_C: Why now? |
| 20:42.48 | Dovid | not* |
| 20:43.05 | Strom_M | Alan_Hicks: no, the logic for determining a circuit for outbound calls happens in the PBX, not in the phone |
| 20:43.13 | Strom_M | that's why it's a PBX and not a key system |
| 20:43.25 | Alan_Hicks | Strom_M: That's what I thought, which is why I was confused. :^) |
| 20:43.49 | Strom_M | Dovid: obviously used hardware when I ordered new, missing pieces of orders, weird policies for credit cards |
| 20:44.16 | nestAr | voipsupply did alright by me, in that they didn't go out of business mid-order on me, like atacomm |
| 20:44.28 | Strom_M | hahah |
| 20:44.36 | Alan_Hicks | haha |
| 20:44.42 | eldon | I guess that's always a plus |
| 20:45.07 | nestAr | i only ordered a single span t1 card from voipsupply... i will keep your comments in mind next time i order a bunch of stuff. |
| 20:45.10 | eldon | question: did they go out of business %Uafter%U they took your money? |
| 20:45.15 | nestAr | yes |
| 20:45.22 | eldon | that sucks |
| 20:45.25 | nestAr | but i got it back from the CC company. |
| 20:45.35 | Alan_Hicks | nestAr: That's fortunate. |
| 20:45.47 | nestAr | i had a really good track record with atacomm, i had done over 15k in business with them.. |
| 20:45.49 | Dovid | Strom_C: good to know. i bounce between the two. recently been using telephonydepot.com more |
| 20:46.04 | Dovid | nestAr: Too bad that they went down |
| 20:46.09 | nestAr | they said that card was on backorder, i wasn't in a rush.. i called a week later to check on it, and they were out of business. |
| 20:46.39 | nestAr | with no one there to contact, i advised our accountant to file a chargeback. |
| 20:46.45 | eldon | businesses shut down all the time at the drop of a hat |
| 20:46.47 | nestAr | we got lucky and got the money back. |
| 20:46.59 | nestAr | yes, i know. i've been on the employee side of that |
| 20:47.18 | Alan_Hicks | Usually they'll get bought out by another company if they have any assets worth having though. |
| 20:47.19 | nestAr | with customers who didn't get their money back. actually, they owe me money still too. |
| 20:47.39 | Alan_Hicks | Of course, simple merchants like say Amazon.com rarely have anything worth having beyond their current inventory. |
| 20:47.42 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 20:47.47 | nestAr | yea |
| 20:48.01 | nestAr | a lot of these companies don't even have any inventory |
| 20:48.04 | nestAr | it's all drop ships |
| 20:48.18 | nestAr | i do a lot of drop shipping from my basement. |
| 20:48.20 | Dovid | yea. I needed a phone once |
| 20:48.31 | Dovid | 3 companies had the same excuse |
| 20:48.36 | Dovid | it was on back order ;) |
| 20:48.40 | nestAr | but i try to only deal in shit that's actually in stock for shipping. |
| 20:48.41 | eldon | with Amazon though they have brand recognition... |
| 20:49.00 | Alan_Hicks | And the idiotic 1-Click patent. |
| 20:49.02 | nestAr | the name is worth something there, a lot, i'd say.. |
| 20:49.05 | Dovid | i had an issue with voip supply and they said that it was shipped "from their other whare house" |
| 20:49.26 | eldon | um... whore house? |
| 20:49.26 | *** join/#asterisk VJFROMGT (n=vjfromgt@68.161.227.229) |
| 20:49.33 | nestAr | just like the company i work for now, they have offers from companies that just want their domain name. |
| 20:49.33 | Alan_Hicks | hahaha |
| 20:49.44 | VJFROMGT | does myone know if i can add wildcards in host= ? |
| 20:49.54 | *** join/#asterisk __deg__ (n=deg@200.195.161.164) |
| 20:52.16 | *** join/#asterisk DMeloUK (n=info@64.129.93.147) |
| 20:52.22 | lirakis | later all |
| 20:52.28 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:53.05 | Alan_Hicks | Are all the instructions here still valid for the Polycom 501 and Asterisk? The page appears to be over a year old. http://tinyurl.com/br75h |
| 20:53.15 | DMeloUK | can I get support on the asterisk appliance here |
| 20:54.02 | orakle | well, if it runs asterisk someone here probably knows what's going on with it |
| 20:54.03 | orakle | :) |
| 20:54.08 | nestAr | lol http://piratewars4.piratewarsonline.com/lolcats_prod//images/cat366.jpg |
| 20:54.10 | orakle | what's the trouble DMeloUK? |
| 20:55.15 | __deg__ | Is this possible to tell a queue to ring the agents through a macro instead of dial them directly(like SIP/101)? |
| 20:55.48 | __deg__ | What i need is to do some checks before send the call to an agent(SIP agent) |
| 20:55.54 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:57.37 | *** join/#asterisk mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 20:58.42 | hmmhesays | ok what the hell |
| 20:58.49 | hmmhesays | my a200 sangoma I only found 2 channels |
| 20:58.59 | hmmhesays | I have 2 fxo modules, that should be 4 channels |
| 21:00.27 | russellb | you're using the wrong card, man :-p |
| 21:00.41 | DMeloUK | I cannot seem to get eyebeam to connect to the asterisk appliance |
| 21:00.52 | DMeloUK | I have an s800i |
| 21:00.52 | hmmhesays | say what? |
| 21:01.07 | twisted | hey russellb, you back in town? |
| 21:01.15 | russellb | twisted: just got in, yes |
| 21:01.15 | DMeloUK | it says 503 service unavailable |
| 21:01.28 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 21:01.41 | twisted | russellb, cool |
| 21:01.43 | hmmhesays | the a200 card should hand for 4 fxo channels |
| 21:02.28 | hmmhesays | *handle even |
| 21:02.41 | joat | anyone know of a better streaming interface to asterisk than ices? getting the thing configured properly is giving me fits |
| 21:03.51 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 21:04.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:05.45 | SparFux | For some reason I get busy tone when Dial()ing from my sipgate context to other devices. |
| 21:08.28 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 21:08.34 | pat2man | __deg__: when an agent loggs in (if you are using agentcallbacklogin) they log in with an extension and a context, just have that context dial however you want, I wanted mine to dial longer than our usual length so I could specify the timeout in queues.conf so: |
| 21:08.36 | pat2man | [from-queues] |
| 21:08.36 | pat2man | exten => _X.,1,Dial(sip/${EXTEN},60,twW) ; Dial for as long as we need |
| 21:09.16 | pat2man | and agents log in with AgentCallbackLogin(${CALLERID(num)}||${CALLERID(num)}@from-queues) |
| 21:09.23 | fujin_ | grargh |
| 21:09.31 | fujin_ | don'tuseagentcallbackloginit'sbrokenanddeprecated |
| 21:09.37 | pat2man | yeah yeah yeah |
| 21:10.00 | Alowishus | Is running Asterisk on VMware ESX server advisable? |
| 21:10.16 | mercestes | Alowishus, no. |
| 21:10.29 | mercestes | Alowishus, Vmware ruins the timing sources required for the queues and zap. |
| 21:10.33 | peanut- | why would you ever do that? |
| 21:10.36 | Zenith77 | mercestes, just out of curiosity, why would I need the linux version? |
| 21:10.46 | pat2man | fujin_: I assume __deg__ is using callbacklogin, otherwise it would not be an issue |
| 21:10.48 | mercestes | Uh, high availability, fail over, intelligence. |
| 21:11.01 | mercestes | Zenith77, because it works. |
| 21:11.05 | peanut- | mmhmm.. |
| 21:11.06 | Alowishus | Mercestes: afraid of that... was thinking it could be done if using external media gateways nad then I get the benefit of all the redundancy already built into the cluster |
| 21:11.30 | Alowishus | Mercestes: but I'll still have timing issues? I would be using ztdummy for conferencing |
| 21:11.43 | mercestes | Alowishus, Yea, it blocks access to the rtc. |
| 21:11.48 | Alowishus | hrm |
| 21:11.56 | hmmhesays | ok this damn thing is only finding ports 3 and 4 |
| 21:11.57 | mercestes | Alowishus, Callweaver *claims* to work in Vmware but....I've never seen it work. |
| 21:12.25 | mercestes | VmWare and asterisk would be sweet. |
| 21:12.29 | fujin_ | heh |
| 21:12.51 | Alowishus | ok then so I'll have to build redundancy externally... which is doable... given the need for a pair of T1 interfaces, are there external gateway options for that or should I be using internal cards? |
| 21:12.54 | fujin_ | asterisk tends to be more reliant upon direct access to hardware from what I've seen |
| 21:13.01 | mercestes | Alowishus, use SER and asterisk. |
| 21:13.06 | hmmhesays | anyone running a sangoma a200? |
| 21:13.10 | fujin_ | although I've heard cases of it running most happily within Xen |
| 21:13.13 | fujin_ | although xen isn't that super anyway |
| 21:13.15 | mercestes | Alowishus, and there are t1 fail over devices. |
| 21:13.15 | [TK]D-Fender | Zenith77, It isn't so much a "Linux version". You missed the point that Asterisk was coded for *nix. There is no WINDOWS VERSION. Nothing compiles for Windows. What you have is a FAKE Unix envirnment running under Win32. Configuing * will be EXACTLY the same, only more limited and don't bet on newer versions working to well, and FORWGET hardware support |
| 21:13.24 | Alowishus | Mercestes: any pointers? |
| 21:13.32 | [TK]D-Fender | hmmhesays, "wanrouter hwprobe" |
| 21:13.43 | Zenith77 | [TK]D-Fender, hmmm |
| 21:13.49 | mercestes | Alowishus, google asterisk high availability |
| 21:13.54 | fujin_ | I'm still loling @ asterisk on windows |
| 21:13.56 | mercestes | I'm not a SER administrator. |
| 21:13.57 | Zenith77 | Was Asterisk ported by a 3rd part or something? |
| 21:13.58 | fujin_ | seriously, why would someone port it |
| 21:14.03 | Alowishus | fujin_: well with paravirtualization under Xen, the guest would still have access to rtc |
| 21:14.08 | Alowishus | fujin_: so that would explain |
| 21:14.30 | [TK]D-Fender | Zenith77, There is no Windows Click&Go BS version of *. Stop wasting your time. Denial won't get you far in life. |
| 21:14.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:14.34 | Zenith77 | fujin_, because from what I see, it opens up a lot more doors ofr asterisk |
| 21:14.47 | [TK]D-Fender | Zenith77, NOT PORTED. What aren't you understanding? |
| 21:14.58 | Zenith77 | -.- |
| 21:15.10 | fujin_ | you mean, it closing down half of the open doors? |
| 21:15.19 | fujin_ | what are these doors you're referring to |
| 21:15.22 | [TK]D-Fender | Zenith77, it runs under Cygwin with mean to * its still on *nix! It is living one big LIE. |
| 21:15.23 | fujin_ | and uh, don't you mean windows? :P |
| 21:15.40 | Zenith77 | [TK]D-Fender, ah. |
| 21:15.47 | Zenith77 | lol fujin_ |
| 21:15.49 | [TK]D-Fender | jksadhjlsadhjkljdasdhlkajsdhjklasgsdfklafa |
| 21:15.57 | fujin_ | yuck, asterisk incide cygwin? |
| 21:15.59 | fujin_ | thats' even worse |
| 21:16.03 | fujin_ | I thought it might have been a native port |
| 21:16.08 | [TK]D-Fender | NO |
| 21:16.22 | [TK]D-Fender | Holy crap what's it take to drill this into people's heads?! |
| 21:16.48 | Zenith77 | Sorry, I'm an asterisk noob ^^. |
| 21:16.54 | fujin_ | s/asterisk// |
| 21:17.09 | [TK]D-Fender | Zenith77, No, if you couldn't follow what I've said all this time, you're just blind :) |
| 21:17.11 | Zenith77 | (So3kris): |
| 21:17.14 | Zenith77 | err |
| 21:17.16 | Zenith77 | so, |
| 21:17.17 | Zenith77 | http://www.asteriskwin32.com/ |
| 21:17.35 | fujin_ | wow, 99999 downloads |
| 21:17.39 | Zenith77 | lol |
| 21:18.13 | Zenith77 | so, it's not even a port eh? |
| 21:18.21 | fujin_ | heh |
| 21:18.25 | fujin_ | they're not even compliant to the GPL |
| 21:18.31 | fujin_ | (don't provide source) |
| 21:18.32 | Zenith77 | Just runs under an emulator? |
| 21:18.37 | fujin_ | piss that, you wouldn't catch me dead running it |
| 21:18.47 | mercestes | fujin_, lmao |
| 21:18.58 | Zenith77 | fujin_, source is available for download O.o |
| 21:19.08 | [TK]D-Fender | Cygwin1.dll application conflict <---------- |
| 21:19.17 | hmmhesays | this is giving me hell |
| 21:19.25 | Zenith77 | well |
| 21:19.32 | Zenith77 | Can you at least tell me if my problem is related to Windows. |
| 21:19.42 | [TK]D-Fender | By default AsteriskWin32 is installed in a directory named cygroot on your system. It will create four subdirs asterisk, bin, lib, tmp. AsteriskWin32 executables are located in bin directory. |
| 21:19.42 | [TK]D-Fender | If you have already cygwin installed on your system you must install AsteriskWin32 inside cygwin root directory, so change the default cygroot: install directory to your cygwin directory. |
| 21:19.44 | mercestes | No, your problem is your retarded. |
| 21:19.47 | Zenith77 | Because if it isn't, and I install Linux, I'm going to be pissed. |
| 21:19.50 | [TK]D-Fender | so DUH, Cygwin. |
| 21:20.03 | [TK]D-Fender | Zenith77, www.drphil.com |
| 21:20.08 | fujin_ | asterisk inside cygwin is stupid and not recommended |
| 21:20.12 | fujin_ | you're not goign to find any help here |
| 21:20.18 | Zenith77 | I did yesterday. |
| 21:20.20 | mercestes | Not tryign to be mean, but, seriously....it's not going to work |
| 21:20.28 | fujin_ | They were probably pretending to help. |
| 21:20.33 | [TK]D-Fender | Zenith77, this is the point where I tell you to "cry me a river" so we can hold your head under :p |
| 21:20.34 | mercestes | It's broken, unsupported, and hopeless. |
| 21:20.36 | *** join/#asterisk smultron (n=lukas@cpe-67-9-146-21.austin.res.rr.com) |
| 21:20.38 | Zenith77 | yay, rivers |
| 21:20.40 | fujin_ | diaf |
| 21:20.50 | smultron | does the asterisk appliance support ISDN PRI lines? |
| 21:20.54 | mercestes | Just grab one of yoru cheap throw away PCs, install linux, and be done with it. |
| 21:21.02 | Zenith77 | I would just like closure, my error is caused by AsteriskWin32 correct? |
| 21:21.08 | fujin_ | yes |
| 21:21.09 | fujin_ | eof |
| 21:21.11 | Zenith77 | and it SHOULD work running on Linux. |
| 21:21.17 | fujin_ | s/SHOULD/does/ |
| 21:21.20 | Zenith77 | lol |
| 21:21.38 | fujin_ | it is DEVELOPED, DESIGNED FOR and RUNS ON Linux primarily |
| 21:21.43 | Zenith77 | kk |
| 21:21.47 | Zenith77 | Ubuntu okay? |
| 21:21.51 | Zenith77 | (don't laugh) |
| 21:21.51 | Qwell | sure |
| 21:21.53 | fujin_ | That's what I run. |
| 21:21.56 | Zenith77 | kk |
| 21:22.04 | fujin_ | 50~ phone callcentre, 2k~ calls/day |
| 21:22.05 | Qwell | or debian, or centos, or gentoo...it doesn't really matter |
| 21:22.18 | Zenith77 | time to email meh employer :D |
| 21:22.20 | [TK]D-Fender | Most people on FreeBSD are fanatical enough to beat it into relatively fully functionality, OpenBSD and the rest not so lucky on the zaptel side |
| 21:22.45 | Zenith77 | So which distro would you guys recomend? |
| 21:22.46 | mercestes | But OpenBSD is so perfect otherwise. >.> |
| 21:22.52 | [TK]D-Fender | Zenith77, Ubuntu won't make your * any better..... |
| 21:23.07 | Zenith77 | So, OpenBSD? |
| 21:23.07 | mercestes | Zenith77, Gentoo is pure asterisk numminess. |
| 21:23.18 | mercestes | Do you *read* what we type? |
| 21:23.26 | lemanal | Is there any video conferencing support for asterisk? Something like a meetme conference with video? |
| 21:23.36 | [TK]D-Fender | Zenith77, Best choices in terms of getting support : CentOS, Debian, Gentoo (in roughly that order). |
| 21:23.37 | mercestes | [TK]D-Fender: OpenBSD and the rest not so lucky on the zaptel side |
| 21:23.47 | Zenith77 | okay... |
| 21:23.53 | Zenith77 | [TK]D-Fender, did you look at my logs? |
| 21:23.58 | [TK]D-Fender | mercestes, BSD's that is |
| 21:24.05 | mercestes | I know. |
| 21:24.05 | [TK]D-Fender | Zenith77, No, I didn't |
| 21:24.09 | Zenith77 | Are you sure it's caused by running under windows |
| 21:24.12 | mercestes | I honestly wouldn't vote for FreeBSD either. |
| 21:24.22 | Zenith77 | http://zenith.ampaste.net/110254 |
| 21:24.32 | Zenith77 | ^----- [TK]D-Fender |
| 21:24.39 | mercestes | What part of unsupported escaped you, Zenith? |
| 21:24.40 | lemanal | I've seen one solution but it looks windows only. and I hear windows is not so good for a asterisk server. |
| 21:24.45 | [TK]D-Fender | Zenith77, that doesnt' actually say anything meaningfull all by itself. |
| 21:24.46 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 21:24.51 | mercestes | s/unsupported/we don't care if it works or not it's not our problem/ |
| 21:25.10 | mercestes | bwahahaha |
| 21:25.19 | Zenith77 | So, somehow by magically swithcing to Linux I get help? |
| 21:25.29 | mercestes | Not magically, but yes. |
| 21:25.54 | [TK]D-Fender | Zenith77, That is running an OLD version whose LATEST incarnation isn't even supported. That is 1.2 series. |
| 21:25.55 | Zenith77 | I just wanted people to confirm before I switch to Linux... |
| 21:26.14 | Zenith77 | Because we also have another program that needs to run on the boxes |
| 21:26.17 | [TK]D-Fender | Zenith77, And being under Win32... LOL <- Trust me, poeple couldn't care less. |
| 21:26.39 | Zenith77 | It's being remade atm so it will be cross-platform compatiable. |
| 21:26.40 | [TK]D-Fender | Zenith77, Good then keep that box for those purposed and buy ANOTHER. |
| 21:26.47 | *** part/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 21:26.55 | Zenith77 | bah |
| 21:26.56 | Zenith77 | poo on you! |
| 21:26.58 | Zenith77 | :) |
| 21:26.59 | mercestes | don't even buy another.....just use an old 486. |
| 21:27.11 | [TK]D-Fender | Zenith77, Get over this childhood trauma of yours and splurge 100$ on a friggen PC for * and stop whining. |
| 21:27.16 | mercestes | hell, dig out an old Wrt54gl and install openwrt on that and install asterisk and use that. |
| 21:27.27 | Zenith77 | Umm, I'm not whining. |
| 21:27.35 | Zenith77 | It it were my actually computer, I would go ahead and do it. |
| 21:27.50 | Zenith77 | I would just like to figure out the error, I don't care if it's fixed, I just want to know at least what's causing it. |
| 21:27.58 | Zenith77 | In fact, that's all I really wanted to know in the first place :) |
| 21:28.07 | [TK]D-Fender | Zenith77, Sure you are ; "Oh PLEASE tell me what I want to hear, I'm afraid of Linux and I can't live outside my Windows box. If you have to, just LIE and tell me I can!" |
| 21:28.18 | Zenith77 | The box |
| 21:28.19 | Zenith77 | is |
| 21:28.20 | Zenith77 | not |
| 21:28.21 | Zenith77 | mine |
| 21:28.24 | mercestes | The fact that your running out of date, unsupported, unwanted, and abandoned software in windows that was originally written for linux wrapped up in the world's worst linux emulator. |
| 21:28.38 | [TK]D-Fender | mercestes, With a cherry-on-top! |
| 21:28.44 | Zenith77 | I can take that as a satisfactory answer :) |
| 21:29.29 | fujin_ | ./part |
| 21:29.38 | Zenith77 | >.< |
| 21:31.06 | [hC] | Who's the chan_zap master? |
| 21:32.29 | *** join/#asterisk _matt (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 21:32.34 | hmmhesays | first sangoma card ever and the damn thing is bad |
| 21:32.48 | [hC] | there seems to be an issue with zap not waiting long enough after hanging up before considering a line usable again, and it causes problems where lines will be picked back up before they're 'ready' from the telco, resulting in dead air, etc.. |
| 21:33.02 | [hC] | Is there any way to configure how long that process is? |
| 21:33.33 | *** join/#asterisk syneus (n=syneus@host23-25-dynamic.180-80-r.retail.telecomitalia.it) |
| 21:33.52 | mercestes | [hC], how many lines are available? |
| 21:34.15 | mercestes | hmmhesays, Might just need firmware updates, honestly. Sangomas have been flakey for me. |
| 21:34.20 | [hC] | mercestes: 4. I already accept incoming on 1-4 and dial out from 4-1, but with so few lines it doesnt help enough. |
| 21:34.25 | mercestes | hmmhesays, they have great technical support though. |
| 21:34.34 | hmmhesays | yeah I just called and they told me to rma it |
| 21:34.37 | hmmhesays | how do I upgrade the firmware? |
| 21:34.58 | mercestes | hmmhesays, it's under /etc/wanpipe/firmwares I'm not 100% on the exact procedure. If Sangoma said RMA it I would. |
| 21:35.15 | mercestes | [hC], what card? |
| 21:36.09 | [hC] | mercestes: sangoma a200d. |
| 21:36.32 | [hC] | mercestes: i dont think its a sangoma thing thouh, it would seem like zaptel has to be told to delay before considering a channel usable again. |
| 21:37.19 | mercestes | [hC], could be the telco, honestly. I think * is pretty quick about clearing up a channel. THere is a variable somewhere that specifies a tiemout to allow for call teardown to fix *other* issues, such as...dead air. |
| 21:37.36 | mercestes | so your problem could be the opposite, asterisk is not waiting long enough to flag a channel as usuable. |
| 21:38.40 | [hC] | mercestes: thats is exactly the problem. thats what im asking. how do i instruct asterisk to wait longer to deal with -whomevers- problem it is |
| 21:38.43 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 21:38.50 | [hC] | mercestes: the problem is asterisk is not waiting long enough, and i dont know how to make it wait longer. |
| 21:38.50 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 21:39.05 | mercestes | KATTY |
| 21:39.08 | TrentCreek | with the wait command? |
| 21:39.10 | Katty | herro |
| 21:39.20 | Katty | mew? |
| 21:39.53 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 21:40.04 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584094.dsl.bell.ca) |
| 21:40.04 | fujin_ | I don't think the wait command is quite what he's looking for |
| 21:40.23 | Katty | ahem. |
| 21:40.37 | TrentCreek | can't use WAIT 2, <next command>? |
| 21:40.54 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 21:40.59 | mercestes | mew. |
| 21:40.59 | fujin_ | no, the issue he's having is relating to a channel reporting that is ready, when it is not |
| 21:41.01 | fujin_ | zap |
| 21:41.01 | fujin_ | too |
| 21:41.41 | [hC] | mercestes: do you know of a configurable option to force chan_zap to wait a little longer between teardown/availability? |
| 21:41.49 | mercestes | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 21:41.54 | mercestes | I think it's tehre under timing parameters. |
| 21:42.04 | [hC] | thanks. |
| 21:42.07 | Netgeeks | Katty: How was your chicken dinner last night? Did you do fajitas? |
| 21:42.30 | [hC] | ahhhhh |
| 21:42.34 | [hC] | . thats what the wink stuff is for. :) |
| 21:42.41 | [hC] | It made no sense since i didnt know what 'wink' meant. |
| 21:42.46 | mercestes | Strom_c is probably a better resource for that. |
| 21:42.57 | peanut- | I love going into a mexican restaruant and mispronouncing the food horribly while ordering |
| 21:43.35 | mercestes | I love going into a mexican restaurant and ordering the "vagina con queso" and looking innocent. |
| 21:43.41 | peanut- | yes I will have one FAH-gee-tah and some extra tor-till-ye-ahs |
| 21:44.15 | peanut- | mercestes: I don't believe that you possess the ability to look innocent. |
| 21:44.16 | *** join/#asterisk moprilo (n=jjohn@201.198.78.23) |
| 21:44.29 | mercestes | True...but I try |
| 21:44.35 | peanut- | it doesn't work. |
| 21:44.38 | peanut- | everyone knows it |
| 21:44.46 | Katty | mercestes: mew! |
| 21:44.55 | mercestes | Katty, did ya miss me? |
| 21:44.56 | Katty | Netgeeks: i got lazy and ended up eating left overs |
| 21:45.06 | Netgeeks | Katty: bah! no fun |
| 21:45.07 | Katty | Netgeeks: feelin pretty tired again tonight :/ |
| 21:45.11 | Katty | Netgeeks: i know :< |
| 21:45.35 | Katty | Netgeeks: i will try it tho! :> |
| 21:45.37 | mercestes | I'm *SICK* |
| 21:45.41 | moprilo | guys.. I have a calls moving through 3 asterisk boxes ending up in a zap, but sometimes it takes up to 3 seconds so that the channel is completely up. |
| 21:45.44 | mercestes | I have a nasty cold... |
| 21:45.45 | Katty | mercestes: really? |
| 21:45.47 | Katty | mercestes: eww |
| 21:45.54 | Katty | mercestes: stay on your side of the continent |
| 21:45.57 | mercestes | so I'm taking cold medicine |
| 21:46.04 | mercestes | s/taking cold medicine/drinking melon balls/ |
| 21:46.08 | Netgeeks | Katty: you need one of these! http://www.hammacher.com/publish/74750.asp |
| 21:46.12 | moprilo | like someone will answer and say.. hello.., but it will take up to 3 seconds to get the first response from the other user (sip0 |
| 21:46.27 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 21:46.33 | moprilo | and is no lag.. |
| 21:46.53 | mercestes | moprilo, Sounds like a networking issue. |
| 21:47.16 | Katty | Netgeeks: ! |
| 21:47.19 | Katty | Netgeeks: izzocute :>>>>>>>> |
| 21:47.59 | mercestes | and lag means nothing to me. |
| 21:49.00 | mercestes | This concept of "ping times" needs to be removed from your mind. Your in voip territory now. The only thing useful ping gives you is assurance that your target responded to an ICMP echo so most likely it is online, or looped back to you. |
| 21:49.33 | Katty | voip is udp |
| 21:49.47 | mercestes | :) |
| 21:50.10 | [TK]D-Fender | Katty, Not entirely true |
| 21:50.26 | Katty | [TK]D-Fender: neither are YOU |
| 21:50.59 | [TK]D-Fender | Katty, No, I am an absolute :) |
| 21:51.52 | Katty | there's no such thing as an absolute smiley. |
| 21:51.57 | [hC] | hmm. zaptel is telling me that its ignoring my wink and flash timer settings. |
| 21:52.11 | Katty | [hC]: don't you hate it when zaptel talks back? |
| 21:52.21 | [hC] | Katty: haha :) |
| 21:53.26 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 21:54.05 | SparFux | Yeah, now works. What the hell, I hate testing. I think it was all about I added insecure=yes to the wrong section, not [sipgate], but [sipphone] accidentally :-( |
| 21:56.40 | [hC] | Grr. of course, there is no clear explanation why asterisk claims to ignore wink settings. |
| 21:56.52 | hmmhesays | yeah that sucks |
| 21:56.57 | hmmhesays | I have to rma a card right away |
| 21:57.27 | [hC] | annnnd i got it. |
| 21:57.32 | [hC] | hmmhesays: what isnt working? |
| 21:57.49 | [hC] | i thought i had a dead sangoma card once, and it turned out it wasnt... |
| 21:59.15 | *** join/#asterisk BBHoss (n=hoss@146.229.183.84) |
| 22:00.50 | *** join/#asterisk StevenElvisda_ (n=Steven_E@202.47.107.60) |
| 22:02.58 | [hC] | I wish there was a cli command to refuse incoming calls (IAX/SIP/ZAP) for a time period, so when it comes time to do a restart, i dont have to wait so long for people to 'get off the phone' what with incoming calls and all |
| 22:03.07 | [hC] | (and i know about 'when convenient' - it still takes a long time) |
| 22:03.11 | BBHoss | just do restart NOW |
| 22:03.17 | BBHoss | instead of just restart |
| 22:03.49 | [hC] | haha. |
| 22:03.59 | [hC] | oh if it were that easy.. |
| 22:04.41 | BBHoss | oh so you want to wait until people are finished, but don't want any new calls so they dont start on another call |
| 22:06.46 | Nugget | that wouldn't be too tough to accomplish via the dialplan |
| 22:09.56 | mercestes | exten => _x.,1,Playback(tt-monkeys) exten => _x.,2,Hangup() |
| 22:10.43 | fujin_ | I would have assumed that when convenient should do such a thing, like, reject incoming calls via sip peers |
| 22:10.51 | fujin_ | or another option to do precisely that |
| 22:12.07 | mercestes | Nothing that useful. |
| 22:12.12 | ajohnson | restart gracefully |
| 22:12.23 | ajohnson | doesn't accept new calls until the restart goes through |
| 22:14.52 | CBU[^_^]M`` | hello... i have sipura 3102 ... do i set it as SIP or ZAP for the FXO port |
| 22:15.17 | BBHoss | #freepbx |
| 22:16.20 | *** join/#asterisk Op3r (n=Op3r@125.212.127.87) |
| 22:21.58 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
| 22:23.15 | RageMax | has anyone had experience with the asterisk appliance? |
| 22:23.19 | BBHoss | yes |
| 22:23.38 | BBHoss | IMHO it sucks |
| 22:23.59 | [hC] | BBHoss: why do you think so? I am looking into using them and would love to hear your feedback |
| 22:24.24 | BBHoss | they dont work with all ITSPs, the DTMF detection didnt work on mine |
| 22:24.27 | RageMax | BBHoss: basically, I told a guy a custom solution would be better, since I tried the digium GUI and I hated it |
| 22:24.41 | BBHoss | yes the digium gui is not all that good |
| 22:24.44 | RageMax | I'm trying to convince him not to go with it until it's been proven in the field |
| 22:24.51 | Qwell | BBHoss: You can add a custom voip provider |
| 22:24.52 | BBHoss | plus, there is near ZERO support for it online |
| 22:25.04 | [hC] | i didnt mind the gui at all. what didnt it do for you? |
| 22:25.09 | BBHoss | yes, but it didnt work with my IAX proivder |
| 22:25.16 | [hC] | I guess im also not afraid to get into the code and make it do what i want... heh |
| 22:25.18 | RageMax | it broke quite a bit in firefox when I used it |
| 22:25.33 | [hC] | its been updated quite a bit recently |
| 22:25.48 | RageMax | and the options aren't there, I spent more time using the GUI than just editing a couple config files |
| 22:25.59 | BBHoss | the main problem with mine was the DTMF detection, and the sound levels on the FXO/FXS ports |
| 22:26.00 | [hC] | Qwell: that reminds me, i added some more functionality into the voice menus section in asterisk-gui, how could I go about getting svn access to resubmit patches? |
| 22:26.22 | fujin_ | be nice to people in here after creating an account |
| 22:26.28 | Qwell | [hC]: bug tracker - at least for now |
| 22:26.34 | fujin_ | ah doh |
| 22:26.36 | fujin_ | wrong project |
| 22:26.39 | [hC] | Qwell: fair enough. |
| 22:26.47 | BBHoss | for the proeconfigured itsps, it works good |
| 22:27.01 | BBHoss | but i couldnt get IAX2 to work with mine |
| 22:27.02 | [hC] | fujin_: when wasnt i nice? |
| 22:27.03 | RageMax | BBHoss: DTMF detection on the FXO ports? |
| 22:27.06 | BBHoss | yes |
| 22:27.15 | BBHoss | i think it would have worked correctly |
| 22:27.16 | RageMax | hrm, how bad? |
| 22:27.17 | Qwell | BBHoss: Did you call Digium support? |
| 22:27.28 | BBHoss | but i had to jack up the gain to be able to hear people |
| 22:27.28 | [hC] | well, under the hood its still all asterisk. i dont see why it would behave any differently. |
| 22:27.35 | *** join/#asterisk sb_mx (n=sb_mx@201.155.80.181) |
| 22:27.42 | [hC] | BBHoss: have you used fxo ports in other installs where you DIDNT have to adjust gains? |
| 22:28.00 | BBHoss | not really, but i've never had this much trouble with DTMF |
| 22:28.03 | [hC] | BBHoss: ive had to mess with gains substantially in every install ive ever done with FXO. Digium, Sangoma, or otherwise. |
| 22:28.03 | BBHoss | anyway |
| 22:28.04 | RageMax | what happens in the GUI if you start making manual changes to the config to get thigns to work? |
| 22:28.12 | BBHoss | ive gotta run right now |
| 22:28.18 | [hC] | RageMax: as long as you make them properly, the gui should pick them up. |
| 22:28.18 | BBHoss | ill be back in a few hour |
| 22:28.21 | BBHoss | hours |
| 22:28.23 | [hC] | RageMax: it depends what you do |
| 22:28.26 | BBHoss | ttyl |
| 22:28.46 | *** join/#asterisk javb (n=javb@190.80.234.104) |
| 22:28.56 | RageMax | I'm definitely going to have teliax in my config, and I never saw them as a provider in the gui |
| 22:29.14 | [hC] | RageMax: so just set up a custom voip provider. |
| 22:29.32 | [hC] | there is an option to do so when you add an iax service provider. |
| 22:29.35 | RageMax | I could never get that working correctly with AsteriskNow (essentially the same right?) |
| 22:29.50 | [hC] | I believe its quite similar, i cant say its the same. |
| 22:29.56 | [hC] | ive never used either of them,r eally.. i do use asterisk-gui though. |
| 22:30.28 | mercestes | guis are for windows. |
| 22:30.38 | RageMax | I also heard that it doesn't do DISA for some odd reason |
| 22:30.54 | [hC] | guis are for people who dont want to be in charge of making stupid decisions and want to hire customer support people to take care of dumb changes. :) |
| 22:31.16 | [hC] | it does disa, i used it just yesterday... however again, i updated the gui from SVN yesterday too |
| 22:31.27 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:31.28 | RageMax | I mean the appliance |
| 22:31.45 | [hC] | why would it not? again - its just asterisk under the hood of it all |
| 22:33.32 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 22:34.08 | JT | [hC]: asteriskNow uses asterisk-gui |
| 22:35.02 | RageMax | the appliance uses asterisk business edition, not asterisknow |
| 22:35.22 | [hC] | JT: I know, i just dont know what elese goes into it, compared to the appliance. |
| 22:35.35 | [hC] | but at the end of the day we're talking about small bolt ons |
| 22:35.41 | mercestes | business addition is still 1.2 isn't it? |
| 22:35.46 | [hC] | ALL of these things use "plain old asterisk" under the hood |
| 22:35.48 | ZX81 | hi all, I have a nortel system which is failing at a customer's and needs replacing. It speaks with a server running tapi, and then there is a CRM package which screen pops etc based on tapi. Most of the asterisk tapi things seem to be for one user - anyone have any ideas what to use? |
| 22:35.51 | [hC] | the only thing that may change is the version number. |
| 22:36.10 | mercestes | ZX81, google asttapi |
| 22:36.13 | ZX81 | heh |
| 22:36.43 | ZX81 | it is just for a single person, sitting on the same machine as the tapi program no? |
| 22:36.57 | ZX81 | and for only one extension |
| 22:37.03 | mercestes | that's how I used it. |
| 22:37.15 | ZX81 | for a single person? |
| 22:39.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:39.56 | Dovid | ~jbot |
| 22:39.57 | jbot | somebody said jbot was a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 22:40.48 | mercestes | ~[TK]D-Fender |
| 22:40.49 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 22:40.58 | J4k3 | ~goatse |
| 22:40.59 | jbot | goatse is at http://www.goatse.cx, or (E@3), or http://www.jurito.net/otro/soldatogoatse.jpg, or http://www.hick.org/goat/, or http://www.fugly.com/media/IMAGES/funny/fugly40316227.jpg |
| 22:41.11 | mercestes | ... |
| 22:41.16 | mercestes | ew. |
| 22:41.36 | mercestes | ~lemonparty |
| 22:41.47 | hmmhesays | ~hmmhesays |
| 22:41.48 | jbot | you are probably not really here... |
| 22:41.49 | mercestes | no jbot, lemonparty is at http://www.lemonparty.com |
| 22:42.20 | mercestes | Anyways, I'm out. l8s |
| 22:43.25 | [hC] | I dont think i want to open that url, do i. |
| 22:43.31 | Qwell | No you do not. |
| 22:43.53 | Strom_M | it's .org |
| 22:43.56 | Strom_M | not .com |
| 22:44.05 | Strom_M | I TAKE NO RESPONSIBILITY FOR YOU FOLLOWING THE LINK |
| 22:44.12 | putnopvut | Can someone summarize what's at that URL? |
| 22:44.19 | putnopvut | I'm guessing old person nudity. |
| 22:44.20 | JT | why |
| 22:44.22 | Qwell | putnopvut: You'd rather not know. |
| 22:44.27 | JT | are you really that scared to look |
| 22:44.27 | putnopvut | Am I close? |
| 22:44.43 | putnopvut | JT: I'm at work. I'm guessing I shouldn't look at work. |
| 22:44.48 | JT | ok |
| 22:45.02 | fujin_ | That's hot. |
| 22:45.05 | fujin_ | www.cupchicks.com is hotter. |
| 22:45.54 | hmmhesays | omfg that was awful |
| 22:45.59 | fujin_ | ;] |
| 22:46.37 | Dovid | ughhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhh |
| 22:46.43 | hmmhesays | can anyone recommend me a decent ethernet bridge? |
| 22:46.53 | hmmhesays | wifi to ethernet |
| 22:47.12 | J4k3 | hmmhesays: one unit or lots of units behind the 'client' end of the bridge? |
| 22:47.22 | J4k3 | and will this be a dedicated p2p bridge, or will it be connecting to an existing AP? |
| 22:47.38 | hmmhesays | I want it to connect to an existing access point |
| 22:48.16 | J4k3 | one MAC or lots of MACs behind the client bridge? |
| 22:48.26 | hmmhesays | variable |
| 22:48.51 | J4k3 | ok, the funky part with standard 802.11 stuff is a standard 'station' adapter should only represent one MAC |
| 22:49.56 | denon | in steps wds |
| 22:50.03 | denon | the standard everyone loves to hate |
| 22:50.11 | hmmhesays | i see |
| 22:50.16 | hmmhesays | well what if I wanted more macs than 1 |
| 22:50.26 | denon | hmmhesays: wds :) |
| 22:50.34 | J4k3 | WDS makes this easiest |
| 22:50.43 | *** part/#asterisk sb_mx (n=sb_mx@201.155.80.181) |
| 22:50.48 | J4k3 | if your AP can't support WDS, it gets interesting |
| 22:51.01 | Dovid | I need sleep. Good Night ev1 |
| 22:51.11 | J4k3 | you can either route or nat, but unless its straight IP that can be a pain in the arse |
| 22:51.51 | hmmhesays | so what bridge should I be looking for? |
| 22:56.47 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 23:00.56 | hmmhesays | this buffalo wireless one seems to have gotten good reviews |
| 23:03.16 | hmmhesays | although I can't find the radio power on it |
| 23:04.28 | pkwong | anyone here have a 7970? |
| 23:04.33 | pkwong | (cisco) |
| 23:05.13 | orakle | those are hot phones |
| 23:05.19 | orakle | colour screen.. mm |
| 23:05.23 | pkwong | heh.. i love mine.. |
| 23:05.32 | pkwong | have the whole house wired up with em. |
| 23:05.40 | orakle | i have a lowly 7940 on my desk. |
| 23:05.47 | pkwong | one little issue just popped up though.. |
| 23:06.15 | pkwong | I finally upgraded to 1.4.13 and 8-0-4.SR3A then to 8-3-2SR1 |
| 23:06.26 | pkwong | and the transfer button stopped working on them. |
| 23:06.29 | pkwong | :( |
| 23:06.37 | orakle | Crap. |
| 23:06.43 | orakle | Can you downgrade the firmware? |
| 23:06.43 | pkwong | i was hoping there was a fix i didn't know about.. |
| 23:06.57 | pkwong | yeah.. i can.. the only one that works is 8-0-2SR1.. |
| 23:07.13 | pkwong | but that throws caller id like "xxxxxxxxxx@asterisk" |
| 23:07.14 | orakle | is there any big disadvantage to rolling back to that revision? |
| 23:07.17 | orakle | oh |
| 23:07.24 | pkwong | yeah.. pretty crappy, huh? |
| 23:07.30 | orakle | Yeah. |
| 23:07.43 | pkwong | heh.. it's either no MWI or no transfer.. lol. |
| 23:07.56 | pkwong | i'd put up a bounty to have that fixed.. |
| 23:08.21 | pkwong | hell.. i'd give someone a 7970 if they fixed it for me.. lol. |
| 23:08.26 | pkwong | (I'm not kidding) |
| 23:08.47 | pkwong | I have 3 extra laying around.. |
| 23:09.02 | pkwong | I got em for a steal.. $180 a piece. |
| 23:09.42 | pkwong | there is one incredibly annoying thing about 7970s though.. |
| 23:09.53 | pkwong | can't turn off the display on demand. |
| 23:10.02 | pkwong | lights up my whole bedroom at night. |
| 23:12.21 | pkwong | at least the transfer button works on the 7940! |
| 23:13.08 | tzanger | hmm... is there a C library for asterisk manager interface? I see many languages up on the wiki but no C |
| 23:13.19 | smultron | does the asterisk appliance support ISDN PRI lines? |
| 23:14.09 | pkwong | yes, but you'll need a T1 card |
| 23:14.12 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 23:14.51 | Bl0w_M0nk | pkwong what are those? |
| 23:15.02 | pkwong | it's a card that hooks up to PRIs |
| 23:15.05 | tzanger | pkwong: the appliance has a PCI port? |
| 23:15.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:16.37 | khronos | Are ther any tools out there I can use to stress test my Asterisk server? |
| 23:17.06 | pkwong | hold on.. on phone |
| 23:17.44 | khronos | I'm interested in finding out how many calls my box can handle cpu wise doing ulaw based calls, ulaw to gsm conversions and gsm to gsm calls over sip. |
| 23:19.31 | hmmhesays | another asterisk box |
| 23:19.37 | hmmhesays | SIPP |
| 23:19.53 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:27.16 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:34.49 | *** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60) |
| 23:36.21 | tzanger | hmm |
| 23:36.53 | tzanger | I can't get CDR manager events to work |
| 23:36.59 | tzanger | I log in, the user has read for cdr |
| 23:37.03 | tzanger | but yet no cdr events get emitted |
| 23:48.25 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 23:49.36 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.13 (Oct. 10, 2007) Addons 1.2.8, 1.4.4 (Oct. 16, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php |
| 23:59.49 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
