| 00:03.02 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 00:07.19 | OmbreNoir | I gather whether you know a link that gives a good example of the file extension.conf to create ...hmm 2-3 postes |
| 00:07.42 | *** join/#asterisk kerx (n=kerx__@76-240-161-68.lightspeed.irvnca.sbcglobal.net) |
| 00:08.48 | bcrisp | source extracted |
| 00:13.07 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 00:17.37 | bcrisp | dlynes: make config? |
| 00:24.00 | dlynes | bcrisp: source extracted, make config, what in blazes are you doing? |
| 00:25.16 | bcrisp | haha |
| 00:26.03 | bcrisp | i moved the kernel source to the linux machine usr/src .. read an article on setting up a link to usr/src/linux |
| 00:26.22 | dlynes | bcrisp: btw...I still don't understand all this bs you're putting yourself through |
| 00:26.34 | dlynes | bcrisp: a 1 and 1 server reimage takes approximately 1 hr |
| 00:26.46 | bcrisp | its the linux part that im afraid of dlynes |
| 00:26.53 | bcrisp | " what you can do is extract the make config for the kernel (if it got compiled into the kernel)" |
| 00:29.22 | kerx | any suggestions on how to bridge an audio file into an active call ? |
| 00:30.01 | dlynes | bcrisp: what 'linux part' are you afraid of? |
| 00:30.19 | bcrisp | well for the past 5 hours im trying to do the same thing all to no avail |
| 00:30.22 | bcrisp | that kind of thing |
| 00:30.25 | dlynes | bcrisp: the server reimage will install a fresh brand spankin' new virgin install of linux |
| 00:30.36 | dlynes | bcrisp: without any corrupted plesk stuff |
| 00:31.01 | dlynes | bcrisp: so people on here will actually be *able* to help you, instead of throwing up their hands in frustration |
| 00:31.22 | bcrisp | is installing kernel headers from source a really difficult task for a novice? |
| 00:31.38 | dlynes | bcrisp: yes |
| 00:32.05 | dlynes | bcrisp: when you're copying it onto a plesk install...even a little bit daunting when plesk isn't involved |
| 00:32.19 | dlynes | for a newbie, that is |
| 00:32.31 | bcrisp | ok, i guess my concern is that if i reimage and cannot get this back up, job = bye bye |
| 00:33.11 | dlynes | bcrisp: that's not an issue....1 and 1 has a serial console app |
| 00:33.12 | bcrisp | so for now ill just give up on installing dahdi |
| 00:33.42 | bcrisp | hmm |
| 00:33.45 | dlynes | bcrisp: so if you fubar your machine so badly that you can't even boot up the kernel, you can still log into the LOM using the serial console, and tell it to boot up from the old kernel |
| 00:34.08 | bcrisp | im not worried about the reimage not working |
| 00:34.16 | bcrisp | im worried about not being able to get * back up |
| 00:34.58 | bcrisp | its ok .. thx |
| 00:35.15 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 00:36.02 | bcrisp | ill just run the demo short of conferencing then ask for permission to re-image |
| 00:37.21 | bcrisp | sorry to have wasted your time dlynes, but thx |
| 00:37.51 | drmessano | Did you hit Ctrl-Alt-Del, Vistamasta? |
| 00:41.30 | *** join/#asterisk pfn (n=pfnguyen@66.218.53.2) |
| 00:49.51 | Katty | i almost fell asleep :< |
| 00:51.53 | lanning | Katty: how do you know you didn't? |
| 01:03.23 | Katty | lanning: i just know |
| 01:03.33 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 01:05.15 | *** join/#asterisk cvnet (n=cvnet@69.172.67.161) |
| 01:05.17 | cvnet | hi all |
| 01:05.28 | Katty | hi |
| 01:06.15 | cvnet | one year ago i read somewhere that there was an web interface for creating voicemail but i dont find it in asterisk website, any clue where i can find that? |
| 01:06.37 | voipmonk | wow |
| 01:06.57 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 01:07.05 | voipmonk | freepbx can do it, trixbox can do it,your perl script you download from a google search might do it , too :) |
| 01:09.34 | Katty | hmm. |
| 01:09.53 | Katty | i seem to recall something about a php thing which allowed you to login and manage your voicemail |
| 01:10.17 | Katty | it was part of the trixbox software package. |
| 01:13.31 | jaytee | silly wabbit, trixbox is for wonks |
| 01:14.24 | Katty | i tinkered with it for awhile, just to see it |
| 01:16.09 | jaytee | same here |
| 01:16.55 | jaytee | and *Now as well. I prefer *Now 1.5 to trixbox of any version |
| 01:18.26 | *** join/#asterisk Ta^3 (n=tacvbo@189.146.170.87) |
| 01:20.53 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:22.18 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:24.19 | Katty | :< |
| 01:24.26 | Katty | it keeps getting cold and colder outside. |
| 01:24.29 | Katty | like it's winter or something |
| 01:25.52 | p3nguin | Not much longer and it will be. |
| 01:26.42 | dlynes | cvnet: contrib/scripts/vmail.cgi in your asterisk source |
| 01:26.57 | Katty | p3nguin: we're not discussing that |
| 01:28.15 | Katty | p3nguin: talking about winter is just bad jumu |
| 01:28.18 | Katty | juju |
| 01:28.44 | p3nguin | I'm not a big fan of it, but I don't despise it. |
| 01:29.02 | Katty | you're very fortunate )= |
| 01:29.09 | Katty | i'm stuck with a perminante ache all winter |
| 01:33.48 | *** join/#asterisk xheliox (n=jeff@207.140.8.67.cfl.res.rr.com) |
| 01:34.25 | xheliox | Howdy all. Running SVN-branch-1.6.2-r233615, anyone notice that the voicemail format config isn't being honored? |
| 01:35.08 | xheliox | I have wav49 specified first, but the VMs are recorded only as GSM (attach=yes|delete=1). |
| 01:39.18 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 01:39.39 | jaytee | I think winter is God's way of saying "Go fuck yourselves!" |
| 01:40.04 | cvnet | i love winter, Canadian winter, if we didnt had winter we would not appreciate summer |
| 01:40.37 | cvnet | its soo nice when its -40 out there |
| 01:42.26 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 01:43.53 | kerx | any suggestions on how to bridge an audio file into an active call? |
| 01:44.32 | Katty | use a call file |
| 01:44.57 | kerx | Katty, I have an active call, and I'd like an audio file to be played, and controlled (pause, stop, etc.) |
| 01:45.04 | kerx | A call file would be the suggested route ? |
| 01:45.21 | kerx | The audio should be streamed to both leg's |
| 01:45.23 | dlynes | cvnet: -40? what part of Canada are you in? Nunavut? |
| 01:46.58 | lanning | kerx, sounds like you need to use meetme for the call (transfer both calls to meetme then play the sound in the conference.) |
| 01:47.28 | kerx | darn, just the thing i was avoiding |
| 01:48.04 | kerx | quick question about that. if the call is part of a answered queue. if i have associated call recording's (Monitor's) on the call, will those Monitor's drop after they are transferred into a MeetMe? |
| 01:48.13 | kerx | lanning, is there any other route? |
| 01:48.40 | lanning | if you are monitoring the channel, the monitoring will not drop. |
| 01:49.05 | kerx | well, i have 2 monitor's running on that channel, plus the regular call recording saveagentcalls=yes. |
| 01:49.07 | lanning | you can't inject audio into a stream |
| 01:49.20 | kerx | Ok, so the only route is to transfer the two leg's into a MeetMe ? |
| 01:49.43 | lanning | yes, the meetme has the audio sync capability |
| 01:49.44 | kerx | What would I use to play the audio, and give the operator (agent) the ability to Pause the audio ? |
| 01:49.54 | kerx | Inside of the MeetMe that is? |
| 01:50.45 | drmessano | A boombox |
| 01:50.49 | lanning | that gets more complicated. |
| 01:51.08 | drmessano | Try app_ghettoblaster |
| 01:51.15 | lanning | heh |
| 01:51.31 | kerx | hey dr |
| 01:51.45 | drmessano | http://en.wikipedia.org/wiki/File:Ghettoblaster-family.jpg |
| 01:51.47 | kerx | can i just use any boombox? from Radioshack? |
| 01:51.53 | drmessano | I really wanna break dance now |
| 01:52.20 | kerx | on a serious note, i would appreciate any suggestions :) |
| 01:52.21 | *** join/#asterisk kam187 (n=kam187@81-179-8-102.dsl.pipex.com) |
| 01:52.29 | drmessano | and listen to songs that are 5% lyric, 10% thump, and 95% scratching |
| 01:52.31 | kam187 | hi |
| 01:52.33 | kerx | play audio into a MeetMe with the ability to Pause |
| 01:52.36 | kam187 | anyone here use chan_sebi? |
| 01:52.59 | drmessano | I asked you what a sebi was, and you ignored me |
| 01:53.09 | kam187 | sorry i had to go :( |
| 01:53.14 | drmessano | You want to do this the hard way, don't you? |
| 01:53.20 | dlynes | drmessano: and that 95% scratching i'm hoping is not the jock strap |
| 01:53.25 | kam187 | its a channel driver that uses usb data modem to make gsm voice calls |
| 01:53.29 | drmessano | Fender, get my pastebin and tire iron, pls |
| 01:53.49 | kam187 | i'm getting choppy audio, i think the PCM rate is wrong |
| 01:54.24 | dlynes | kam187: i would think it's probably the usb data modem...usb generates a hell of a lot of interrupts |
| 01:54.37 | kam187 | hmm it works fine on windows |
| 01:54.48 | kam187 | its not working/choppy/working |
| 01:54.56 | dlynes | kam187: cat /proc/interrupts and you can see how many interrupts it's generated |
| 01:54.59 | kam187 | its a constant chopping like the PCM rate is wrong |
| 01:55.09 | drmessano | Linux is a lot less stable than Windows |
| 01:55.16 | drmessano | Well known fact |
| 01:55.33 | kam187 | but people report this same setup working np |
| 01:55.36 | dlynes | kam187: and you've got asterisk running in windows? |
| 01:55.50 | dlynes | kam187: he's pulling your leg |
| 01:55.54 | kam187 | no, i mean the same dongle sending pcm to the com port |
| 01:56.13 | kam187 | these modesm just send 8khz PCM on a com port, and AT commands on the other com port |
| 02:00.24 | kerx | last ask for help. any suggestions on how to playback audio in a MeetMe that can be controlled (*paused*) by the operator? |
| 02:00.56 | drmessano | I'm not really sure, Pete |
| 02:02.00 | *** join/#asterisk ming_zym (n=ming_zym@114.251.86.0) |
| 02:02.09 | drmessano | My world just fell apart around me |
| 02:02.11 | kerx | drmessano, how do you know my name ? |
| 02:02.31 | drmessano | My former record of 160 days uptime for Windows 2000 was shattered through laziness |
| 02:02.55 | drmessano | Just shut down and restarted a machine with 226 days uptime |
| 02:05.15 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 02:12.52 | dlynes | kerx: call files, and dtmf? |
| 02:14.09 | *** join/#asterisk ticoit (n=ticoit@201.191.190.231) |
| 02:25.15 | kam187 | how do u turn on debug messages in asterisk? |
| 02:25.18 | kam187 | -d doesnt seem to show them |
| 02:26.35 | kam187 | nvm found it :) |
| 02:27.24 | *** join/#asterisk doolph (n=doolph@201.218.73.208) |
| 02:35.16 | *** join/#asterisk chendy (n=chatzill@113.91.38.115) |
| 02:36.07 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:40.40 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 02:40.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:46.04 | *** part/#asterisk korihor (n=korihor@190.205.251.97) |
| 02:46.37 | Katty | ryan's watching a movie that sounds very violet :< |
| 02:46.39 | Katty | violent. |
| 02:47.30 | carrar | violent can be violet |
| 02:48.15 | Katty | http://i.imgur.com/om44Z.jpg <- wow. |
| 02:48.37 | carrar | haha |
| 02:49.11 | carrar | cept you gotta read it before you die in the race car |
| 02:49.47 | carrar | Violet is a death rock girl in her late teens |
| 02:49.54 | carrar | http://www.amazon.com/Violent-Violet-Tara-Vanflower/dp/141378156X |
| 02:53.28 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 02:56.19 | *** join/#asterisk benklop (n=ben@97-118-168-212.hlrn.qwest.net) |
| 02:59.58 | benklop | hey guys! I recently aquired a "brooktrout technology TRxStream Series" PCI card, it appears to have 4 lines in a single rj45 jack, but I have no idea what its capabilities are or anything else about it. the only number I can find on it is TR1000-P4V-4L . I'd love to know if I can use this thing in asterisk, either as an fxo or an fxs, depending on what iot actually does of course. I know it was used in a machine that dealt with telephony in |
| 03:00.00 | benklop | some way, but that's all I really know. |
| 03:00.21 | benklop | Any ideas where I can at least find some more useful info? |
| 03:00.35 | benklop | or hopefully anything about drivers |
| 03:02.35 | Deeewayne | benklop, try dialogic.com |
| 03:03.12 | *** join/#asterisk minotaur01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 03:03.40 | minotaur01 | could someone help me diagnose a iax issue |
| 03:05.32 | minotaur01 | im upgrading from version 1.6.0.10 to .19 but when i upgrade the incoming calls get a busy signal |
| 03:05.58 | minotaur01 | the can shows up in debug but its just busy |
| 03:08.34 | xheliox | Howdy all. Running SVN-branch-1.6.2-r233615, anyone notice that the voicemail format config isn't being honored? |
| 03:08.37 | xheliox | I have wav49 specified first, but the VMs are recorded only as GSM (attach=yes|delete=1). |
| 03:15.38 | dlynes | xheliox: maybe a certain module isn't getting loaded by default for you, anymore |
| 03:16.28 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 03:17.14 | dlynes | xheliox: have you tried module reload format_wav.so or module reload format_wav_gsm.so? (I'm not sure which one it is for wav49 |
| 03:17.36 | xheliox | Actually, looks like like it.. ARNING[26735] file.c: ignoring unknown format 'wav49' |
| 03:18.24 | dlynes | xheliox: actually...I think it's format_pcm.so |
| 03:19.09 | dlynes | xheliox: nvm.... format_wav_gsm.so is for wav49 |
| 03:19.19 | dlynes | xheliox: just confirmed by looking at the C code |
| 03:20.27 | xheliox | [Dec 7 22:20:09] WARNING[10992]: loader.c:771 load_resource: Module 'format_pcm.so' already exists. |
| 03:20.43 | dlynes | xheliox: format_wav_gsm.so, not format_pcm.so |
| 03:20.44 | xheliox | [Dec 7 22:20:39] WARNING[10992]: loader.c:771 load_resource: Module 'format_wav_gsm.so' already exists. |
| 03:20.49 | dlynes | hrm |
| 03:22.09 | xheliox | On this box, not the end of the world, I run the bleeding edge on this box for the specific purpose of helping out Asterisk -- I'll see if I can't run it down and report a bug. |
| 03:22.19 | xheliox | Just thought I'd bring it up. |
| 03:24.15 | minotaur01 | got a problem with incoming calls after upgrading to version 1.6.0.19 from version .10 the call shows up in iax2 debug but it dosnt go in to the dial plan |
| 03:24.23 | minotaur01 | any ideas? |
| 03:27.01 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 03:27.47 | CareBear\ | Bye ppl - be back another time. |
| 03:27.49 | *** part/#asterisk CareBear\ (i=peter@stuge.se) |
| 03:30.46 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:38.25 | *** join/#asterisk jmacz (n=jmacz@190.25.7.149) |
| 03:39.10 | *** join/#asterisk linuxviewer (n=na@173.10.16.233) |
| 03:40.55 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 03:42.02 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 03:42.44 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 03:44.54 | *** join/#asterisk krondorl (n=chatzill@216.191.33.42) |
| 03:50.27 | *** join/#asterisk jmacz (n=jmacz@190.25.7.149) |
| 03:50.42 | dlynes | xheliox: try core show file formats |
| 03:51.17 | dlynes | xheliox: do you get 'gsm wav49 WAV|wav49'? |
| 03:52.20 | dlynes | minotaur01: can you show a call happening (core set verbose 100)? |
| 03:52.57 | *** join/#asterisk codefreeze-lap (n=murf@h125.84.82.206.static.ip.windstream.net) |
| 03:52.58 | dlynes | minotaur01: I don't want to see all the iax2 debug info |
| 03:53.05 | *** part/#asterisk k-man (n=jason@unaffiliated/k-man) |
| 03:53.09 | dlynes | minotaur01: just normal verbose output |
| 03:53.19 | dlynes | minotaur01: and pastebin it? |
| 03:53.21 | dlynes | ~pb |
| 03:53.22 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 03:53.24 | *** part/#asterisk codefreeze-lap (n=murf@h125.84.82.206.static.ip.windstream.net) |
| 03:58.22 | minotaur01 | sorry i was afk ill paste the debug output on pastebin |
| 03:59.37 | dlynes | holy sheepshit....1065KB/sec |
| 03:59.44 | dlynes | on cable at that |
| 04:00.50 | drmessano | 10 megs? |
| 04:00.55 | drmessano | I get that daily |
| 04:01.08 | dlynes | on cable? |
| 04:01.24 | *** join/#asterisk blkry (n=chatzill@96.37.24.97) |
| 04:01.30 | dlynes | i.e. on residential Internet? |
| 04:01.35 | minotaur01 | yes |
| 04:01.42 | minotaur01 | i get almost 14m |
| 04:01.45 | dlynes | damn |
| 04:01.50 | minotaur01 | one residential |
| 04:01.54 | minotaur01 | on* |
| 04:02.03 | dlynes | minotaur01: and you're in hamilton? |
| 04:02.13 | minotaur01 | lol yes |
| 04:02.17 | minotaur01 | why? |
| 04:02.22 | dlynes | I'm in Caledonia :) |
| 04:02.38 | dlynes | I'm just on Bell, not mountain cable |
| 04:02.44 | p3nguin | We've got a 60 Mbps package available. |
| 04:03.23 | minotaur01 | dlynes: you dont work for mountain by any chance? |
| 04:03.42 | dlynes | minotaur01: nope...I do subcontracting for a couple of vancouver voip companies |
| 04:03.57 | kerx | dlynes, how can i use a call file to bridge in audio to an active call? |
| 04:04.10 | kerx | what would i use to read in DTMF actively to be able to pause the channel ? |
| 04:04.18 | dlynes | kerx: don't know...just an idea...I've heard of people on here doing it, using agi scripts, though |
| 04:04.23 | minotaur01 | dlynes: well i get my iax trunk from mountain |
| 04:04.25 | kerx | i see |
| 04:04.31 | dlynes | minotaur01: they do iax2? |
| 04:04.35 | minotaur01 | dlynes: http://pastebin.ca/1706409 |
| 04:04.37 | minotaur01 | yes |
| 04:04.43 | dlynes | minotaur01: damn |
| 04:04.43 | minotaur01 | over fiber |
| 04:04.44 | kerx | i think i'm going to be forced to use the MeetMe method. i want to have a web application to manage the Pause, and Start up of the audio file |
| 04:04.53 | minotaur01 | im the first one |
| 04:04.57 | p3nguin | Unfortunately I don't have a DOCSIS 3 modem, so I can't get the Ultra 60 package. |
| 04:05.00 | minotaur01 | special request |
| 04:05.07 | dlynes | minotaur01: how much a month for the fiber? |
| 04:05.17 | minotaur01 | $150 |
| 04:05.35 | minotaur01 | plus the chanels at 29/each |
| 04:06.23 | dlynes | minotaur01: now you say that your iax2 call isn't going through...is it not going through on the mountain cable side, or not going through on your asterisk side? |
| 04:06.36 | dlynes | minotaur01: iax2 channels, right? |
| 04:06.43 | minotaur01 | yes |
| 04:06.52 | drmessano | dlynes: Comcast runs up to 22 megs on DOCSIS 2.0 |
| 04:07.05 | minotaur01 | i can call out but when i call in i get busy |
| 04:07.14 | dlynes | drmessano: yeah..none of the american cable companies provide internet in Canada...so it doesn't matter |
| 04:07.15 | drmessano | I get around 15 most of the time |
| 04:07.18 | minotaur01 | call the number 9053879287 |
| 04:07.21 | dlynes | drmessano: i know rr supplies fiber, too |
| 04:07.27 | dlynes | drmessano: but it doesn't help me very much |
| 04:08.13 | minotaur01 | dlynes: if i down grade to v1.6.0.10 everything is fine |
| 04:08.36 | p3nguin | DOCSIS 2 is limited to ~30 Mbit. If you go to DOCSIS 3, they can bond multiple channels together and then you're cooking with gas. |
| 04:08.41 | dlynes | minotaur01: give me a couple...just in the middle of a few downloads |
| 04:08.49 | minotaur01 | np |
| 04:08.57 | dlynes | minotaur01: i want to monitor both legs of my call when I make that call |
| 04:09.07 | minotaur01 | k |
| 04:09.13 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 04:09.22 | dlynes | minotaur01: is it just your inbound that's not working? Or is your outbound not working, either? |
| 04:09.30 | minotaur01 | inbound |
| 04:09.36 | minotaur01 | only |
| 04:09.52 | dlynes | ok |
| 04:10.00 | drmessano | p3nguin: DOCSIS 3 is more efficient for them too.. I understand 20 megs is a soft limit they try to hit for DOCSIS 2.0 before pushing customers and the nodes to 3.0, but Comcast is going as high as 22 megs in some areas where the node rollout is slowed |
| 04:10.03 | dlynes | Just getting 1.6.1.11 installed onto a new box |
| 04:10.06 | minotaur01 | it hits the box but dosnt enter the dial plan |
| 04:11.09 | dlynes | Yeah...I'm getting this 1-2MB/s out of Vancouver...not locally here |
| 04:11.37 | dlynes | I'm stuck on that joke that bell calls service |
| 04:11.47 | minotaur01 | lol |
| 04:11.49 | p3nguin | We've got so much available speed floating around my node that I can actually get over 31 Mbps. I can go to other areas around town and only see 17 or so. |
| 04:12.04 | minotaur01 | get mountian so much better |
| 04:12.20 | dlynes | minotaur01: yeah...but I'm trying to control costs right now |
| 04:12.22 | minotaur01 | there up to 10m at home i think |
| 04:12.38 | dlynes | minotaur01: moving into hamilton the end of December...should be able to get more contracting work then |
| 04:12.44 | p3nguin | 3872kB/s right now |
| 04:12.46 | dlynes | minotaur01: then I can afford to get mountain cable instead |
| 04:12.51 | p3nguin | Just downloaded 100M file in 28 seconds. |
| 04:13.01 | minotaur01 | where in hamilton? |
| 04:13.04 | drmessano | p3nguin: I get 15 on my 6 meg service here.. Some guys I know get 25 on their 8 meg service in other areas of town.. But then again, I live in the ghetto |
| 04:13.06 | dlynes | Durand |
| 04:13.13 | dlynes | Bay and Duke |
| 04:13.16 | drmessano | p3nguin: I have drug dealers that protect me from other drug dealers |
| 04:13.25 | minotaur01 | thats cogecos area |
| 04:13.26 | p3nguin | heh |
| 04:13.28 | minotaur01 | 16m |
| 04:13.31 | dlynes | ah |
| 04:13.31 | minotaur01 | at home |
| 04:13.35 | dlynes | better than mountain? |
| 04:13.40 | minotaur01 | yes |
| 04:13.43 | dlynes | cool |
| 04:13.46 | dlynes | better service, too? |
| 04:13.49 | dlynes | cheaper? |
| 04:13.52 | minotaur01 | same price more speed |
| 04:14.01 | minotaur01 | 16m |
| 04:14.02 | dlynes | yeah...don't like mountain's price |
| 04:14.06 | dlynes | that's why i went with bell |
| 04:14.09 | minotaur01 | fully open modem |
| 04:14.14 | dlynes | $35/mo for Internet |
| 04:14.14 | p3nguin | curl http://speedtest2.eastlink.ca/superlarge.bin > /dev/null |
| 04:14.36 | dlynes | p3nguin: btw...was that you that suggested we check with eastlink for pei? |
| 04:14.44 | minotaur01 | moutain is only $39 |
| 04:14.53 | dlynes | minotaur01: not in Caledonia |
| 04:15.02 | p3nguin | dlynes: Doesn't sound familiar. |
| 04:15.09 | dlynes | minotaur01: they wanted $70/mo here, if I didn't have tv or phone lines from them |
| 04:15.24 | minotaur01 | what.... thats nuts |
| 04:15.29 | dlynes | Yeah...no kidding |
| 04:15.55 | dlynes | so $39 a month is just cable internet...no basic cable, right? |
| 04:15.58 | minotaur01 | i pay $80 with tv |
| 04:16.31 | minotaur01 | internet on its own is $44 |
| 04:16.36 | dlynes | Does anyone know what the new files asterisk-moh-opsound are all about? |
| 04:16.40 | dlynes | cool |
| 04:16.48 | dlynes | might be worth switching from bell, then |
| 04:16.56 | dlynes | their service sucks the big kahunas |
| 04:16.57 | *** join/#asterisk yahsyn (n=yahsyn@c-24-6-112-203.hsd1.ca.comcast.net) |
| 04:17.08 | dlynes | it's even worse than telus, if that's possible |
| 04:17.29 | minotaur01 | lol i got telus for phone |
| 04:18.10 | dlynes | minotaur01: you ever try calling their voice response autoattendant hell? |
| 04:18.14 | minotaur01 | i also got a full pri from telus for only $460/month |
| 04:18.35 | dlynes | Yeah...they're more competitive here than they are in BC |
| 04:18.51 | minotaur01 | never tried to call |
| 04:18.57 | minotaur01 | is it bad? |
| 04:19.02 | dlynes | In BC, you're looking about $1100/mo for a PRI with all the other stuff you need to make it work |
| 04:19.08 | dlynes | that's from telus |
| 04:19.27 | dlynes | they're a huge ripoff and tariffed up the wazoo |
| 04:19.28 | minotaur01 | that was my bell quote here |
| 04:19.44 | dlynes | minotaur01: bell quoted me $350/mo for a pri |
| 04:19.55 | dlynes | minotaur01: that was in caledonia, of all places |
| 04:19.55 | minotaur01 | really... |
| 04:20.19 | minotaur01 | that i don't know what the hell the quoted me |
| 04:20.23 | dlynes | that's with none of the extra stuff you need to make it work, but still |
| 04:20.31 | dlynes | well |
| 04:20.36 | dlynes | I've got an inside connection :) |
| 04:20.50 | minotaur01 | ah |
| 04:20.52 | minotaur01 | okay |
| 04:21.00 | dlynes | I've got a connection here with Bell that I know from Alberta |
| 04:21.18 | dlynes | But your $460/mo is probably retail |
| 04:21.22 | dlynes | my $350/mo is wholesale |
| 04:21.31 | minotaur01 | yah it is |
| 04:22.03 | minotaur01 | are you thinking of starting a voip provider |
| 04:22.08 | dlynes | On my last download |
| 04:22.15 | dlynes | I work for two voip providers |
| 04:22.19 | dlynes | as a subcontractor |
| 04:22.25 | dlynes | One's a clec |
| 04:22.32 | dlynes | The other will probably become a clec soon |
| 04:23.15 | minotaur01 | clec? |
| 04:23.30 | dlynes | competitive local exchange carrier |
| 04:23.36 | dlynes | Telus is one of those in Ontario |
| 04:23.38 | dlynes | So is Allstream |
| 04:23.41 | minotaur01 | ah |
| 04:23.49 | *** join/#asterisk TJNII (n=TJNII@207.189.199.62) |
| 04:23.52 | dlynes | Bell is the ilec (incumbent local exchange carrier) |
| 04:24.47 | minotaur01 | in ontario that a olec in eveywhere else |
| 04:24.56 | minotaur01 | there* |
| 04:25.12 | minotaur01 | lol i cant type today |
| 04:25.20 | minotaur01 | clec* |
| 04:26.19 | minotaur01 | dlynes: do you mostly write dial plans or create guis? |
| 04:27.52 | dlynes | nah...a lot of companies are just itsp's, not actually clecs |
| 04:28.02 | drmessano | dlynes steals dialplans from gypsies |
| 04:28.02 | dlynes | And so they can't get the wholesale pricing from the ilecs |
| 04:28.12 | minotaur01 | lol |
| 04:28.37 | dlynes | I only use guis on two machines and soon to be only one...finally won the argument to get rid of freepbx on one of them |
| 04:28.57 | dlynes | I'm working on getting rid of the gui on the other...it's coming soon...just not here yet |
| 04:29.05 | p3nguin | I never figured out why people insist on having a gui for a lot of crap. |
| 04:29.16 | dlynes | p3nguin: because the suits want them |
| 04:29.32 | dlynes | p3nguin: and the 'asterisk programmers' that they hire at minimum wage |
| 04:29.37 | minotaur01 | lol i use freepbx |
| 04:29.37 | minotaur01 | but its a dealership |
| 04:29.37 | p3nguin | The suits don't need to be dicking around with a gui anyway. |
| 04:29.56 | minotaur01 | not a service provider |
| 04:30.02 | dlynes | p3nguin: but they want to hire the guys at minimum wage that don't know how to use the cli |
| 04:30.15 | dlynes | p3nguin: because the guys that use the cli are too expensive |
| 04:30.21 | p3nguin | oh |
| 04:30.49 | dlynes | p3nguin: and then they wonder why the guy doesn't know how to fix something when he breaks it :) |
| 04:31.21 | dlynes | joy |
| 04:31.35 | dlynes | another wonderful predicament I've gotten myself into |
| 04:31.48 | dlynes | I let the boss talk me into using solid state hardware |
| 04:31.56 | dlynes | and now the solid state hardware's locked up |
| 04:32.30 | minotaur01 | lol i just put one of those in the phone server |
| 04:32.39 | minotaur01 | what brand/model |
| 04:32.42 | minotaur01 | ? |
| 04:32.59 | dlynes | nope...never mind |
| 04:33.14 | dlynes | the ethernet connection just went for a shit during a compile for some reson |
| 04:33.23 | *** part/#asterisk yahsyn (n=yahsyn@c-24-6-112-203.hsd1.ca.comcast.net) |
| 04:33.52 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 04:38.24 | dlynes | let's see if it craps out again |
| 04:40.37 | minotaur01 | dlynes: have you done any work with SMS |
| 04:41.02 | dlynes | minotaur01: very limited |
| 04:41.20 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 04:42.30 | dlynes | minotaur01: mostly just interfacing with fido.ca and 163.com |
| 04:42.51 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 04:43.36 | dlynes | grrr |
| 04:43.46 | minotaur01 | dlynes: fido has a SMS service for companies |
| 04:43.57 | dlynes | this machine's starting to piss me off now....2nd time it froze my connection |
| 04:44.58 | dlynes | ok...no wonder |
| 04:45.16 | dlynes | Marvell network cards if i remember correctly are complete garbage |
| 04:48.04 | *** join/#asterisk dkirker (n=dkirker@udp519393uds.csc.calpoly.edu) |
| 04:50.39 | *** join/#asterisk chendy (n=chatzill@113.91.39.184) |
| 04:53.23 | dlynes | minotaur01: did you see that call attempt just now? |
| 04:53.45 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) |
| 04:56.04 | minotaur01 | wasnt wathing |
| 04:56.36 | minotaur01 | i am not |
| 04:56.38 | minotaur01 | now |
| 04:56.47 | JAMMAN2110 | Did anyone ever find a guide / anything on how to get Asterisk / FreePBX to notice an incoming call on FXO, ring the VoIP phones but not answer the actual line until someone answers from a VoIP phone so that POTS phones could still answer? |
| 04:57.42 | dlynes | ok....going |
| 04:57.56 | dlynes | minotaur01: see it? |
| 04:58.05 | dlynes | minotaur01: should have been coming from 416-342-1724 |
| 04:58.06 | minotaur01 | yup |
| 04:58.53 | minotaur01 | http://pastebin.ca/1706464 |
| 04:59.02 | dlynes | minotaur01: can you pastebin your iax.conf? scrub passwords and possibly ip's if you need to |
| 04:59.21 | minotaur01 | k |
| 04:59.40 | dlynes | minotaur01: i want to see where 24.215.5.37 is heading into |
| 05:00.02 | dlynes | minotaur01: username should be iax-ham2-ujtoyota |
| 05:03.52 | minotaur01 | http://pastebin.ca/1706470 |
| 05:04.51 | dlynes | minotaur01: and can I see your 'from-trunk' dialplan context? |
| 05:05.10 | minotaur01 | dlynes: this config work fine under v1.6.0.10 just so you know |
| 05:05.32 | dlynes | minotaur01: also, don't you have [general] defined above all of those iax2 options? |
| 05:05.40 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 05:05.43 | bcrisp | hey |
| 05:06.04 | dlynes | ho |
| 05:06.06 | bcrisp | i keep getting calls from blocked #s and they hang up without speaking.. what is it? |
| 05:06.22 | dlynes | blocked |
| 05:06.37 | bcrisp | they hang up as soon as i answer |
| 05:06.41 | dlynes | you don't want to talk to them, right? |
| 05:06.43 | bcrisp | what would be the point? |
| 05:07.07 | dlynes | and they're not talking to you? |
| 05:07.11 | dlynes | so what's the problem? |
| 05:07.11 | bcrisp | no |
| 05:07.20 | bcrisp | my problem is why they're dialing me |
| 05:07.38 | dlynes | who knows? |
| 05:07.43 | bcrisp | probing for in-use #s or something to defraud people hmm |
| 05:07.57 | dlynes | how is that an asterisk question? |
| 05:08.05 | dlynes | that happens regularly on land lines, too |
| 05:08.07 | bcrisp | its not.. just thought u guys were experts |
| 05:08.23 | bcrisp | ud know all the nasty little tricks they use |
| 05:08.26 | dlynes | on psychoanalysis? |
| 05:08.51 | dlynes | a lot of times people just do it to see if you're home, so they know when's a good time to break in and steal stuff, too |
| 05:09.01 | bcrisp | ya.. i got robbed in june |
| 05:09.25 | dlynes | Or they could be checking to see if it's a fax number or not, so they can spam it |
| 05:09.37 | bcrisp | ya i bet its something like that |
| 05:09.39 | dlynes | or they could be checking to see if a modem's listening to it, so they can try to hack into the network |
| 05:09.46 | dlynes | there's any number of reasons |
| 05:09.52 | JAMMAN2110 | Like the gibsons |
| 05:09.54 | JAMMAN2110 | no wai |
| 05:10.50 | minotaur01 | dlynes: i accidentaly cut off the [general]... and everything is posted |
| 05:11.14 | dlynes | bcrisp: besides...you're in the phillippines...could just be perverts in north america trying to find the bawdy houses |
| 05:11.44 | dlynes | minotaur01: dialplan? |
| 05:12.35 | dlynes | minotaur01: still waiting for it |
| 05:13.10 | minotaur01 | sorry did press submit |
| 05:14.35 | dlynes | minotaur01: ? |
| 05:14.40 | minotaur01 | yes |
| 05:15.04 | minotaur01 | you want the entire file |
| 05:15.10 | minotaur01 | ? |
| 05:15.30 | cmendes0101 | I'm having troubles with a new server getting extensions to work with my SIP users. The users are set to an [allowed] context with a couple extensions in there but i keep getting this "chan_sip.c:18522 handle_request_invite: Call from '201' to extension '200' rejected because extension not found." |
| 05:15.41 | dlynes | minotaur01: just the from-trunk context |
| 05:15.46 | cmendes0101 | Is there a way in CLI to make it display extensions that are loaded? |
| 05:15.55 | dlynes | minotaur01: and any contexts it includes or goes to |
| 05:16.07 | minotaur01 | k |
| 05:16.10 | dlynes | cmendes0101: dialplan show |
| 05:16.13 | p3nguin | cmendes0101: dialplan show |
| 05:16.53 | p3nguin | More specifically, you can do "dialplan show allowed" for the [allowed] context. |
| 05:17.03 | dlynes | cmendes0101: also sip show user 201 |
| 05:17.18 | dlynes | cmendes0101: to make sure the sip user is actually going to the context you think it is |
| 05:17.33 | p3nguin | You could also paste your extensions.conf file into the pastebin so we can examine it. |
| 05:17.48 | cmendes0101 | dlynes, p3nguin: Thanks, for some reason its not loading the context |
| 05:17.49 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 05:18.14 | dlynes | cmendes0101: core set verbose 100, and then do dialplan reload |
| 05:18.25 | dlynes | cmendes0101: see if maybe you don't get some warnings or errors showing up |
| 05:21.01 | dlynes | minotaur01: oh yeah..one other thing |
| 05:21.11 | dlynes | minotaur01: where's your iax2 peer definition? |
| 05:21.47 | dlynes | minotaur01: nvm...you're not making outbound calls...only inbound |
| 05:24.45 | dlynes | minotaur01: do you almost have that dialplan ready? |
| 05:24.53 | dlynes | minotaur01: i'm not here much longer tonight |
| 05:27.11 | minotaur01 | sorry there is a lot of crap to go through |
| 05:27.14 | minotaur01 | http://pastebin.ca/1706487 |
| 05:28.34 | minotaur01 | its the basic fpbx dial plan just so you know |
| 05:30.41 | dlynes | minotaur01: ok...anyways...I use freepbx, but I can't tell why it's not working from looking at their mess of config files |
| 05:30.50 | dlynes | minotaur01: and it's also too late now |
| 05:30.59 | minotaur01 | np |
| 05:31.04 | minotaur01 | maybe tomorrow |
| 05:31.13 | dlynes | minotaur01: if you would've told me originally that you were using freepbx, i could've gotten you to go through the gui on #freepbx |
| 05:31.23 | p3nguin | How odd. I know my cell phone makes anonymous calls, but I just looked in my incoming call record at home and my cell number is showing up. |
| 05:31.30 | dlynes | minotaur01: almost nobody in this channel knows anything about freepbx |
| 05:32.01 | minotaur01 | i know |
| 05:32.40 | dlynes | p3nguin: you have your cell phone set to send my number (on) or send my number (set by network), and the network's sending it |
| 05:32.49 | p3nguin | lol |
| 05:32.56 | p3nguin | Man, I have a StarTAC. |
| 05:33.02 | dlynes | whatever that is |
| 05:33.19 | p3nguin | There's no such setting. And I have line blocking. |
| 05:33.21 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 05:33.23 | dlynes | anyways...i'm gone |
| 05:33.25 | dlynes | need sleep |
| 05:33.37 | minotaur01 | gnite |
| 05:34.25 | p3nguin | It only does it on DIDs from a certain ITSP, but shows anonymous when dialing some other DIDs I have. |
| 05:34.39 | p3nguin | It's kinda weirding me out. |
| 05:40.40 | minotaur01 | gnite all |
| 06:02.44 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 06:06.47 | kam187 | hmm found my problem |
| 06:06.56 | kam187 | seems like asterisk is flooding the modem with voice |
| 06:06.58 | kam187 | not sure why |
| 06:07.02 | kam187 | whats the latest stable 1.6? |
| 06:11.41 | drmessano | 1.6.0 1.6.1 or 1.6.2? |
| 06:12.08 | drmessano | You may as well have asked what the latest stable Asterisk 1 is... |
| 06:13.03 | kam187 | 1.6.1 |
| 06:13.11 | drmessano | --> /topic |
| 06:13.11 | kam187 | i'm using 1.6.11 |
| 06:13.42 | drmessano | There is no 1.6.11 |
| 06:13.48 | kam187 | err 1.6.1.11 |
| 06:13.54 | kam187 | same as the topic :( |
| 06:14.23 | drmessano | A newer version wont fix something thats not been reported |
| 06:14.30 | drmessano | If indeed its even a bug |
| 06:15.13 | p3nguin | If it's a bug, someone surely would have reported it. |
| 06:15.31 | drmessano | Thats presumptious |
| 06:15.36 | p3nguin | of course |
| 06:16.17 | p3nguin | I presume that most "bugs" are really just incompatibility of either hardware or user with the software. |
| 06:16.23 | drmessano | So is it being a bug, or even a problem with asterisk itself and not this app_pokemon |
| 06:16.43 | drmessano | Sorry, chan_pokemon |
| 06:17.16 | kam187 | i _think_ its a bug in this channel driver actually |
| 06:17.20 | drmessano | ..... |
| 06:17.26 | kam187 | it seems to assume there's data to send if there's data to read |
| 06:17.40 | kam187 | which would be true most of the time if there was comfort noise i guess |
| 06:17.44 | drmessano | So .... where does a newer version of asterisk come into play? |
| 06:18.04 | *** join/#asterisk Tim_Toady (n=moi@79.103.5.48) |
| 06:18.07 | kam187 | i just wanted to check if this was a stable release |
| 06:18.28 | drmessano | Doesnt matter.. it can be unstable as hell, and you still have a buggy channel driver by your own admission |
| 06:18.52 | drmessano | You could download 1.0.7 and this thing would still be buggy :) |
| 06:18.59 | kam187 | yeah of course :) just if it wasnt a stable release i'd dump this version and go to a stable one, and _then_ fix my channel driver |
| 06:19.06 | kam187 | lol i remember 1.0.7 |
| 06:19.40 | kam187 | its odd the guy who made this thing says it works on 1.6.1 |
| 06:20.02 | p3nguin | I would ask for a demo. |
| 06:20.03 | drmessano | and BTW, there is no such thing as a stable software release |
| 06:20.43 | kam187 | yeah stable as in !trunk, and used by alot of people |
| 06:20.47 | drmessano | There is current, there is beta, alpha, RC, but stable is based on fuzzy logic and doesn't actually exist in a tangible form |
| 06:20.58 | kam187 | it just means that if there were bugs its more likely they would be found or reported |
| 06:21.03 | *** join/#asterisk ticoit (n=ticoit@201.191.190.231) |
| 06:21.06 | drmessano | No true |
| 06:21.20 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 06:21.29 | drmessano | 1.4.19 was "stable" until the minute they found IAX regression out the yin yang |
| 06:21.44 | kam187 | you're not understanding what i'm trying to say! |
| 06:22.00 | kam187 | if i use 1.6.1.0.4.5 trunk, and only 3 other people in the world were using it |
| 06:22.16 | kam187 | when/if it doesnt work, there isnt much to google for is there |
| 06:22.17 | drmessano | First off, trunk doesn't have a version number |
| 06:22.36 | kam187 | ok fair enough |
| 06:22.39 | drmessano | Releases have version numbers |
| 06:22.57 | kam187 | ok |
| 06:23.08 | drmessano | So you could be asking for a release.. but again, this requirement of a stability tag is some unicorn fairytale |
| 06:23.24 | kam187 | yeah ok |
| 06:29.43 | *** join/#asterisk kevm (n=kmaharaj@dsl-243-207-229.telkomadsl.co.za) |
| 06:32.46 | *** join/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 06:33.12 | *** part/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 06:44.42 | cmendes0101 | p3nguin: Here is my extensions.conf http://pastebin.com/m16c1b610 |
| 06:44.49 | cmendes0101 | The contexts are not loading |
| 06:46.25 | p3nguin | What is this, FreePBX? |
| 06:47.00 | cmendes0101 | no just the samples from asterisk. I just added mine at the bottom |
| 06:47.17 | p3nguin | Did you "dialplan reload" after saving the file? |
| 06:49.18 | kevm | when I try to upgrade asterisk I get this err: (what is wrong?) |
| 06:49.20 | kevm | Downloading Packages: |
| 06:49.22 | kevm | Running rpm_check_debug |
| 06:49.22 | kevm | ERROR with rpm_check_debug vs depsolve: |
| 06:49.22 | kevm | zaptel is needed by (installed) elastix-additionals-1.1-2.noarch |
| 06:49.22 | kevm | Complete! |
| 06:49.22 | kevm | (1, [u'Please report this error in http://yum.baseurl.org/report']) |
| 06:50.37 | cmendes0101 | p3nguin: "dialplan reload" gives no such command. I've been just doing a "reload" |
| 06:50.49 | cmendes0101 | but i did a reload after editing the extensions.cofn |
| 06:50.53 | cmendes0101 | conf* |
| 06:50.53 | p3nguin | Are you running asterisk on Windows? |
| 06:50.58 | cmendes0101 | no linux |
| 06:51.07 | p3nguin | How can there be no "dialplan reload"? |
| 06:51.21 | cmendes0101 | i dont know.. the "help" also does not display that command |
| 06:51.36 | p3nguin | What about "core show version"? |
| 06:51.50 | cmendes0101 | Asterisk 1.6.1.9 built by root @ s1 on a x86_64 running Linux on 2009-11-10 07:18:45 UTC |
| 06:52.47 | p3nguin | You're not the first person to say that certain vital commands are not available. It's boggling to me. |
| 06:53.44 | p3nguin | I don't even know what "reload" actually reloads, so maybe a full restart would do some good. |
| 06:53.56 | cmendes0101 | I've never had an issue from 1.2. I havent rebuild a system in like 3 years so not sure if there is a new setting I need to switch to get extensions working |
| 06:54.00 | cmendes0101 | I'll try the restart |
| 06:54.04 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 06:54.05 | p3nguin | "The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead." |
| 06:54.14 | kevm | ? |
| 06:54.41 | cmendes0101 | nothing, let me do a restart |
| 06:55.13 | p3nguin | Well, you don't need to reload any specific modules, anyway, so that would be useless to you. |
| 06:57.39 | ChannelZ | module reload pbx_config |
| 06:57.54 | p3nguin | ah |
| 06:58.53 | cmendes0101 | No such module 'pbx_config'? |
| 06:59.18 | ChannelZ | YOU SYSTEM IS SCREW! |
| 06:59.58 | ChannelZ | or maybe it changed names in 1.6 |
| 07:00.04 | ChannelZ | But I'm leaning towards screw |
| 07:00.33 | cmendes0101 | hmm.. ok.. in 1.6 how do i do a stop gracefully? |
| 07:00.57 | cmendes0101 | used to work in 1.2 |
| 07:01.08 | cmendes0101 | oh nvm |
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| 07:03.28 | cmendes0101 | awsome.. yah the reboot fixed it. Not sure why nothing was being reloaded and a reboot was needed... |
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| 07:54.21 | DigitalFlux | Guys |
| 07:54.43 | DigitalFlux | i have a Centos Box with asterisk installed on it from rpms |
| 07:54.57 | DigitalFlux | i need to install a 2nd asterisk instance on this same centos installation |
| 07:55.01 | DigitalFlux | any hints on how to do that ? |
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| 08:03.04 | Tim_Toady | compile from source and use different prefix for installation |
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| 08:16.55 | DigitalFlux | OK, i am using Centos 5.2 with Kernel 2.6.18-92.1.22.el5xen |
| 08:17.28 | DigitalFlux | but when i try to run the configure script it shows "./config.guess: unable to guess system type" |
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| 08:43.49 | lost_soul | can you tell it what system type in the make file? or by appending options? |
| 08:45.15 | drmessano | What if you speak softly, use small words? |
| 08:46.42 | lost_soul | or.. lemme go out on a limb... try using the rpm created for use on CentOS systems |
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| 08:49.44 | DigitalFlux | lost_soul: I used --prefix and --exec-prefix, configure finished fine, however i am now facing problems with make |
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| 08:50.57 | lost_soul | DigitalFlux: well, one problem down.. at least your making progress |
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| 08:53.58 | DigitalFlux | lost_soul: Indeed |
| 08:54.06 | DigitalFlux | lost_soul: I have a lot of undefined references ! |
| 08:55.34 | lost_soul | DigitalFlux: well, worse case scenerio you can still always try the package if all else fails |
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| 08:56.18 | DigitalFlux | lost_soul: I already have an Asterisk installation on that Centos Box, what i am trying to do is to install another instance of asterisk on the same machien |
| 08:56.23 | DigitalFlux | machine * |
| 08:56.27 | DigitalFlux | i need that to test something |
| 08:57.27 | lost_soul | can you make jails in centos? |
| 08:57.49 | lost_soul | maybe conflicts are causing your issues if the second install isn't being jailed? |
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| 08:58.03 | DigitalFlux | lost_soul: Hmm.. may be |
| 08:58.18 | DigitalFlux | Let me show you what i am trying to do |
| 08:58.37 | DigitalFlux | i have a Cisco CM that trunks calls to the asterisk box |
| 08:58.57 | DigitalFlux | and i handle IVR and queues, and all the other stuff on my asterisk installation |
| 08:59.07 | DigitalFlux | yesterday i had a very weird behaviour |
| 08:59.17 | DigitalFlux | without changing any of the cisco/asterisk configs |
| 09:00.38 | DigitalFlux | when a call gets transferred to asterisk, it simply hangs up |
| 09:00.46 | DigitalFlux | with the following on the asterisk CLI |
| 09:00.49 | DigitalFlux | http://pastebin.com/m420e863c |
| 09:00.54 | DigitalFlux | lost_soul: Can u please check it |
| 09:01.29 | lost_soul | sec, looking |
| 09:02.47 | lost_soul | line 19 appears to be where the issue is |
| 09:05.25 | lost_soul | I would look in your extensions.conf, try to comment out where you feel the issue lies, put a wait after and see if asterisk responds correctly |
| 09:06.29 | lost_soul | if the call doesn't terminate "weird" you'll have an idea where to start looking, I know you said nothing in extensions.conf changed.. but something is causing this |
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| 09:12.52 | kaldemar | there's no issue in that CLI output. enable sip debug and do another pastebin. something hangs up, but you can't see what from that. |
| 09:13.47 | kaldemar | pasting the dialplan won't hurt either. |
| 09:22.52 | DigitalFlux | lost_soul: kaldemar : I understand, i will enable sip debugging and pastebin it |
| 09:28.07 | DigitalFlux | Here is the debug info with sip debugging enabled |
| 09:28.09 | DigitalFlux | http://pastebin.com/m6f384c64 |
| 09:30.10 | kaldemar | DigitalFlux: nothing there either. show your dialplan, specifically [ivr-13]. |
| 09:31.23 | DigitalFlux | kaldemar: OK, i will pastebin it, just a notice i also tried changing this inbound route to another extension/ivr menu and i still get the same |
| 09:31.54 | DigitalFlux | http://pastebin.com/d219eec89 |
| 09:32.07 | DigitalFlux | kaldemar: This is [ivr-13] |
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| 09:38.06 | DigitalFlux | kaldemar: lost_soul : nothing wrong with that context too, no ? |
| 09:38.52 | DigitalFlux | kaldemar: lost_soul : i tried redirecting this route to a specific extension, it also hangs up |
| 09:39.56 | DigitalFlux | That's why i tried to install a 2nd asterisk instance on the same server, so i can see what's wrong |
| 09:43.47 | kaldemar | interesting. check the dialplan with dialplan show ivr-13 and see if something's still wrong. what version are you using btw? |
| 09:46.06 | DigitalFlux | kaldemar: OK, here is dialplan show ivr-13 http://pastebin.com/d60bf22a3 |
| 09:46.25 | DigitalFlux | Asterisk 1.4.24 |
| 09:50.16 | *** join/#asterisk Nitrus^ (n=anonymou@cpe-24-30-144-218.san.res.rr.com) |
| 09:50.27 | Nitrus^ | i have a sip trunk configure with the peer section with type=peer and no context listed, the user section has type=friend with context=from-trunk. with this configuration i can make calls but any incoming calls get a not in service, if i comment out the type=peer in the outgoing section then i can receive incoming but cannot call out. any ideas? im probably just missing a line or have the context set wrong or something |
| 09:51.06 | DigitalFlux | kaldemar: Could that possibly be a bug or something like that ? |
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| 09:55.07 | kaldemar | DigitalFlux: you could try enabling core debug if it shows something. other than that, i can't think of anything now. |
| 09:56.59 | kaldemar | Nitrus^: when you have the type=peer, the incoming calls most likely match to it. since you don't define a context for it, whatever is defined under [general] is used. if nothing is defined, default context is used. when you comment out the type=peer definition, the incoming calls match something that works. |
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| 09:57.55 | Nitrus^ | thank you |
| 09:58.05 | Nitrus^ | that helped |
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| 10:06.26 | Nitrus^ | now i have a new problem. my call gets dropped after about a minute or so with a BYE from my box to my sip trunk . Hangup cause is normal clearing |
| 10:07.06 | Nitrus^ | oops im being lazy, i didnt google yet |
| 10:13.14 | Chainsaw | Nitrus^: If you're not hanging up yourself, you may want to turn up core verbosity to 10. |
| 10:13.21 | Chainsaw | Nitrus^: To force Asterisk to tell you more of what's going on. |
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| 10:14.01 | Nitrus^ | looks like this is only for outbound calls |
| 10:14.04 | Nitrus^ | weird. |
| 10:14.11 | Nitrus^ | ill check the verbose info |
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| 10:20.02 | Nitrus^ | shows nothing useful |
| 10:20.13 | Nitrus^ | just that it hung up |
| 10:20.20 | Nitrus^ | i took verbosity up to 99 |
| 10:20.23 | Nitrus^ | still nada |
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| 10:25.21 | Nitrus^ | i think it might be because im using a soft phone behind a nat that's a tad too many router hops away using ulaw |
| 10:26.18 | lost_soul | I use a softphone behind a nat, shouldn't cause you issues |
| 10:26.29 | Nitrus^ | what codec? |
| 10:26.36 | lost_soul | so long as the sip.conf is set to nat that extension |
| 10:26.49 | lost_soul | it's set to use either ulaw or alaw |
| 10:27.07 | Nitrus^ | hmm then it's probably another problem |
| 10:28.54 | lost_soul | if it connects the call, and your getting a hangup signal shortly after. I would say it's prolly a dialplan issue |
| 10:29.05 | lost_soul | though I'm no asterisk guru by far |
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| 10:55.26 | guyvdb_ | Hi, I am setting up asterisk for the first time. I have SIPs configured and working. I have a TDM800 card with 8 FXO's. It seems to be working. When I do a dmesg | grep FXO I see all 8 channels. At the CLI I do dahdi show channels and none are listed. What am i missing? |
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| 11:09.21 | drepan | which is the better softphone/SIP client to use with Asterisk? |
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| 11:15.11 | sajam | Hello, i want to record a message in audio format on windows and put on Asterisk, but can't find a software that convert from wma to gsm |
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| 11:15.43 | sajam | So what does Asterisk support extension from windows other than gsm????? |
| 11:17.13 | Chainsaw | sajam: There's an easier way to do this. |
| 11:17.31 | sajam | Chainsaw, great what is it? |
| 11:17.34 | Chainsaw | sajam: Install a soft phone client on Windows, call your voicemail box and make your recording. Then just steal that recording from the voicemail directory. |
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| 11:21.07 | sajam | Chainsaw, i didn't make the mailbox yet, so i don't have mailboxed. |
| 11:21.07 | sajam | is there any other solution |
| 11:21.36 | Chainsaw | sajam: Yes, setting up a mailbox for this. Finding Windows software for a task tends to mean additional expense. |
| 11:22.00 | Chainsaw | sajam: Not to mention, we're unlikely to be familiar with it here. |
| 11:24.21 | sajam | Chainsaw, thank you for your support, could you please give a reference or advice how to install a mailbox? |
| 11:24.47 | Chainsaw | sajam: What documentation are you following at the moment? |
| 11:25.27 | sajam | Chainsaw, Asterisk the future of telephony. |
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| 11:26.35 | Chainsaw | sajam: And you're sure it's not described in there? |
| 11:27.12 | kaldemar | how about just using app Record? |
| 11:28.20 | sajam | Chainsaw, thanks i will make some search about that. |
| 11:29.22 | kaldemar | exten => 12345,1,Record(/path/to/file.wav) |
| 11:30.00 | kaldemar | then just dial 12345, make the recording and hang up. voila. |
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| 11:40.14 | drepan | Which is the best softphone to use in windows, X-lite? |
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| 12:15.33 | TommyBotten | Does the current (1.6.1) asterisk support DNS srv records? |
| 12:17.25 | TommyBotten | I need a high availability solution for my outbound trunk, and would like to know if I have to do it by dialplan or not |
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| 13:00.17 | Ccomp5950 | holy netsplit batman |
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| 13:00.52 | TommyBotten | mhm |
| 13:01.13 | TommyBotten | Does the current (1.6.1) asterisk support DNS srv records? |
| 13:01.16 | TommyBotten | I need a high availability solution for my outbound trunk, and would like to know if I have to do it by dialplan or not |
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| 13:09.15 | Delido1983 | is there any way to set the context while when "Got SIP response 302 "Moved Temporarily" |
| 13:09.30 | siera04 | anybody tells me whether an analog phone plug in FXS port(in Zaptel card) or not??? |
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| 13:14.41 | lenne_dk | I need an idea for doing a test and then going back in the dialplan, parsing the dialled number. |
| 13:16.24 | lenne_dk | if business-hours then allow calls, else deny |
| 13:17.05 | TommyBotten | lenne_dk: exten => 234873,1,GotoIfTime(9:00-17:00|mon-fri|*|*?frontdesk,s,1) |
| 13:17.18 | guyvdb_ | I am setting up asterisk for the first time. In my dial plan I have exten => 9,1,Dial(DAHDI/g1/5551212) I get Everyone is busy/congested at this time (1:0/0/1) when I try dial 9. Can someone give me some pointers? I can dial between SIPs and receive PSTN calls on the FXO but cannot dial out |
| 13:17.50 | TommyBotten | guyvdb_: Are you sure you are in the right context? |
| 13:18.31 | guyvdb_ | that line is in a context called from-sip and the sip phones are defined in that context. Is this right? |
| 13:18.32 | lenne_dk | TommyBotten, Then I have to replicate the entire dialplan, and adding the tests for every rule, for this phone which should be registered. |
| 13:18.38 | guyvdb_ | i.e. in sip.conf |
| 13:19.32 | TommyBotten | lenne_dk: You really don't. You just add that to the incoming context, and if it is within the business hours, send to the context where the phones reside. Else play some message |
| 13:19.57 | TommyBotten | guyvdb_: Cool. But your dahdi also resides in a context. Which context is that? |
| 13:20.07 | lenne_dk | You misunderstand. Its the outgoing, I want to limit. |
| 13:20.18 | TommyBotten | lenne_dk: Ahh.. So I did. |
| 13:20.55 | TommyBotten | lenne_dk: In that case, do it the other way around. Use it as a default context for all users |
| 13:21.09 | lenne_dk | Actually, somebody keeps stealing my portable, so I want to lock it :-) |
| 13:21.10 | guyvdb_ | dahdi is in another context... from-ptsn |
| 13:21.10 | TommyBotten | lenne_dk: And then pass to the outgoing context instead |
| 13:21.12 | siera04 | anybody tell me how to check whether an analog phone plug in FXS port(in Zaptel card) or not??? |
| 13:21.20 | [TK]D-Fender | [08:19]<TommyBotten>guyvdb_: Cool. But your dahdi also resides in a context. Which context is that? <- no |
| 13:21.38 | [TK]D-Fender | guyvdb_: pastebin your chan_dahdi.conf |
| 13:21.40 | [TK]D-Fender | ~pb |
| 13:21.41 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:21.42 | [TK]D-Fender | ^^^^^^^66 |
| 13:21.44 | guyvdb_ | ok |
| 13:22.05 | TommyBotten | siera04: There is no autodetection of analog devices. |
| 13:22.47 | guyvdb_ | let me take the comments out of chan_dahdi it is huge... one min |
| 13:22.50 | [TK]D-Fender | DAHDI isn't IN a context. Calls FROM DAHDI channels land in that context, but it has no impact on outbound which is what he's asking about |
| 13:23.03 | [TK]D-Fender | guyvdb_: Remove those permanenently |
| 13:23.23 | guyvdb_ | i will |
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| 13:25.22 | lenne_dk | TommyBotten, I have a special context for that phone, with a .X_ to match everything it dialls, to a test if calls are allowed. If allowed, I want to take the dialled number, and continue into the normal context, so the local, long-distance etc are matched. |
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| 13:27.40 | guyvdb_ | [TK]D-Fender: my chan_dahdi.conf http://www.pastie.org/733531 |
| 13:28.00 | [TK]D-Fender | guyvdb_: #include dahdi-channels.conf <------------ and this obviously |
| 13:28.17 | [TK]D-Fender | guyvdb_: and please include the complete call attempt's CLI output |
| 13:28.18 | guyvdb_ | getting it |
| 13:28.24 | lenne_dk | I have run into the same problem with enum; I do a enum-lookup, if it fails, I want to continue with parsing the normal way, but the digits "have already been used", so I can't |
| 13:28.31 | guyvdb_ | http://www.pastie.org/733533 |
| 13:28.52 | [TK]D-Fender | lenne_dk: digits "have already been used" <--- huh?! |
| 13:29.54 | [TK]D-Fender | [08:17]<guyvdb_>I am setting up asterisk for the first time. In my dial plan I have exten => 9,1,Dial(DAHDI/g1/5551212) I get Everyone is busy/congested at this time (1:0/0/1) when I try dial 9. Can someone give me some pointers? I can dial between SIPs and receive PSTN calls on the FXO but cannot dial out <--- your channels are all in GROUP=0, and you are trying to dial GROUP=1. This is... |
| 13:29.55 | [TK]D-Fender | ...why it fails |
| 13:30.26 | guyvdb_ | I tried group 0 first |
| 13:30.34 | guyvdb_ | let me try again on g0 |
| 13:30.40 | lenne_dk | If 001234567890 matches an entry in the dialplan, I do a enum_lookup. But if it fails, I cant match it against 00123X_ because it have already been matched |
| 13:30.55 | guyvdb_ | here is CLI output http://www.pastie.org/733536 |
| 13:31.15 | [TK]D-Fender | guyvdb_: You are still diaing GROUP=1 there <- |
| 13:31.31 | guyvdb_ | yep... changing it now |
| 13:31.35 | [TK]D-Fender | guyvdb_: Your channels are all group=0 <- |
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| 13:31.52 | guyvdb_ | http://www.pastie.org/733539 here is output from g0 |
| 13:31.54 | lenne_dk | I want to jump to another context, then start matching again there |
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| 13:32.34 | [TK]D-Fender | lenne_dk: so do it |
| 13:33.07 | [TK]D-Fender | guyvdb_: PB "dhadi show channels" , "dahdi show channel 1", "dahdi show status" |
| 13:34.19 | lenne_dk | How? How do I get to the [outgoing] context? |
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| 13:35.16 | Gido-E | lenne_dk with a goto? |
| 13:35.54 | guyvdb_ | show channels/channel 1/status http://www.pastie.org/733549 |
| 13:35.55 | [TK]D-Fender | lenne_dk: Goto <- |
| 13:36.16 | lenne_dk | goto (outgoing,s,1) But it doesn't take the dialled number and parse it. |
| 13:36.33 | guyvdb_ | wait... i have 8 channels but only 2 have pstn lines in them... could that be what is going on? |
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| 13:37.08 | [TK]D-Fender | guyvdb_: Quite possibly |
| 13:37.20 | [TK]D-Fender | guyvdb_: target specific DAHDI channels for your test |
| 13:37.29 | [TK]D-Fender | guyvdb_: * will not try to detect which are plugged |
| 13:37.45 | [TK]D-Fender | lenne_dk: Because YOU told it "s" <- |
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| 13:37.59 | guyvdb_ | i know 1 and 2 are active how do i do that Dial(DAHAI/0-1/5551212) ?? is that right |
| 13:38.07 | [TK]D-Fender | lenne_dk: Maybe you should tell it something SMARTER.... like I dunno... the NUMBER YOU DIALED |
| 13:38.19 | [TK]D-Fender | guyvdb_: DAHDI/1/12345 |
| 13:38.20 | lenne_dk | I've tried goto(outgoing,<dialled number>,1) but it doesn't do matching. goto(outgoing,00123456,1) doesn't go to 001X - extension |
| 13:38.32 | [TK]D-Fender | lenne_dk: show us |
| 13:38.32 | guyvdb_ | thx |
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| 13:40.00 | lenne_dk | Also if [outgoing] includes other contexts, a goto(outgoing,(something in an included context),1) doesn't work |
| 13:40.27 | [TK]D-Fender | lenne_dk: Less talk, more pastebin |
| 13:40.42 | lenne_dk | [leifoutgoing] |
| 13:40.42 | lenne_dk | exten => _X.,1,noop(Dette er leifs) |
| 13:40.42 | lenne_dk | exten => _X.,n,noop(${CALLERID(dnid)}) |
| 13:40.42 | lenne_dk | exten => _X.,m,goto(outgoing,${CALLERID(dnid)},1) |
| 13:42.07 | lenne_dk | a goto goto's a specific exten, it doesn't allow parsing for a match |
| 13:42.44 | guyvdb_ | exten => 9,1,Dial(DAHDI/1/0846780045) and exten => 9,1,Dial(DAHDI/1/ww0846780045) but still I get busy/congested when dialing out |
| 13:43.56 | kaldemar | lenne_dk: pastebin a CLI output of a call |
| 13:44.31 | kaldemar | lenne_dk: and take a long look at this: exten => _X.,m,goto |
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| 13:46.05 | TommyBotten | 2:D |
| 13:46.57 | lenne_dk | I dial 0012345, I goto(outgoing,0012345,1), but there is no exten 0012345 in outgoing, because it should match _XXXXX for normal calls. |
| 13:47.56 | kaldemar | lenne_dk: no it shouldn't. |
| 13:49.25 | lenne_dk | EXAXTLY! I want to enter [outgoing], like it was a virgin call, which hasn't been matched by another rule. |
| 13:49.26 | [TK]D-Fender | lenne_dk: pastebin <- |
| 13:49.33 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 13:50.47 | lenne_dk | paste what? exten => _X.,m,goto(outgoing,${CALLERID(dnid)},1) results in goto(outgoing,0012345,1) which doesn't exist. |
| 13:51.05 | kaldemar | lenne_dk: your dialplan and a failed call attempt |
| 13:51.23 | kaldemar | lenne_dk: again you're pasting syntactically false stuff |
| 13:51.24 | TommyBotten | is 'm' really valid? |
| 13:51.29 | [TK]D-Fender | TommyBotten: NO. |
| 13:51.29 | kaldemar | TommyBotten: no |
| 13:51.40 | TommyBotten | Exactly... Was just wondering why noone said anything :> |
| 13:51.42 | *** join/#asterisk mintos (n=mvaliyav@117.206.18.96) |
| 13:51.50 | lenne_dk | Oh, n. |
| 13:51.54 | [TK]D-Fender | TommyBotten: Which is why I want an F-ing pastebin. because I don't trust him at face value. |
| 13:51.59 | kaldemar | TommyBotten: i told him to take a look at it but maybe it didn't stick. |
| 13:52.04 | lenne_dk | Hang on a sec |
| 13:52.06 | TommyBotten | Hehe.. Aha :) |
| 13:52.18 | TommyBotten | "You know what they say. M is the new N" |
| 13:52.19 | [TK]D-Fender | kaldemar: I've asked about half a dozen times, don't see why you should get any better results :p |
| 13:53.04 | kaldemar | [TK]D-Fender: i see you have. maybe he just needs repetition. or considers you untrustworthy. :) |
| 13:53.35 | TommyBotten | But he listens to me. Must be my overly loud voice and big ears. |
| 13:53.38 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 13:54.04 | lenne_dk | Egg-face time :-! |
| 13:54.14 | [TK]D-Fender | TommyBotten: You're jsut a little lost yourself.... so this is the blind leading the blank ;) |
| 13:54.25 | TommyBotten | :D |
| 13:54.34 | lenne_dk | it works |
| 13:54.38 | [TK]D-Fender | lenne_dk: Move along... these aren't the droids you're looking for. |
| 13:54.40 | guyvdb_ | [TK]D-Fender I have tried to call a DAHDI channel directly and also modified chan_dahdi.conf to only define one channel. I can call into the channel but still cannot call out. Any ideas? http://www.pastie.org/733574 |
| 13:54.50 | [TK]D-Fender | lenne_dk: OMG, its a MIRACLE |
| 13:55.13 | lenne_dk | walks flushing out. |
| 13:55.33 | lenne_dk | means face, not toilet... |
| 13:55.52 | TommyBotten | We're all so nice here :D |
| 13:56.01 | TommyBotten | *Holding hands, singing combaya* |
| 13:56.02 | [TK]D-Fender | guyvdb_: I see a group dial in there again, is hould be seeing at least your last single-channel dial format instead |
| 13:56.19 | *** part/#asterisk lenne_dk (n=Leif@cpe.atm2-0-74391.hknxx4.customer.tele.dk) |
| 13:56.39 | guyvdb_ | I tried both Dial(DAHDI/1/xxxx) and DAHDI/g0/xxxx |
| 13:57.24 | guyvdb_ | Is the index 1 based? Ie channel 1 is DAHDI/1 and not DAHDI/0 |
| 13:59.56 | [TK]D-Fender | guyvdb_: Correct |
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| 14:17.56 | Katty | :< |
| 14:18.01 | Katty | rainy, wet, cold. |
| 14:18.03 | Katty | :<<< |
| 14:18.06 | Katty | who authorized this weather?! |
| 14:18.21 | dlynes | Katty: the national weather service |
| 14:18.26 | *** join/#asterisk smooth_penguin (n=smoove@59.95.24.25) |
| 14:18.29 | Katty | cries |
| 14:18.36 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 14:18.57 | *** join/#asterisk Petaris (n=Petaris@r3000.frederic.k12.wi.us) |
| 14:19.01 | Petaris | Hello all |
| 14:19.03 | *** join/#asterisk De_Mon (i=de_mon@fl-71-0-150-223.dhcp.embarqhsd.net) |
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| 14:20.26 | Petaris | I just installed AsteriskNow and I can't seem to find any documentation on what the defaul admin username/password is? I am following the "Asterisk: The Future of Telephony" AsteriskNow section |
| 14:20.32 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:20.58 | *** part/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 14:21.27 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:21.27 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 14:21.29 | Petaris | there was a note about the live cd using admin and password but that doesn't seem to work |
| 14:21.40 | Petaris | neither does root and my root pw |
| 14:21.53 | Petaris | or admin and my root pw |
| 14:22.06 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:22.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:22.08 | Petaris | Does anyone happen to know where this might be documented? |
| 14:23.06 | Katty | Petaris: i don't work with asterisknow, i work with Asterisk |
| 14:23.59 | Petaris | ahh, I think I might have found it |
| 14:24.02 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 14:24.20 | [TK]D-Fender | Petaris: the book details the OLD AsteriskNOW which was a completely different build |
| 14:24.23 | dlynes | Petaris: you might try asking on #asterisknow almost nobody in this channel uses it |
| 14:24.45 | [TK]D-Fender | Petaris: Old = rPath + AsteriskGUI. new (year old+) = centOS + FreePBX |
| 14:25.20 | Petaris | ahh |
| 14:25.28 | Petaris | ok, thanks guys/gals |
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| 14:25.41 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:25.54 | Petaris | I wasn't aware Asterisk and AsteriskNow were so different |
| 14:27.04 | Petaris | I am looking at it as a posiblility to replace two Nortel Maridian phone systems, does any one have a recomendation of which product would be the most appropriate for this? |
| 14:27.37 | ManxPower-work | Asterisk isn't really a PBX. Asterisk is a TOOLKIT that helps you build a PBX. |
| 14:27.59 | Katty | although i'm sure someone would love to provide you a quote |
| 14:29.04 | TommyBotten | Does asterisk support the dns SRV record? |
| 14:29.07 | Petaris | ahh |
| 14:30.21 | Petaris | What is the reason that all of you have chosen Asterisk instead of AsteriskNow? Was it because of customizations or is there another reason? |
| 14:30.56 | Chainsaw | Petaris: Because I've already provided an OS. |
| 14:31.02 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
| 14:31.05 | Chainsaw | Petaris: I just want a telephony server. |
| 14:31.45 | [TK]D-Fender | Petaris: AsteriskNOW is an entire distro and includes a GUI that confines what you can do. |
| 14:31.45 | Katty | Petaris: i know exactly what i want. i don't need something to do it for me. and possibly snickerdoodle it up |
| 14:31.46 | Petaris | Chainsaw: Ok. |
| 14:32.15 | Katty | eppigy: :> |
| 14:32.23 | eppigy | Katty: :> |
| 14:32.34 | eppigy | mcdonalds breakfast: :> |
| 14:32.35 | Katty | it's so good to see you |
| 14:32.38 | Katty | WHAT?! |
| 14:32.40 | Katty | :<<< |
| 14:32.42 | *** join/#asterisk zoap (n=zoap@53501AB4.static.ziggozakelijk.nl) |
| 14:32.42 | eppigy | likewise |
| 14:32.45 | ManxPower-work | TommyBotten: yes, controlled by srvlookup in sip.conf |
| 14:32.49 | eppigy | oh no |
| 14:32.55 | eppigy | I have spoiled the mood already :< |
| 14:33.02 | cusco | hi |
| 14:33.13 | Petaris | [TK]D-Fender, Katty: Ok. Thanks for the info. |
| 14:33.13 | Katty | you shouldn't be stuffing icky things down yourself! |
| 14:33.30 | TommyBotten | ManxPower-work: I've seen that one. But the (very outdated) documentation I've read, states that it only cares about the first entry hence doesn't really comply with the RFC. |
| 14:33.44 | Katty | eppigy: i haven't had breakfast yet. |
| 14:33.44 | ManxPower-work | TommyBotten: as I understand it that is correct. |
| 14:33.54 | Katty | eppigy: left home in a rush :< |
| 14:33.55 | eppigy | Katty: let's go get something |
| 14:34.01 | zoap | hello, i have some trouble starting asterisk, can i just fire away? |
| 14:34.03 | eppigy | I will throw out the deluxe breakfast |
| 14:34.20 | Katty | deluxe grease. |
| 14:34.34 | eppigy | :[ |
| 14:34.40 | TommyBotten | ManxPower-work: Ok. Thanks for the info. |
| 14:34.41 | TommyBotten | zoap: Please do |
| 14:34.43 | Katty | what did you get at mcd? |
| 14:34.55 | TommyBotten | zoap: Though use pastebin for log/CLI output and config |
| 14:35.07 | zoap | it shows me: "asterisk: symbol lookup error: /usr/lib/asterisk/modules/codec_gsm.so: undefined symbol: k6maxmin" when i try to start it |
| 14:35.45 | TommyBotten | zoap: That seems like the gsm module is missing some library to load properly. |
| 14:35.49 | TommyBotten | zoap: How did you install asterisk |
| 14:35.51 | TommyBotten | ? |
| 14:35.55 | zoap | TommyBotten: will do so if you require long info |
| 14:36.09 | zoap | compiled it from source |
| 14:36.43 | TommyBotten | zoap: On the same system as you are running ut? |
| 14:37.21 | zoap | yes |
| 14:37.51 | zoap | did a make clean, configure, make, make install on the system where it runs. well, should run... |
| 14:38.53 | TommyBotten | zoap: Well, the error is related to K6/MMX optimizations for the GSM codec. But can you check if you have libgsm and libgsm-dev installed? |
| 14:40.07 | *** join/#asterisk SuPrSLuG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:40.26 | zoap | libgsm-devel not there... will install it an recompile, see if that helps. tnx |
| 14:40.40 | TommyBotten | No problem |
| 14:41.43 | ManxPower-work | zoap: chances are you'll have to delete the asterisk source and unpack from the tarball again |
| 14:41.55 | Katty | eppigy: what did you get at mcd |
| 14:42.13 | TommyBotten | zoap: Or 'make distclean' |
| 14:42.22 | ManxPower-work | TommyBotten: did not seem to work for me. |
| 14:42.28 | ManxPower-work | never has worked for me. |
| 14:42.42 | zoap | ManxPower-work: ah, so make clean is not engough? will do so then |
| 14:43.00 | ManxPower-work | zoap: try TommyBotten's suggestion first, but don't be suprized if it doesn't wokr. |
| 14:43.33 | *** part/#asterisk Petaris (n=Petaris@r3000.frederic.k12.wi.us) |
| 14:43.35 | zoap | ManxPower-work: will do so |
| 14:43.45 | TommyBotten | ManxPower-work: Ah.. To bad. It "should" ;) |
| 14:44.18 | zoap | and it does! |
| 14:44.25 | ManxPower-work | TommyBotten: yes, it should. |
| 14:44.25 | zoap | running now :) tnx alot! |
| 14:44.33 | xheliox | Using SVN 1.6.2-r233615, I'm getting this error: 'file.c: ignoring unknown format 'wav49'' and also format 'wav'. |
| 14:44.38 | xheliox | Anyone experiencing this? |
| 14:44.51 | ManxPower-work | xheliox: sounds like a package instal |
| 14:45.00 | xheliox | ManxPower-work: It's not. |
| 14:45.16 | xheliox | And it was working up until I updated a couple of days ago. |
| 14:45.38 | TommyBotten | zoap: You're welcome |
| 14:46.48 | xheliox | Both format_wav.so and format_wav_gsm.so are loaded. |
| 14:47.25 | xheliox | Seems to have happened as of the file.c commit on December 1st. |
| 14:47.38 | ManxPower-work | xheliox: we don't fix broken packages |
| 14:48.14 | xheliox | Not a package. |
| 14:48.18 | ManxPower-work | If you compiled from SVN then report an issue on issues.digium.com or ask on #asterisk-dev. |
| 14:49.14 | ManxPower-work | zoap: using asterisk 1.4 or 1.6 |
| 14:49.20 | xheliox | Ok, I'll mention it on #asterisk-dev.. I'm merely trying to be helpful, it's no big deal to revert a few revisions -- and it's on a test box. |
| 14:49.42 | zoap | ManxPower-work: 1.4 |
| 14:50.13 | ManxPower-work | xheliox: I suspect not-a-large-number of people use SVN on this channel |
| 14:50.34 | ManxPower-work | Most of us seem to want stability rather than running development boxes. |
| 14:50.36 | xheliox | And I normally don't. Just something I noticed when playing around. |
| 14:52.59 | Katty | http://farm3.static.flickr.com/2524/4169385532_2680d7d594_b.jpg |
| 14:55.06 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
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| 14:59.48 | smooth_penguin | heh |
| 15:03.10 | eppigy | Katty: deluxe breakfast |
| 15:03.23 | Katty | eppigy: idon't know what that is |
| 15:04.52 | *** join/#asterisk jpcansa (n=jpbenavi@200.91.99.148) |
| 15:05.07 | Katty | googles |
| 15:08.29 | Katty | http://farm4.static.flickr.com/3075/2821279139_3b872a2bba_b.jpg <- eppigy: that? |
| 15:08.48 | ManxPower-work | Katty: let us know when you post something *interesting* |
| 15:08.48 | ManxPower-work | runs |
| 15:09.05 | Katty | ManxPower-work: k |
| 15:14.06 | eppigy | Katty: yesh, plus hash brown |
| 15:14.28 | jaytee | I like the Big Breakfast but I don't care for their pancakes so I never go with the Deluxe Breakfast |
| 15:15.53 | Katty | manx was mean :< |
| 15:16.49 | eppigy | comforts Katty |
| 15:16.54 | eppigy | there there |
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| 15:30.23 | eppigy | cold and rainy :C |
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| 15:32.14 | guyvdb_ | is there a way to dial from the CLI for debug purposes? |
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| 15:33.44 | DocAwesome | guyvdb_: 'dial' or 'originate' |
| 15:34.24 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:35.00 | guyvdb_ | thx |
| 15:35.05 | Kobaz | DocAwesome: there is no 'dial' in the console interface |
| 15:35.28 | DocAwesome | Kobaz: I thought there was if you had the appropriate drivers, but maybe I'm thinking something else -- originate definitely is though |
| 15:35.31 | Kobaz | just originate, or 'core originate' for 1.6.2 |
| 15:35.35 | guyvdb_ | hm I also do not see originate |
| 15:35.46 | DocAwesome | guyvdb_: then you're missing chan_oss or chan_alsa likely |
| 15:35.52 | Kobaz | you might need core for 1.6.1 |
| 15:35.54 | bcrisp | good god we had a heck of a storm last night |
| 15:35.55 | Kobaz | DocAwesome: nope |
| 15:36.14 | Kobaz | DocAwesome: originate is channel driver/codec independent |
| 15:36.34 | DocAwesome | ok, then obviously I haven't used that command for a very long time |
| 15:36.35 | Kobaz | guyvdb_: what version? |
| 15:36.36 | guyvdb_ | I am having problems placing an outgoing call on PSTN. I can do sip to sip and PST to sip but not Sip to pstn. I am pretty new to asterisk |
| 15:36.39 | *** join/#asterisk DMeloUK (n=DominicM@127.159.119.70.cfl.res.rr.com) |
| 15:36.41 | guyvdb_ | trunk |
| 15:36.48 | Kobaz | core originate |
| 15:36.52 | guyvdb_ | trunk from a few days ago |
| 15:37.43 | [TK]D-Fender | guyvdb_'s got un unknown issue. DAHDI reports channels, can't seem to pull it to dial. All ports claimed to have been tested individually, and using a TDM800 |
| 15:37.43 | Kobaz | guyvdb_: define pstn... what hardware are you using |
| 15:37.47 | [TK]D-Fender | Kobaz: See above |
| 15:37.56 | Kobaz | mm |
| 15:37.56 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733516.dsl.bell.ca) |
| 15:38.14 | guyvdb_ | TDM800 8 FXOs |
| 15:38.55 | Kobaz | have you tried a non-bleeding edge release? |
| 15:39.21 | Kobaz | stuff breaks all the time in trunk |
| 15:39.45 | guyvdb_ | I tried 1.6.1.10 but could not get the dahdi to compile until i moved to trunk |
| 15:40.20 | DocAwesome | guyvdb_: that is a known issue, and unfortunately I've seen the same thing -- only able to use DAHDI on newer kernels with trunk |
| 15:40.29 | DocAwesome | assuming you ran into the same issue I did |
| 15:40.34 | guyvdb_ | what is newer kernel? |
| 15:40.41 | Kobaz | guyvdb_: that makes no sense |
| 15:40.49 | Kobaz | guyvdb_: dahdi doesn't need asterisk to build |
| 15:40.52 | guyvdb_ | I am running 2.6.31-14-server 64 bit |
| 15:40.58 | DocAwesome | that'd do it |
| 15:41.07 | DocAwesome | for me it was anything on Ubuntu 9.04 latest |
| 15:41.12 | guyvdb_ | one min i will try build it again |
| 15:41.21 | Kobaz | guyvdb_: you are using dahdi trunk? or asterisk trunk? or both? |
| 15:41.29 | DocAwesome | and Kobaz is right -- asterisk depends on DAHDI for certain modules, but DAHDI doesn't care about asterisk at all |
| 15:41.48 | DocAwesome | changing asterisk versions would have zero effect on DADHI |
| 15:41.52 | DocAwesome | DAHDI* |
| 15:42.00 | guyvdb_ | http://pastie.org/733727 |
| 15:42.09 | Kobaz | DocAwesome: depends what you mean by effect |
| 15:42.19 | DocAwesome | lets not get too pedantic here |
| 15:42.29 | guyvdb_ | the error is struct net_device has no member named do_ioctl |
| 15:42.46 | guyvdb_ | that is on dahdi-linux-complete-2.2.0.2+2.2.0 |
| 15:42.47 | DocAwesome | guyvdb_: yes, use a newer DAHDI |
| 15:42.50 | Kobaz | guyvdb_: anyways... before we go further... tell us what versions of everything you have |
| 15:42.52 | DocAwesome | (i.e. trunk) |
| 15:43.03 | Kobaz | either that or downgrade your kernel |
| 15:43.08 | bcrisp | DocAwesome, will the next release of * require dahdi installed for MeetMe? |
| 15:43.12 | Kobaz | is this for a production or test box? |
| 15:43.26 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:43.31 | DocAwesome | bcrisp: I already pointed you at the answer yesterday on blogs.asterisk.org |
| 15:43.32 | guyvdb_ | I have asterisk trunk rev 233687 |
| 15:43.47 | bcrisp | DocAwesome, sorry to repeat it then, must have missed your post ty |
| 15:44.06 | Kobaz | guyvdb_: do you require 2.6.31 |
| 15:44.24 | guyvdb_ | no.. not realy |
| 15:44.36 | guyvdb_ | It is just what installed |
| 15:44.43 | Kobaz | good, download and build 2.6.27.39 |
| 15:44.51 | DocAwesome | guyvdb_: https://issues.asterisk.org/view.php?id=14285 |
| 15:45.04 | guyvdb_ | ok |
| 15:45.09 | guyvdb_ | I will try that |
| 15:45.32 | DarkRift | guyvdb_, dahdi trunk not asterisk trunk |
| 15:45.42 | Kobaz | guyvdb_: it will make all your problems go away (most likly), and it's a more stable kernel anyway |
| 15:45.55 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 15:45.57 | guyvdb_ | dahdi rev 7666 |
| 15:45.59 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 15:46.19 | Kobaz | 2.6.31 tends to crash when you get high memory contention |
| 15:46.41 | *** part/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 15:46.50 | guyvdb_ | ok got to read up on building/installing a new kernel... :) |
| 15:47.00 | Kobaz | there's something to be said for "old, tried, and true" |
| 15:47.55 | guyvdb_ | Kobaz: dahdi rev 7666 does build however. that is what i currently have installed |
| 15:48.05 | Kobaz | but it doesn't work (tm) |
| 15:48.19 | guyvdb_ | :) |
| 15:48.37 | Kobaz | let the other guys who use the bleeding edge stuff figure it out |
| 15:48.41 | Kobaz | heh |
| 15:49.02 | guyvdb_ | so you don't think it is a config issue but rather an issue with dahdi trunk? |
| 15:49.32 | Kobaz | we can elimiate dahdi problems by using a solid, older release |
| 15:49.54 | Kobaz | and, you're better off not using trunk for dahdi/asterisk anyway |
| 15:50.02 | Kobaz | since your guaranteed to run into problems |
| 15:50.33 | Kobaz | trunk is if you *really do* need the latest, or you're a developer writing new code |
| 15:50.39 | Kobaz | other than that, it's best to stay away from it |
| 15:51.42 | Kobaz | since i'm assuming you want this box to run healthily |
| 15:51.48 | Kobaz | and not randomly crash and lock up |
| 15:51.58 | guyvdb_ | yip it is eventualy for production |
| 15:52.31 | Kobaz | so, once you get a sane asterisk, then we can go and debug your issue |
| 15:53.24 | Kobaz | otherwise, you can head over to #asterisk-dev. put on your debugging and programming hat, and help improve asterisk/dahdi |
| 15:53.25 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 15:53.28 | Kobaz | heh |
| 15:53.31 | fcois93 | hello all |
| 15:53.37 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:53.50 | guyvdb_ | ok... might take me a bit to get the kernel done.. should i be building asterisk 1.6.11 and dahdi 2.2.0 |
| 15:54.18 | fcois93 | is it possible to do a WAITEXTEN with a fixed digits dtmf ? |
| 15:54.21 | Kobaz | i recommend 1.6.0.x or 1.6.1.x |
| 15:54.44 | Kobaz | I haven't uses 1.6.1.x much, but 1.6.0.x has been pretty solid for me |
| 15:54.57 | guyvdb_ | ok |
| 15:55.06 | Kobaz | I'm waiting for 1.6.2 to get released and be stable, because I need some features from it |
| 15:55.35 | Kobaz | but yeah, dahdi 2.2.0 |
| 15:57.15 | [TK]D-Fender | fcois93: Huh? |
| 15:57.43 | Kobaz | fcois93: what are you trying to do |
| 15:58.53 | fcois93 | [TK]D-Fender & Kobaz: I just want that when the user enter 3 digit that the waitexten stop waiting :-) because, the user enter digits and have to wait 3seconds more |
| 15:59.30 | [TK]D-Fender | fcois93: then make it occur in a context with extens that will stop matching at 3 digits |
| 16:01.10 | fcois93 | [TK]D-Fender: then If I have exten =>_X.,n,... it is an infinite possibility then it wait 3seconds ? |
| 16:01.27 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 16:01.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:01.30 | fcois93 | [TK]D-Fender: I have to do _XXX. to accept 3 digits ? |
| 16:01.36 | [TK]D-Fender | fcois93: that says 2 or more... it will make you wait. Come up with smarter patterns if you want it to stop at 3 |
| 16:01.49 | *** join/#asterisk simplydrew_ (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
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| 16:02.08 | jdnwest | Anyone using Vitelity's P2P T1's or their colo space (or anyone toured it?) |
| 16:02.31 | fcois93 | [TK]D-Fender: mistake, it wait for 2 digits if I have _XX ? |
| 16:03.30 | [TK]D-Fender | fcois93>[TK]D-Fender: mistake, it w <- yes |
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| 16:04.10 | fcois93 | [TK]D-Fender: I tested it! thank you |
| 16:04.28 | fcois93 | [TK]D-Fender: it could be fun to have a limit in the WaitExten! |
| 16:07.38 | [TK]D-Fender | fcois93: Waitexten runs off your DIALPLAN. You can have it wait for 3, and its YOUR job to put matches in there |
| 16:08.00 | fcois93 | [TK]D-Fender: you're right |
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| 16:13.06 | guyvdb_ | Kobaz is it possiable to copy my current kernel config to use in building the 2.6.27 kernel? |
| 16:13.40 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:15.35 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 16:17.33 | bcrisp | when using waitexten, i know i can pass in a timeout period, but what I'm looking for is quick recognition when the user types a single digit extension instead of waiting for a possible nnn extension. What options do i have? |
| 16:18.24 | ManxPower-work | bcrisp: Asterisk will NOT wait unless the digit can match a longer sequence. |
| 16:18.43 | bcrisp | ok so moving longer extensions out of the context would remedy it? |
| 16:18.45 | *** join/#asterisk mr_ian_ (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 16:19.26 | ManxPower-work | of course. |
| 16:19.26 | bcrisp | the menu has 1,2,3 as options but there is also some 1nn extensions in the same context |
| 16:19.28 | bcrisp | k |
| 16:19.31 | bcrisp | thanks |
| 16:19.50 | ManxPower-work | well stop overlapping the options. make it 1nn extensions and 2, 3, 4 option. |
| 16:20.01 | ManxPower-work | then you have no overlapping patterns |
| 16:20.13 | bcrisp | makes perfect sense, thank you |
| 16:20.30 | Kobaz | guyvdb_: kinda... you can copy the kernel config, and it won't match every single option, but it will be close |
| 16:20.31 | ManxPower-work | bcrisp: proper dialplan planning/design is one of the most important parts of telecom |
| 16:20.44 | guyvdb_ | ok that is what i did |
| 16:20.49 | bcrisp | ManxPower-work: reading everything i can get my hands on |
| 16:21.19 | Kobaz | guyvdb_: keep your old kernel in your grub config... there's a high liklyhood you'll miss a driver and the new one wont boot |
| 16:21.21 | ManxPower-work | bcrisp: not hard to do. don't overlap patterns and you won't have to rely on timeouts |
| 16:21.23 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 16:21.38 | bcrisp | ManxPower-work: yeah, bit of a brain fart - editing it now thanks |
| 16:22.25 | guyvdb_ | yip :) i am sure that will happen\ |
| 16:25.22 | krondorl | Hmm, Ok, we installed freepbx is there a channel for help with freepbx or is here still ok to ask questions? |
| 16:27.06 | kaldemar | krondorl: #freepbx |
| 16:27.06 | DocAwesome | #freepbx I believe |
| 16:28.33 | krondorl | thanks |
| 16:30.42 | Naikrovek | what are our thoughts on Avaya PoE switches |
| 16:30.46 | Naikrovek | http://cgi.ebay.com/NEW-AVAYA-P333T-PWR-POWER-OVER-ETHERNET-STACKBLE-SWIT_W0QQitemZ300322918360QQcmdZViewItemQQptZCOMP_EN_Hubs?hash=item45eca40fd8 |
| 16:30.59 | Kobaz | netgear makes a nice 24 port full poe switch for like 650 |
| 16:31.22 | Kobaz | $137 isn't bad at all |
| 16:31.31 | Naikrovek | seems rather good to me considering they're new |
| 16:32.12 | DocAwesome | huh |
| 16:32.25 | Kobaz | they must be stolen |
| 16:32.27 | Kobaz | heh |
| 16:32.46 | Naikrovek | yesterday that auction had "more than 10" now there are 4 |
| 16:35.07 | Kobaz | the P330 stack is managed |
| 16:35.08 | Kobaz | as a single entity. |
| 16:35.15 | Kobaz | interesting |
| 16:35.27 | Naikrovek | yes very |
| 16:35.29 | Naikrovek | want them |
| 16:35.54 | Naikrovek | of course i told my supervisor all this yesterday, told him we needed them, and that we should order |
| 16:35.59 | Kobaz | heh |
| 16:36.06 | Naikrovek | dork hasn't even checked his email yet, in a day. |
| 16:37.11 | Naikrovek | they're old but unused |
| 16:37.16 | Naikrovek | manuf date jun 2002 |
| 16:38.03 | coppice | manufactured jun 2002 |
| 16:38.05 | coppice | DHL delivered dec 2009 |
| 16:42.54 | Kobaz | Naikrovek: oh |
| 16:42.56 | Kobaz | Naikrovek: umm |
| 16:43.02 | Naikrovek | what |
| 16:43.05 | Kobaz | Naikrovek: i just talked to my avaya guy in the next room |
| 16:43.17 | Kobaz | those only work with propriatary avaya phones |
| 16:43.26 | Naikrovek | the poe only works with avaya phones? |
| 16:43.29 | Kobaz | yeah |
| 16:43.31 | Naikrovek | lame |
| 16:43.37 | Naikrovek | lame lame lame |
| 16:43.37 | Kobaz | he's like |
| 16:43.39 | Naikrovek | oay |
| 16:43.44 | Naikrovek | okay* |
| 16:43.46 | Kobaz | i have ten of those in my basement... how many do you want? |
| 16:43.51 | Naikrovek | lol |
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| 16:47.39 | guyvdb_ | Kobaz: still building.... grr it is taking a long time |
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