| 00:00.03 | dlynes_office | 97F is about 40C or so |
| 00:00.09 | hads | This is the first settled snow here in 8 years or something |
| 00:00.19 | justinu|laptop | it's hot |
| 00:00.22 | justinu|laptop | and it sucks |
| 00:00.27 | P-NuT | We never get snow here.. :-( |
| 00:00.30 | Nugget | 97F is 36C |
| 00:00.32 | dlynes_office | it's the temperature at which old people start croaking |
| 00:00.36 | Nugget | 40C is ridiculously hot |
| 00:00.42 | P-NuT | no its not. |
| 00:00.51 | P-NuT | 40 is hot, but not unbearable. |
| 00:00.52 | Nugget | it's a lot hotter than 97F. :) |
| 00:00.54 | coppice | 36 is fine if the humidity is low |
| 00:00.57 | hads | justinu|laptop: Lucky you aren't a hippy and have aircon |
| 00:01.02 | P-NuT | try 47 degrees, then complain to me. |
| 00:01.05 | dlynes_office | coppice: you're used to hong kong though :) |
| 00:01.07 | justinu|laptop | 40C is hot when the humidity is 100% |
| 00:01.15 | justinu|laptop | like it is in singapore |
| 00:01.17 | dlynes_office | 40C is hot, period |
| 00:01.17 | Nugget | it doesn't get much over 42C here. |
| 00:01.22 | dlynes_office | i don't care if it's humid or not |
| 00:01.23 | coppice | 30 is pretty hot at 97% humidity |
| 00:01.38 | justinu|laptop | yeah, i was in HK in september |
| 00:01.49 | orlock | goddamn this is annoying me |
| 00:02.10 | dlynes_office | orlock: so go outside and enjoy the girls with the short skirts for a while |
| 00:02.11 | orlock | can somebody give me the output of a sip debug? |
| 00:02.20 | orlock | i just want to see what the Contactstring looks like |
| 00:03.56 | P-NuT | dlynes_office: Why does he need a short skirt to enjoy girls? |
| 00:04.31 | justinu|laptop | the right girl in the right short skirt is pretty enjoyable |
| 00:04.45 | hads | amen |
| 00:05.13 | dlynes_office | P-NuT: you must not get many asian girls in sydney :) |
| 00:05.31 | justinu|laptop | dude |
| 00:05.33 | P-NuT | I think you missed my point. |
| 00:05.34 | P-NuT | LOL |
| 00:05.37 | justinu|laptop | there are loads of asians in sydney |
| 00:05.39 | dlynes_office | P-NuT: a chinese girl in a nice short white skirt, or a nice short plaid skirt is divine :) |
| 00:05.56 | P-NuT | I agree. |
| 00:06.16 | P-NuT | I was trying to make a joke about orlock in a skirt. |
| 00:06.18 | dlynes_office | or some nice tight tennis shorts :) |
| 00:06.24 | P-NuT | it failed dismaly. |
| 00:06.28 | orlock | heh |
| 00:06.35 | justinu|laptop | lol |
| 00:06.42 | justinu|laptop | aussies and kiwis |
| 00:06.44 | dlynes_office | well, your wording is all screwed up :p |
| 00:06.49 | justinu|laptop | always at each other's throats |
| 00:07.10 | orlock | they keep trying to fuck our sheep! |
| 00:07.16 | justinu|laptop | heh |
| 00:07.22 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:07.27 | hads | orlock: Are you a kiwi? |
| 00:07.49 | orlock | hell no |
| 00:08.04 | justinu|laptop | oh i thought one of you was |
| 00:08.11 | *** join/#asterisk notjason (n=notjason@ool-457183bb.dyn.optonline.net) |
| 00:08.17 | hads | I am. I was minding my own! :) |
| 00:08.24 | justinu|laptop | ahh, my bad |
| 00:09.03 | orlock | heheh |
| 00:09.15 | orlock | ANybody? Contact: string in sip registration? |
| 00:09.57 | CunningPike | Later, chaps |
| 00:10.06 | dlynes_office | have a good scoot |
| 00:10.31 | P-NuT | LOL |
| 00:10.35 | P-NuT | kiwis. |
| 00:10.56 | hads | ? |
| 00:11.03 | P-NuT | Aussies wording isn't that screwed. We talk proper like. |
| 00:11.04 | P-NuT | LOL |
| 00:11.29 | justinu|laptop | so sheila is the proper term for a female? |
| 00:11.44 | P-NuT | yes |
| 00:12.02 | P-NuT | we have our own method of safe sex also. |
| 00:12.39 | P-NuT | We don't spray red x's on the sheep that kick though. |
| 00:12.44 | P-NuT | ;-) |
| 00:12.55 | justinu|laptop | haha |
| 00:12.58 | P-NuT | annnnnd. back on topic. |
| 00:13.22 | P-NuT | *sigh* |
| 00:13.30 | P-NuT | anyone want to give me a new job? |
| 00:14.54 | P-NuT | LOL |
| 00:15.17 | *** part/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 00:15.24 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 00:16.33 | *** join/#asterisk adker (n=adker@67-136-213-243.dsl1.glv.ny.frontiernet.net) |
| 00:16.49 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:17.46 | P-NuT | So, on Voxilla, they do have tutorials on the SPA2000 and asterisk. |
| 00:18.07 | P-NuT | Is this the same as the SPA3000? |
| 00:18.14 | justinu|laptop | pretty much |
| 00:18.30 | P-NuT | should I use though to setup the 3000? |
| 00:18.46 | justinu|laptop | all the sipura products work pretty much the same way |
| 00:18.50 | dlynes_office | P-NuT: they have one on voxilla for the 3000 as well |
| 00:19.01 | dlynes_office | P-NuT: to set it up as an analog gateway for asterisk |
| 00:25.50 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 00:27.36 | rene- | what about a room full of agents using blueetooth headsets.. does that work? is interference an issue? |
| 00:31.14 | rene- | a noname bluetooth headset + bluetooth adapter is comparable in price to some usb headsets and handsets, so i wonder if anyone has installed those (bluetooth) at large |
| 00:33.17 | orlock | Hmm.. |
| 00:33.34 | orlock | is anybodys Asterisk setup here sending a username for Contact: tats NOT just "s"? |
| 00:39.13 | P-NuT | dlynes_office: I only see the 2000's articles. |
| 00:39.20 | P-NuT | Nothing on setting the 3000 up. |
| 00:41.03 | *** join/#asterisk bugz (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 00:41.40 | bugz | so fellas, who is gonna write the voip firefox plugin for linux? |
| 00:41.59 | Qwell | bugz: you |
| 00:42.05 | bugz | Qwell: probably |
| 00:42.08 | bugz | how about... |
| 00:42.26 | bugz | voip://18005553333/4 |
| 00:42.30 | bugz | for ext 4 ;) |
| 00:42.32 | bugz | or |
| 00:42.34 | bugz | for option 4 |
| 00:42.36 | bugz | then |
| 00:42.45 | bugz | voip://18005553333/4/3/1 |
| 00:43.04 | bugz | the protocol would be registered to the plugin based on that plugins connection |
| 00:43.31 | bugz | voip://200@192.168.1.13 |
| 00:43.52 | Qwell | lame |
| 00:44.08 | bugz | ok, all this feedback is overwhelming |
| 00:44.18 | bugz | one flame at a time please |
| 00:45.02 | heath__ | are you trying to autonavigate thru menus |
| 00:45.25 | bugz | just an idea |
| 00:45.31 | bugz | nees to be rfc'e |
| 00:45.37 | bugz | rfc |
| 00:46.23 | heath__ | cool, but why not just have your local dialplan do it... senddtmf(xxx) wait(3) senddtmf etc etc |
| 00:46.45 | bugz | firefox plugin |
| 00:47.04 | bugz | activates the user's sound stuff |
| 00:47.21 | bugz | based on a settings dialogue for an existing voip connection |
| 00:47.28 | bugz | iax/h323/sip registry etc |
| 00:47.31 | bugz | yay! |
| 00:48.09 | bugz | thunderbird would be just as easy |
| 00:48.17 | bugz | "Click to Call" |
| 00:48.25 | heath__ | what i mean is... you don't need a protocol to do it... you can do iax://whatever/1234#4#3#1 and tokenize it automatically in the db, unless i'm missing something |
| 00:48.30 | dlynes_office | bugz: someone's already done that |
| 00:48.32 | heath__ | db = dp |
| 00:48.50 | dlynes_office | bugz: it's available as a firefox and thunderbird plugin |
| 00:48.58 | bugz | dlynes_office: for linux? |
| 00:49.09 | dlynes_office | bugz: he's working on it for linux...currently only windows |
| 00:49.12 | heath__ | check moziax and jiaxclient.. both work for linux |
| 00:49.25 | dlynes_office | well, not plugin...extension |
| 00:49.39 | dlynes_office | he's also working on an equivalent for internet exploder |
| 00:49.56 | heath__ | ohhhh click to call... yeah, that's the tel:// protocol |
| 00:50.29 | dlynes_office | When you click on the phone number, it'll call it up through the softphone that you've registered with the extension |
| 00:50.59 | heath__ | yeah, moziax does that... or you could pop a small hacked jiaxclient window and achieve the same thing |
| 00:51.11 | heath__ | i've even tokenized like that too :) |
| 00:53.26 | bugz | dlynes_office: so you need a softphone registered to the plugin? |
| 00:53.34 | dlynes_office | bugz: correct |
| 00:58.08 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 01:00.58 | Idle | hmmm, my zaptel isn't showing up in lspci, so modprobe is failing |
| 01:01.22 | justinu|laptop | it should show up as a tigerjet card |
| 01:01.23 | bugz | Idle: make sure the power is connected to the card.. |
| 01:01.55 | Idle | it is |
| 01:02.51 | Idle | hm, I may move everything up 1 slot |
| 01:06.02 | Idle | should it light with just molex power? |
| 01:06.35 | *** join/#asterisk sharp (n=sharp@c-68-45-160-72.hsd1.pa.comcast.net) |
| 01:06.43 | bugz | typically it lights up when the module is loaded |
| 01:06.49 | Idle | ah, ok |
| 01:06.51 | bugz | but ive seen some cards light up on boot for some reason |
| 01:06.52 | Idle | just checking |
| 01:07.01 | bugz | maybe during pci probe |
| 01:07.02 | sharp | what's the copyright status on the voice prompts that come with asterisk? |
| 01:07.37 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 01:07.48 | bugz | sharp: if you steal them, 'the voice' will haunt your dreams |
| 01:07.52 | bugz | it would be bad karma |
| 01:08.03 | sharp | :) |
| 01:08.28 | sharp | say i want to make a song with them?... |
| 01:09.27 | Idle | its still not showing in lspci |
| 01:10.02 | bugz | what does lspic show? |
| 01:10.04 | bugz | pastebin.com |
| 01:11.22 | bugz | what card is it? |
| 01:14.53 | *** join/#asterisk raulz (n=Beginner@c-67-176-156-43.hsd1.il.comcast.net) |
| 01:17.20 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:17.27 | *** part/#asterisk sharp (n=sharp@c-68-45-160-72.hsd1.pa.comcast.net) |
| 01:17.59 | *** join/#asterisk dant (n=dan@2001:618:400:3f8d:204:76ff:fe1e:585e) |
| 01:19.47 | *** join/#asterisk dan__t (n=dant@72.232.74.146) |
| 01:19.48 | dan__t | hi. |
| 01:20.51 | dan__t | I've not used Asterisk for a very, very long time. I was wondering if there would eventually be a method, ir currently exists such a method, where I can have a PBX dial a phone number, and be able to tell which numbers were pressed on a keypad when that call is hung up? |
| 01:21.06 | dan__t | :s/a PBX/Asterisk |
| 01:21.17 | dan__t | :s/hung up/answered |
| 01:21.59 | h3x | show application Read |
| 01:22.53 | dan__t | word |
| 01:24.10 | dan__t | This may work. |
| 01:25.46 | dan__t | Thanks |
| 01:26.25 | Idle | bugz: sorry |
| 01:26.31 | Idle | its a tdm400 |
| 01:26.38 | Idle | works great on my desktop :D |
| 01:28.25 | Idle | lspci shows my ISA adaptor, IDE adaptor, Display adaptor, and something else... |
| 01:29.58 | Idle | yea, this is fucked |
| 01:30.17 | dan__t | what's wrong with it |
| 01:30.32 | Idle | doesn't show up on the PCI bus at all |
| 01:30.39 | Idle | I'm sure its just a hardware thing |
| 01:31.15 | dan__t | does lspci -vvvv show anything else of interest? |
| 01:31.21 | dan__t | does dmesg bitch about it? |
| 01:31.21 | Idle | no |
| 01:31.24 | Idle | no |
| 01:31.37 | Idle | the bios literally doens't see it |
| 01:31.40 | dan__t | does the card have any jumpers on it or some shit? |
| 01:31.57 | Idle | nope |
| 01:32.05 | dan__t | Got a hammer? |
| 01:32.51 | Idle | Host bridge: Intel Corp. 430FX - 82437FX TSX [Triton 1] rev 02 |
| 01:33.18 | Idle | and the PIIX ISA, and IDE PIIX, and my ATI 3d rage pro |
| 01:33.47 | Idle | I'm gonna yank the ethernet card, see if that helps |
| 01:34.20 | dan__t | werd |
| 01:35.41 | Idle | nothing |
| 01:35.47 | Idle | I bet this thing only has 4 interupts |
| 01:35.59 | dan__t | Am I going to get beat if I ask in here where I get Asterisk SRPMs? |
| 01:37.14 | dan__t | looks like the .tar.gz is rpmbuild-able |
| 01:37.15 | dan__t | nm |
| 01:38.57 | dan__t | Or not. |
| 01:39.19 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 01:39.28 | benjamin7062 | Are humans here? |
| 01:40.11 | benjamin7062 | Please say yes... preferably 'anyone' who can answer some questions |
| 01:40.59 | h3x | no |
| 01:41.15 | benjamin7062 | Bummer |
| 01:41.37 | Bullseye_Network | no humans here. |
| 01:41.51 | Bullseye_Network | The answer to your question is..... Red |
| 01:42.00 | benjamin7062 | I'm desperately trying to convince my company to use Asterisk and replace our Avaya... I've been reading about this for years. We are about to spend 20K to upgrade |
| 01:42.26 | benjamin7062 | I have the check in my hand ... if I can get some answers I can push a week to test... |
| 01:42.34 | benjamin7062 | if I can prove it works.. we switch to this |
| 01:42.45 | hads | it works |
| 01:42.57 | benjamin7062 | Heh, obviously |
| 01:43.03 | Bullseye_Network | I run 4 call centers using asterisk. |
| 01:43.17 | benjamin7062 | I'm 'very' strong in linux... chances of getting this up in 1 week? |
| 01:43.27 | Bullseye_Network | How many people? |
| 01:43.33 | benjamin7062 | For the test... 5 |
| 01:43.37 | benjamin7062 | for the total install... 45 |
| 01:43.40 | benjamin7062 | two sites |
| 01:43.48 | benjamin7062 | 20'ish at each site. |
| 01:43.55 | Bullseye_Network | Piece of cake. |
| 01:43.58 | benjamin7062 | I love you |
| 01:44.02 | dan__t | I'm kinda faced with the same thing here. |
| 01:44.05 | Bullseye_Network | I out.!!! |
| 01:44.08 | dan__t | This will be interesting. |
| 01:44.08 | Bullseye_Network | lol |
| 01:44.19 | dan__t | We have an Intertel Axxcess PBX, which scares me just looking at it. |
| 01:44.23 | benjamin7062 | If I gather enough information.. Owner said I have 1 week to prove it works... |
| 01:44.29 | benjamin7062 | If it works... We start buying gear. |
| 01:44.34 | Bullseye_Network | Where are you? |
| 01:44.38 | benjamin7062 | Austin, TX |
| 01:44.46 | Bullseye_Network | im in Phoenix... |
| 01:44.50 | benjamin7062 | Damn |
| 01:44.58 | dan__t | Where in Phoenix, Bullseye_Network |
| 01:44.58 | dan__t | heh |
| 01:45.08 | Bullseye_Network | Downtown |
| 01:45.15 | dan__t | Word. |
| 01:45.27 | Bullseye_Network | Central and Indian School |
| 01:45.33 | dan__t | Sucker. |
| 01:45.53 | Bullseye_Network | ? |
| 01:46.01 | dan__t | Just F'ing around, sorry. I'm in Chandler. |
| 01:46.02 | benjamin7062 | Okay, I have read read read... And will more... Hopefully you guys can save me time... Which phones do you use with this. There are tons of sips compliant phones. I need 20+ish programmable buttons. |
| 01:46.04 | benjamin7062 | thoughts? |
| 01:46.41 | Bullseye_Network | 20 buttons? |
| 01:46.45 | Bullseye_Network | speed dials? |
| 01:46.57 | benjamin7062 | well, in addition to keypad |
| 01:47.11 | benjamin7062 | Like 'transfer'... conference... mute.. speaker.. park.. etc |
| 01:47.23 | Bullseye_Network | I like Cisco 79xx |
| 01:47.33 | benjamin7062 | Awesome. |
| 01:47.35 | benjamin7062 | Looking them up now |
| 01:47.36 | Bullseye_Network | 7960 are good |
| 01:47.47 | Bullseye_Network | about $260 each |
| 01:47.51 | benjamin7062 | OMG |
| 01:47.53 | benjamin7062 | really? |
| 01:48.24 | Bullseye_Network | to cheap I can get you some for $499. :) |
| 01:48.43 | benjamin7062 | That's awesome... we pay $350 each for 'proprietary' |
| 01:48.45 | benjamin7062 | I can't spell |
| 01:48.52 | benjamin7062 | anyway... how programmable is that screen? |
| 01:49.26 | Bullseye_Network | its supposed to be able to do alot. But I dont use it for more than call info |
| 01:49.28 | benjamin7062 | Can I feed it pretty much whatever I send it via some protocol? |
| 01:50.14 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 01:50.49 | Bullseye_Network | I have never set it up, but, supposely it works as a browser to some extent with xml |
| 01:51.05 | benjamin7062 | Awesome |
| 01:51.07 | *** join/#asterisk b4ka (i=WinNT@200-127-239-114.cab.prima.net.ar) |
| 01:51.12 | benjamin7062 | do you use these phones in your call center... good quality? |
| 01:51.36 | Bullseye_Network | Time for me 2go home.... |
| 01:51.41 | benjamin7062 | Damn |
| 01:51.46 | Bullseye_Network | We use softphone |
| 01:51.47 | benjamin7062 | Thank you 'very' much for the info! |
| 01:51.50 | benjamin7062 | OOohhh |
| 01:51.51 | benjamin7062 | gotcha |
| 01:51.52 | benjamin7062 | okay |
| 01:51.55 | Bullseye_Network | we have 90+ LInux machines |
| 01:52.01 | Bullseye_Network | with sjphones on them |
| 01:52.15 | benjamin7062 | Heh |
| 01:52.22 | benjamin7062 | Well, I have my work cut out... |
| 01:52.41 | Bullseye_Network | email me... info @ bullseyenetworks . com |
| 01:52.45 | Bullseye_Network | Cya L8r |
| 01:52.56 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:52.57 | benjamin7062 | Later man... thank you very much |
| 01:53.01 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 01:53.07 | *** part/#asterisk Bullseye_Network (n=info@72.1.186.66) |
| 01:53.15 | benjamin7062 | Damn, a wealth of info just left the building |
| 01:53.21 | benjamin7062 | Dan -- how far have you gotten? |
| 01:56.46 | raulz | Q: Are there any hardware requirements to have the Asterisk system up and running.I have the box installed with Asterisk@Home |
| 01:57.37 | dan__t | Does it run? |
| 01:58.28 | Idle | hmmm, lame |
| 01:58.46 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 01:58.57 | Idle | it seems a p120 w/ 32mb of ram, wont even detect my wildcard |
| 01:59.35 | P-NuT | do you have zaptel installed? |
| 01:59.38 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:59.41 | Idle | doesn't matter |
| 01:59.52 | P-NuT | doesn't in? |
| 01:59.53 | Idle | its not showing on the PCI bus |
| 02:00.10 | P-NuT | dodgy hardware? |
| 02:00.16 | Idle | p120 hardware :D |
| 02:00.21 | P-NuT | hahahaha |
| 02:00.23 | P-NuT | so true |
| 02:00.35 | Idle | :) |
| 02:00.48 | Idle | I need a half-assed machine... few hundred mhz would do |
| 02:00.49 | P-NuT | if you move it from 1 PCI port to the other does it detect? |
| 02:00.53 | Idle | nope |
| 02:01.10 | P-NuT | If you put it in a real machine, does it work? |
| 02:01.14 | Idle | I've swapped it around with my 2 different video cards a few times... in slots I know that work |
| 02:01.27 | file | is it a TDM400? |
| 02:01.32 | Idle | yea |
| 02:01.42 | file | requires a version 2.2 PCI BUS |
| 02:02.05 | Idle | rofl |
| 02:02.07 | Idle | thatd do it |
| 02:02.31 | DrkShdw | raulz: technically, no you don't need any hardware other than the machine. However, you won't be able to use a POTS line with asterisk without it. |
| 02:02.41 | Idle | it doens't even come to mind anymore... I see a machine and go 'yea, its fine'... forget we used to run into shitty problems like this |
| 02:02.57 | P-NuT | speaking of POTS.. |
| 02:03.03 | file | POTS and pans! |
| 02:03.04 | Idle | n pans ? |
| 02:03.09 | Idle | haha |
| 02:03.26 | P-NuT | Have anyone had an issue with an extension having eexcho, but not on the other end of the PSTN line? |
| 02:04.19 | DrkShdw | So.. I'm a little annoyed. we had a storm come through. 3 tornados touched down (minor damage) but the power spiked several times. it killed my battery backup AND the power supply in my main XP machine :/ |
| 02:04.50 | raulz | is there a full step by step website that guides me in setting up an asterisk system except asterisk.org ? |
| 02:07.55 | bugz | http://arstechnica.com/news.ars/post/20060621-7101.html |
| 02:07.59 | bugz | we are doomed. |
| 02:08.07 | bugz | i think i'll have a coke. |
| 02:08.40 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 02:15.43 | FuriousGeorge | i have a client that wants to use his voip server (opteron 142 w/ scsi drives) as a file server for a few people with low bandwith needs. i told him i dont think its a great idea but i wanted to talk it over with someone. the pbx itself is not at a load at all |
| 02:17.03 | FuriousGeorge | on a scale of 1 to 10 how bad of an idea is that? |
| 02:17.56 | FuriousGeorge | i guess a better question would be if consistent media io on scsi drives will significantly effect the quality of their phone system |
| 02:22.22 | YoYo | how many concurrent calls are you talking about? |
| 02:22.55 | YoYo | and are you monitoring them? |
| 02:26.16 | DrkShdw | Relaistically, your phone system is a poor choice to use as your file sharing system. I personally would nix the idea. |
| 02:26.55 | DrkShdw | realistically too. I can type, I swear it. |
| 02:27.19 | b4ka | what the hell is this bug of the -rx commands? isnt it fixed in any stable version? i dont want to go patching |
| 02:28.14 | benjamin7062 | I've found a ton of wiki's and sites listing a 'ton' of Hard Phones... Are any of you guys using hard phones? If yes, which ones? |
| 02:28.47 | DrkShdw | Snom 320 here |
| 02:28.49 | b4ka | sipura and linksys adapters |
| 02:28.55 | b4ka | like 20 of each |
| 02:29.14 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 02:29.21 | P-NuT | are Snom phones any good? |
| 02:29.28 | P-NuT | Their incredibly pricy |
| 02:29.43 | benjamin7062 | Looking these up right now... thank you... Specifically I'm looking for phones with a lot of programmable buttons (call fwd, transfer, etc)... if you have any suggestions on those. Price is no object |
| 02:29.50 | benjamin7062 | well, reasonably |
| 02:29.57 | DrkShdw | the 320's are good. the 360's were a pile. |
| 02:29.57 | benjamin7062 | I will be buying around 50 |
| 02:30.00 | b4ka | polycom phones are pretty nice |
| 02:30.05 | b4ka | they have all the crap |
| 02:30.13 | benjamin7062 | Looking those up too |
| 02:30.17 | b4ka | and are very well made |
| 02:32.00 | b4ka | http://www.polycom.com/products_services/0,1443,pw-34-182-12072,00.html |
| 02:32.14 | benjamin7062 | heh |
| 02:32.17 | benjamin7062 | top link on google |
| 02:32.38 | benjamin7062 | And they just happen to be here in Austin TX |
| 02:32.39 | benjamin7062 | suweet |
| 02:32.42 | b4ka | i have 2 of those |
| 02:32.54 | b4ka | and some soundstations |
| 02:33.13 | benjamin7062 | Do you recall cost? |
| 02:33.19 | benjamin7062 | ballpark? |
| 02:33.59 | b4ka | not really |
| 02:34.17 | benjamin7062 | anything under $300 works for me |
| 02:34.36 | benjamin7062 | Maybe I'll get a bulk rate @ 50 units... maybe |
| 02:34.38 | benjamin7062 | =) |
| 02:35.02 | benjamin7062 | Will an asterisk system handle 50+ users well? |
| 02:35.06 | benjamin7062 | 2 PRI's? |
| 02:35.16 | b4ka | we have more than 50 |
| 02:35.18 | b4ka | with 1 pri |
| 02:35.21 | benjamin7062 | OMG |
| 02:35.23 | benjamin7062 | Awesome |
| 02:35.27 | benjamin7062 | Don't know if you saw above |
| 02:35.31 | benjamin7062 | but I have 1 week to get this working |
| 02:35.34 | benjamin7062 | and prove it |
| 02:35.38 | benjamin7062 | then we replace Avaya |
| 02:35.40 | b4ka | of course at 60% capacity it would collapse |
| 02:35.51 | b4ka | but then, we never get to that point |
| 02:36.09 | b4ka | heh |
| 02:36.20 | benjamin7062 | 60 Capacity.. meaning.. using all 23 channels on your pri? |
| 02:36.37 | b4ka | 23_ |
| 02:36.51 | b4ka | it depends on you link |
| 02:36.51 | benjamin7062 | Right |
| 02:36.58 | b4ka | you can have more or less channels |
| 02:37.18 | benjamin7062 | Right... I get that... but as far as your 'system'.. i can handle the 'load' of more than 23 |
| 02:37.19 | b4ka | and we do have an IAX provider, 4 telco lines and a gsm phone attached to the asterisk |
| 02:37.24 | benjamin7062 | if you had the pri capacity |
| 02:37.27 | b4ka | plus the t1 and a channelbank |
| 02:37.42 | benjamin7062 | woah |
| 02:38.22 | benjamin7062 | If I'm 'very' solid in linux... to the dev level.. chances of getting 5 phone demo up in a week? |
| 02:38.34 | b4ka | in half a day |
| 02:38.38 | DrkShdw | benjamin7062: thats a relative question.. a 486 system running asterisk? probably not. a p4 2.0ghz could |
| 02:38.39 | benjamin7062 | OMG |
| 02:38.42 | benjamin7062 | so sweet |
| 02:39.04 | b4ka | if you are just going to put 5 phones interconected and nothing else |
| 02:39.08 | b4ka | its a piece of cake |
| 02:39.20 | DrkShdw | you hooking a POTS line up to it? |
| 02:39.25 | b4ka | basic dialplan and some sip config |
| 02:39.40 | benjamin7062 | DrkShdw -- of course... I'll be more specific... I'll run whatever hardware necessary to run 50+ simultaneous internal and external calls... |
| 02:39.44 | benjamin7062 | Money is no object |
| 02:39.50 | benjamin7062 | we are about to blow 30K upgrading this avaya |
| 02:39.58 | benjamin7062 | I want to replace it |
| 02:39.59 | b4ka | hehe |
| 02:40.01 | DrkShdw | hardware notwithstanding, yes. asterisk is capable of it. |
| 02:40.02 | benjamin7062 | Owner gave me a week to prove it |
| 02:40.16 | b4ka | i just replaced our old asterisk yesterday |
| 02:40.29 | benjamin7062 | We are hooking 2 pri's.. not pots... |
| 02:40.34 | b4ka | its on a p4 nice intel mobo 2gb ram and satas |
| 02:40.36 | b4ka | nothing fancy |
| 02:40.40 | benjamin7062 | but for the test.. I just need 5 internal phones talking to each other with extensions |
| 02:40.49 | DrkShdw | benjamin7062: for a demo? just download trixbox, throw it on a machine. add 5 extensions, and setup an IVR. 2 hours, and you have a working demo. |
| 02:40.51 | b4ka | get yourself a wan card |
| 02:40.56 | b4ka | a decent machine |
| 02:40.57 | b4ka | and you're done |
| 02:41.23 | b4ka | i have 3 sangoma cards |
| 02:41.27 | b4ka | AFT102 |
| 02:41.44 | benjamin7062 | So, the actual config is somewhat trivial? I read about this project years ago and it seemed pretty involved.. Nothing I can't do.. Just the time frame to learn it .. if it is crazy involved. |
| 02:42.01 | benjamin7062 | But the demo must be using hard phones... |
| 02:42.02 | benjamin7062 | =) |
| 02:42.05 | b4ka | well the dialplan can get complicated |
| 02:42.20 | b4ka | but for only 5 phones and nothing else |
| 02:42.21 | benjamin7062 | Polycom happens to be here in Austin, TX... so I might be able to get my hands on 5 in a couple days |
| 02:42.24 | b4ka | its piece of cake |
| 02:42.30 | DrkShdw | right. the timeframe (learning curve) is the part that'll kill you. hence the trixbox suggestion. you can set it up and show how it works in 2 hours. THEN if you want, roll your own asterisk install. |
| 02:42.38 | benjamin7062 | Dial plan on 'any' phone switch can be complicated.. =) |
| 02:43.05 | b4ka | you can get voip adapters too benjamin |
| 02:43.11 | benjamin7062 | Okay... so trixbox is 'kinda' like a knoppix'ish version of asterisk... I couldn't tell if it was a fork or what... |
| 02:43.12 | b4ka | with 3 you can connect 6 phones |
| 02:43.24 | b4ka | and i think they are cheapear |
| 02:43.26 | b4ka | than phones |
| 02:43.28 | *** join/#asterisk smackus2 (n=smackus2@c-67-161-244-209.hsd1.ut.comcast.net) |
| 02:43.47 | benjamin7062 | Well, replacing this phone switch.. they want to see similar on phone functionality.. like, transfers, conference calls, hold, etc. |
| 02:43.52 | DrkShdw | yes, trixbox is the OS (CentOS) FreePBX, Asterisk, and a few other niceties all on a single "boot and install" cd |
| 02:44.04 | benjamin7062 | Sweet |
| 02:44.18 | benjamin7062 | That is what I'll do... |
| 02:44.25 | b4ka | i would just try to understand the extensions.conf |
| 02:44.28 | b4ka | sip.conf is trivial |
| 02:44.30 | DrkShdw | trixbox seems to be taboo in this channel. I personally like it. :) |
| 02:44.37 | b4ka | and you dont need much more to do what you want |
| 02:45.03 | b4ka | and you could get a grasp about what you'll have to do for 50 machines then ;P |
| 02:45.10 | denon | it's not that it's taboo, just that we're not here to support it |
| 02:45.13 | denon | we're here to support asterisk |
| 02:45.22 | DrkShdw | you can have a digital receptionest, music on hold, transfers, and conferencing setup in no time |
| 02:45.31 | benjamin7062 | I can learn this stuff relatively quickly (I hope)... Been managing Nortel's and Other systems for years.. I get the concepts.. just perhaps not the asterisk specifics... but I understand the concepts of DIalplan's, etc. |
| 02:45.34 | DrkShdw | denon: I realize that ;-) |
| 02:45.35 | denon | you'll find more knowledge on trixbox in a channel devoted to it |
| 02:45.41 | benjamin7062 | Man, you guys rock.. thank you! |
| 02:46.08 | b4ka | speaking of which, whats with that -rx not showing output bug! |
| 02:46.23 | b4ka | why wasnt it fixed in 1.2.9 :( |
| 02:46.27 | *** join/#asterisk Samoied (n=Samoied@201.22.205.152.adsl.gvt.net.br) |
| 02:46.49 | benjamin7062 | Well, understanding that it is taboo, I will probably roll a full asterisk install but might use trixbox for the demo if you guys feel it accomplishes the task of proving that 5 hardphones can have business like functions, etc. |
| 02:47.32 | DrkShdw | trixbox would be ideal for a quick demo. the point of the demo is to show it's capable. save the rolling your own, for after the project is accepted. |
| 02:47.58 | DrkShdw | no sense spending half a day rolling out a demo that they may reject (IMO) |
| 02:48.21 | smackus2 | ok, so i am trying to get all of my channels to work on my te411p. I have 4 t1s in the card. all are pri. so i am setting them up with channel => 1-23,25-47,49-71,73-95. |
| 02:48.21 | smackus2 | 1-47 load correctly, then i get the error Jun 21 20:36:14 ERROR[17978]: chan_zap.c:10317 setup_zap: Unable to reconfigure channel '49-71' |
| 02:48.21 | smackus2 | where did i go wrong? |
| 02:48.53 | smackus2 | sorry about the name change... problems with my irc client tonight |
| 02:48.57 | benjamin7062 | Exactly... although, it won't be rejected if it works... They are going to spend the 30K tomorrow... I've been reading about this for years ... They are willing to take it on if it will definately support 50 users. |
| 02:49.21 | smackus2 | i currently have 88 users working awesome |
| 02:49.42 | benjamin7062 | 30+ hitting the phones pretty hard? |
| 02:49.47 | smackus2 | yes |
| 02:49.51 | benjamin7062 | sounds like it with 4 pri's... =) |
| 02:49.53 | smackus2 | it is an outbound call center with dialer |
| 02:49.58 | DrkShdw | well, like I said, a very modest system can handle 5 phones for a demo. as in, a p200 would be overkill. for 50 lines, your hardware demands will rise a bit ;-) |
| 02:50.00 | benjamin7062 | Sweet |
| 02:50.06 | smackus2 | i am building a redundant system right now. |
| 02:50.16 | smackus2 | quad xeon procs and lots of ram. |
| 02:50.21 | smackus2 | we do alot of recording and reporting. |
| 02:50.29 | benjamin7062 | Okay |
| 02:50.30 | smackus2 | we also are doing a public conference bridge. |
| 02:50.37 | smackus2 | it gets hit hard |
| 02:50.45 | smackus2 | the secret in my opinion is to scale out. |
| 02:51.03 | smackus2 | use other servers for things like converting recordings to mp3/gsm |
| 02:51.08 | benjamin7062 | Multiple boxes talking to each other and fishing out specific services to other boxes? |
| 02:51.14 | smackus2 | reports databases also off of the asterisk |
| 02:51.18 | smackus2 | yeah, something like that |
| 02:51.31 | smackus2 | nfs is what I use. I just pull the recordings off to another box |
| 02:51.40 | benjamin7062 | That'll be no problem |
| 02:51.54 | smackus2 | and the odbc and mysql support in asterisk is nice. I just push that to my database/web server |
| 02:51.58 | smackus2 | for reporting |
| 02:52.34 | benjamin7062 | I work for a company that stores 40+ terabytes of digital images... So hardware for this sounds rather trivial... which I like to hear. PRI cards... You guys all suggest the Digi's? |
| 02:52.46 | smackus2 | i do |
| 02:53.09 | DrkShdw | man for the $30,000 they were planning to drop on a new system.. they could have a beefy asterisk setup, AND still afford to give you $25k ;-) |
| 02:53.09 | smackus2 | they have made great improvements in processing so that the load on the server is not so harsh |
| 02:53.10 | benjamin7062 | I notice the voltage difference... is that simply for the different PCI spec's... (old and newer) |
| 02:53.33 | smackus2 | exactly, with 2 servers with raided hd and dual procs, we are only in $3000 |
| 02:53.53 | smackus2 | that is with the T1 card for the asterisk box |
| 02:54.02 | benjamin7062 | Well, remember, I have to use 50+ hard phones... so add $200ish x 50 ... |
| 02:54.04 | benjamin7062 | 10K ish |
| 02:54.07 | benjamin7062 | still WAY cheaper |
| 02:54.10 | smackus2 | no... |
| 02:54.18 | smackus2 | $100 ish |
| 02:54.34 | smackus2 | I use the polycom 301 and 501... 301 for agents, and 501 for managers. |
| 02:54.34 | dlynes_home | smackus2: no $200ish |
| 02:54.35 | benjamin7062 | Well, thing is... they want all the business buttons on the phones... someone suggested the Polycoms... |
| 02:54.41 | file | buying Digium hardware helps developers buy food, please - think of the developers! |
| 02:54.48 | dlynes_home | smackus2: where the hell are you getting 501s for $100? |
| 02:54.54 | smackus2 | i pay $105 for the 301 and $150 for the 501 |
| 02:55.03 | dlynes_home | smackus2: damn...where? |
| 02:55.05 | benjamin7062 | I will .. in fact.. most likely we'll buy business edition JUST for the fact that we WANT to support them |
| 02:55.10 | smackus2 | local dealer |
| 02:55.21 | benjamin7062 | smackus -- don't suppose you are in TX? |
| 02:55.25 | smackus2 | gimme a sec, and I will see if i can find the contact info. |
| 02:55.27 | smackus2 | they ship |
| 02:55.28 | dlynes_home | smackus2: and that's fully supported? i.e. you get firmware updates with that? |
| 02:55.53 | smackus2 | if you get set up with a sales rep, they should provide the fw updates. |
| 02:56.17 | dlynes_home | smackus2: and that's wholesale price? |
| 02:56.35 | smackus2 | well, thats the price we got for ordering 50+ |
| 02:56.40 | benjamin7062 | smackus2 - dif between a 500 and 501? |
| 02:56.45 | file | yay BE |
| 02:56.52 | dlynes_home | benjamin7062: 501 is 500 with newer firmware |
| 02:56.58 | benjamin7062 | okay |
| 02:56.59 | benjamin7062 | kewl |
| 02:57.02 | smackus2 | that is my understanding |
| 02:57.32 | dlynes_home | benjamin7062: so the memory might have increased as well...but hardware-wise, i think the two are the same |
| 02:57.58 | benjamin7062 | We'll probably go with the 601 for the screen size... key is, I need as many buttons as I can get on the phone |
| 02:58.09 | benjamin7062 | smackus2 - those buttons are completely programmable via asterisk right? |
| 02:58.33 | dlynes_home | benjamin7062: i believe only the soft keys are |
| 02:58.36 | smackus2 | that I have not gotten into, the phone is fully programmable, as to what each button is capable of, I do not know. |
| 02:58.45 | dlynes_home | benjamin7062: and even then, it's phone programmable to an extension on asterisk |
| 02:58.54 | benjamin7062 | Gotcha |
| 02:59.05 | benjamin7062 | Ie... it dials some 'code' for asterisk that = a function |
| 02:59.07 | benjamin7062 | gotcha |
| 02:59.17 | dlynes_home | exactly |
| 02:59.35 | smackus2 | i think this is the correct contact info.... |
| 02:59.38 | smackus2 | Alliance Communication Systems & Wiring |
| 02:59.43 | benjamin7062 | OMG |
| 02:59.45 | benjamin7062 | that's who I was going to use |
| 02:59.46 | smackus2 | (303) 679-1371 |
| 02:59.54 | smackus2 | they are a dealer for polycom |
| 03:00.16 | dlynes_home | smackus2: will they ship out of country? |
| 03:00.24 | file | freaky |
| 03:00.25 | smackus2 | that, I cannot answer |
| 03:00.31 | dlynes_home | ah |
| 03:00.36 | smackus2 | I do not see why not, however it may up the price a bit :-D |
| 03:00.46 | benjamin7062 | Wish they were here in Austin |
| 03:00.55 | benjamin7062 | They are outta Plano |
| 03:01.05 | smackus2 | well, i am in utah, so shipping wont be too much more |
| 03:01.19 | benjamin7062 | It's timing really |
| 03:01.23 | smackus2 | ah |
| 03:01.27 | smackus2 | i get them in two days |
| 03:01.28 | benjamin7062 | I'd love to get these 'tomorrow' |
| 03:01.35 | benjamin7062 | as in, go to a store.. =) |
| 03:01.47 | benjamin7062 | Polycom is here in Austin, so maybe I can call them direct. |
| 03:01.49 | [TK]D-Fender | dlynes_home :Soft keys aren't really that programmable |
| 03:01.50 | dlynes_home | cool...emailed it myself |
| 03:02.03 | dlynes_home | [TK]D-Fender: but you can program them to an extension right? |
| 03:02.04 | benjamin7062 | They aren't? |
| 03:02.13 | [TK]D-Fender | dlynes_home : Sure. And for presence. |
| 03:02.14 | dlynes_home | [TK]D-Fender: that's what i gathered from reading the manual, anyways |
| 03:02.16 | file | [TK]D-Fender: ! |
| 03:02.20 | *** join/#asterisk dviner (n=dviner@70-38-9-7.vnnyca.adelphia.net) |
| 03:02.22 | [TK]D-Fender | dlynes_home : today I set mine for ACD login/out |
| 03:02.41 | smackus2 | [TK]D-Fender: i still have not gotten that to work |
| 03:02.44 | dlynes_home | and so what do you mean by "soft keys aren't really that programmable" then? |
| 03:02.47 | dlynes_home | kinda misleading |
| 03:02.49 | [TK]D-Fender | dlynes_home : Don't have Bweschke's tree compiled, but wanted to see the phone side. Nice |
| 03:03.02 | [TK]D-Fender | dlynes_home : Sofkey's ~= line keys |
| 03:03.05 | [TK]D-Fender | != |
| 03:03.13 | dlynes_home | Yeah, I'm aware of that |
| 03:03.15 | dlynes_home | never said they were |
| 03:03.22 | dlynes_home | never told benjamin7062 that, either |
| 03:03.30 | [TK]D-Fender | benjamin7062 : Odds are if you need it, Polycom's got it. |
| 03:03.51 | dlynes_home | bweschke's tree for what? phone side for what? |
| 03:04.15 | smackus2 | benjamin7062 take a look at the SIP.cfg for polycom, that will give you a good indication of all of the available features |
| 03:04.18 | benjamin7062 | Perhaps I could be more vague. I need keys that can login/out of ACD, Transfer calls, Conference calls, etc... IE function keys. I don't know what they are called.. but can this be 'accomplished' even if it's some hack ass solution? |
| 03:04.26 | benjamin7062 | okay |
| 03:04.27 | benjamin7062 | I will |
| 03:04.39 | dlynes_home | benjamin7062: yes, through the soft keys |
| 03:04.41 | smackus2 | it is available, without a hack ass solution |
| 03:04.46 | benjamin7062 | awesome |
| 03:04.51 | smackus2 | take a look at the SIP.cfg |
| 03:04.51 | [TK]D-Fender | benjamin7062 : They you're set with any Polycom. Its jsut a question about PoE / Speakerphone, call volume, etc |
| 03:04.57 | benjamin7062 | Going to do that right now |
| 03:05.14 | [TK]D-Fender | Then* |
| 03:05.39 | benjamin7062 | They don't support PoE? Bummer. |
| 03:05.44 | [TK]D-Fender | benjamin7062 : Are you hoping to use PoE with your phones? What kind of call volume? Budget issues? |
| 03:05.50 | [TK]D-Fender | benjamin7062 : Sure they do |
| 03:05.53 | smackus2 | the 501 is Poe |
| 03:06.02 | smackus2 | as is the 301 |
| 03:06.02 | [TK]D-Fender | smackus2 : ALL of them can do PoE |
| 03:06.03 | benjamin7062 | Budget = 30K .. start to finish... System + 50 Phones |
| 03:06.16 | smackus2 | that would be major overkill |
| 03:06.29 | benjamin7062 | Oh yeah, I have 1 week to learn asterisk and 3 weeks to build the system |
| 03:06.31 | [TK]D-Fender | benjamin7062 : You'll come in WAY under that. |
| 03:06.38 | smackus2 | what features are you wishing to offer? ie call recording, reporting, conference bridge? |
| 03:06.38 | dlynes_home | [TK]D-Fender: i thought the 301 didn't come with poe though? i.e. it was extra? |
| 03:06.39 | benjamin7062 | I know |
| 03:06.51 | smackus2 | it does come with a power cord :-D |
| 03:06.54 | [TK]D-Fender | dlynes_home : It IS extra, but its still THERE |
| 03:06.55 | benjamin7062 | but 30K is what we are about to spend on Avaya... Which is why I'm hear soaking everything in I can |
| 03:07.11 | [TK]D-Fender | benjamin7062 : So again, are you looking to use PoE? |
| 03:07.16 | benjamin7062 | Yes sir |
| 03:07.34 | smackus2 | avaya is going to be the same thing as asterisk, but a rip off in price |
| 03:07.39 | [TK]D-Fender | benjamin7062 : Ok, a la cheap : D-Link DES-1526 24port PoE Switch = $400 +/- |
| 03:07.46 | smackus2 | it is linux based, and you have to pay for each license and application need |
| 03:08.05 | file | [TK]D-Fender: ooh |
| 03:08.09 | [TK]D-Fender | benjamin7062 : And for your lower end users, get them IP430's, and your receptionist an IP601 + 2 Attendant Modules |
| 03:08.23 | [TK]D-Fender | benjamin7062 : All PoE native. |
| 03:08.27 | smackus2 | for what I have built comapared to what I needed from avaya, they wanted to charge me 1.5 million |
| 03:08.41 | file | only 1.5 million? :) |
| 03:08.48 | [TK]D-Fender | benjamin7062 : Anybody you'd consider getting an IP 501 for you're probably better off just getting them a 601. |
| 03:09.00 | smackus2 | that was after stripping the system down to only one fail over box |
| 03:09.17 | benjamin7062 | Features: ACD/Hunt Groups, Basic Business Stuff (Transfers, Conference, HOld Music), Reporting (if it's in a DB/FIles, I can write the code for this), Programmable Hard PHones, Good quality, 2 PRI support, etc... basically, same functionality of an Avaya IP system... are there apps to monitor call queues and stuff out there? Ie.. Open source projects? |
| 03:09.44 | dlynes_home | smackus2: is their phone system made out of brass? |
| 03:09.56 | [TK]D-Fender | benjamin7062 : Yup, Queue reporting stuff out there already, * does al the basic PBX stuff you need, Polycom's are your #1 choice, and easy on the budget. |
| 03:09.57 | smackus2 | form the price i had thought tTi |
| 03:09.59 | DrkShdw | benjamin7062: yes, there are apps out there for all that |
| 03:10.00 | smackus2 | Ti |
| 03:10.37 | smackus2 | most apps are built into asterisk, you just need to figure out your front end.. ie, php, java |
| 03:10.38 | DrkShdw | there are even apps out there for calling card billing, and such. it's really limitless |
| 03:10.43 | dlynes_home | smackus2: cause they sure have brass balls if they're charging 1.5M for a phone system |
| 03:10.52 | smackus2 | i hear ya!!!! |
| 03:11.11 | benjamin7062 | I can use any, php, java, x windows if needed,, hopefully windows interface (or java or something that will port nicely) |
| 03:11.14 | smackus2 | that was with all of the reporting and recording and such that I already have running in asterisk for free |
| 03:12.24 | dlynes_home | file: is digium planning any upgrades on their hardware to support any of the modern advances in hardware? |
| 03:12.55 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 03:13.32 | smackus2 | [TK]D-Fender: can you tell me the link you used for svn for the Bweschke's tree? i was unsuccessful in connecting it |
| 03:14.17 | [TK]D-Fender | smackus2 : I haven't DL'd it myself yet |
| 03:14.45 | file | dlynes_home: define modern advances |
| 03:15.17 | file | even if I did know I couldn't tell you |
| 03:15.20 | smackus2 | then he would kill you |
| 03:15.47 | [TK]D-Fender | LOL! |
| 03:15.56 | file | yay killing |
| 03:16.11 | dlynes_home | file: i was thinking stuff like autodetecting 3.3V vs 5.0V, onboard carrier grade echo cancellers, ... |
| 03:16.15 | [TK]D-Fender | file : If they really earn it I'll lend you my new toy :) |
| 03:16.19 | file | ah |
| 03:16.28 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:16.59 | file | there's one thing that's iffy, it's in zaptel but I don't know if an announcement has been made yet |
| 03:17.13 | dlynes_home | file: about? |
| 03:17.27 | file | carrier grade echo cancellation |
| 03:17.44 | dlynes_home | software-based carrier grade echo cancellation? |
| 03:17.48 | dlynes_home | or hardware-based? |
| 03:17.50 | file | hardware |
| 03:18.10 | dlynes_home | so digium is going to have carrier grade echo cancellers now? |
| 03:18.15 | dlynes_home | on analog and pri cards? |
| 03:18.37 | [TK]D-Fender | dlynes_home : If you don't know about it the answer is "not yet". |
| 03:19.00 | file | uh let me phrase it this way |
| 03:19.08 | dlynes_home | [TK]D-Fender: i thought it was RSN? |
| 03:19.14 | file | the driver for a board that uses the Octasic stuff *may* be in zaptel trunk |
| 03:19.29 | dlynes_home | file: oh..you mean sangoma? |
| 03:19.30 | [TK]D-Fender | dlynes_home : RSN? |
| 03:19.35 | dlynes_home | "real soon now" |
| 03:19.38 | file | dlynes_home: no. |
| 03:19.46 | [TK]D-Fender | :D |
| 03:19.54 | [TK]D-Fender | dlynes_home : Time will tell, and someone elses validation :) |
| 03:20.07 | *** join/#asterisk sep (n=sep@217.17.211.40) |
| 03:20.12 | dlynes_home | file: ah...they were just the only ones i knew of that used octasis |
| 03:20.13 | dlynes_home | file: ah...they were just the only ones i knew of that used octasic |
| 03:20.34 | dlynes_home | file: everyone i know using hwec with digium is using tellabs |
| 03:20.35 | file | dlynes_home: octasic's echo canceller is very very good, but also very very expensive |
| 03:20.54 | file | maybe it's only one very... |
| 03:21.08 | [TK]D-Fender | dlynes_home : Oh! You mean manxpower! |
| 03:21.09 | [TK]D-Fender | ;) |
| 03:21.30 | dlynes_home | [TK]D-Fender: he's the only one i know of using hwec with digium cards :p |
| 03:21.30 | [TK]D-Fender | file : I prefer to think of it as "very very worth every penny for even 1 port" ;) |
| 03:21.51 | Qwell | file: only one very on which? |
| 03:21.52 | orlock | heh |
| 03:21.53 | [TK]D-Fender | dlynes_home : I've seen plenty come through here with TE406P's and an odd TE411P |
| 03:21.59 | orlock | one of our clients complained about echo |
| 03:22.11 | orlock | they were calling the voip phone hands-free from a phone next to it.. DUH |
| 03:22.13 | file | Qwell: price of the Octasic |
| 03:22.15 | dlynes_home | [TK]D-Fender: what's a te406p or a te411p? |
| 03:22.30 | Qwell | two, very very good, and very (perhaps an additional very) expensive? |
| 03:22.37 | file | yes |
| 03:22.38 | Qwell | s/two/so/ |
| 03:22.46 | Qwell | not sure how I slaughtered that so badly |
| 03:22.52 | dlynes_home | [TK]D-Fender: te405 and te410 with hwec? |
| 03:22.58 | [TK]D-Fender | dlynes_home : OMG... go rad up on Digium's products! |
| 03:23.17 | [TK]D-Fender | Read* |
| 03:23.18 | dlynes_home | [TK]D-Fender: ? |
| 03:23.25 | file | our product numbers blow my mind, I just... nod my head |
| 03:23.29 | *** join/#asterisk techie (n=gus@voipops.net) |
| 03:25.22 | file | eep |
| 03:25.25 | file | I expect that back! |
| 03:25.35 | Qwell | You can have it back in...18 days! |
| 03:25.45 | file | ooh 18? is that it?!? |
| 03:25.55 | Qwell | I think so...unless my math is horribly b0rked |
| 03:26.02 | file | might be |
| 03:26.10 | Qwell | nope, it's correct(ish) |
| 03:26.23 | Qwell | that last week kinda flew by, eh? |
| 03:26.33 | file | I don't remember it |
| 03:26.42 | benjamin7062 | How does EC relate to all this VOIP... is it analyzing the voice and trying to remove unwanted noise from the audio? |
| 03:27.02 | file | benjamin7062: we're talking about in relation to analog and PRIs |
| 03:27.04 | Qwell | benjamin7062: no, EC (echo cancellation) is mostly for analog/pri |
| 03:27.51 | [TK]D-Fender | benjamin7062 : http://www.voip-info.org/wiki/view/Causes+of+Echo |
| 03:27.58 | DrkShdw | head -18 file? I get it! haha a *nix joke! |
| 03:28.06 | file | although you could incorporate a software echo canceler into an asterisk channel... hrm |
| 03:28.09 | Qwell | DrkShdw: -n18! |
| 03:28.12 | Qwell | sheesh |
| 03:28.14 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:28.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:28.22 | file | russellb: !!?!! |
| 03:28.35 | russellb | greetings |
| 03:28.40 | Qwell | russellb: omg! |
| 03:28.40 | file | greets to my peep! |
| 03:29.01 | Qwell | DrkShdw: You're a pretty hardcore geek btw... |
| 03:29.09 | DrkShdw | yes, yes I am |
| 03:29.27 | Qwell | anybody who can make a CLI joke, with 1980s syntax...well...yeah |
| 03:29.42 | DrkShdw | I dream in binary :/ it's pretty sad actually |
| 03:30.21 | DrkShdw | 1980's syntax. psshhh head -18 file will work today :P |
| 03:30.29 | file | I feel so violated |
| 03:30.38 | [TK]D-Fender | :O |
| 03:30.39 | Qwell | DrkShdw: but, perhaps not tomorrow |
| 03:30.44 | Qwell | in fact, any time now |
| 03:30.47 | DrkShdw | it will on MY boxen :P |
| 03:30.53 | Qwell | it's been deprecated for like...I don't know... |
| 03:30.54 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 03:30.55 | Qwell | TEN YEARS |
| 03:31.10 | Qwell | That's one hell of a deprecation, if you ask me :p |
| 03:31.39 | DrkShdw | thats odd, for a deprecated command, it's standard on every bsd/linux distro I can think of |
| 03:31.48 | Qwell | yeah...it bugs me |
| 03:31.55 | Qwell | incredibly so |
| 03:32.05 | file | buggy Qwell! |
| 03:32.15 | dlynes_home | head -18 only shows you the first 18 characters of the file, no? |
| 03:32.23 | Qwell | dlynes_home: no, it's equiv to -n18 |
| 03:32.26 | DrkShdw | 18 lines |
| 03:32.32 | dlynes_home | Qwell: not on my distro |
| 03:32.36 | Qwell | dlynes_home: everywhere :p |
| 03:32.42 | Qwell | it's a former posix standard |
| 03:32.48 | benjamin7062 | head is not 80's syntax.. =) |
| 03:32.56 | Qwell | benjamin7062: no, but -18 is |
| 03:32.59 | Qwell | -18 vs -n18 |
| 03:33.08 | file | Qwell: btw you're not invited to my special muffin party |
| 03:33.14 | Qwell | WTF |
| 03:33.18 | DrkShdw | lol |
| 03:33.22 | Qwell | die |
| 03:33.33 | DrkShdw | if we rm -rf file, will he be defiled? |
| 03:33.34 | benjamin7062 | Hmm, thought that was still relevent on Solaris |
| 03:33.35 | russellb | file: am i? |
| 03:33.39 | benjamin7062 | I lose at life |
| 03:33.42 | dlynes_home | hrm...could've sworn one of my installs didn't do 18 lines with that |
| 03:33.44 | file | russellb: of course, you're the cohost of the party! |
| 03:33.48 | russellb | w00t |
| 03:33.50 | Qwell | O M G |
| 03:33.59 | *** join/#asterisk blebleble (n=ble@d60-65-143-132.col.wideopenwest.com) |
| 03:34.10 | Qwell | That's fine... No special Qwell muffins for you |
| 03:34.18 | blebleble | anyone had luck with tying asterisk + iaxmodem + hylafax? |
| 03:34.38 | file | Qwell: excellent, I heard they were explosives anyway |
| 03:34.43 | Qwell | file: extremely so |
| 03:34.54 | DrkShdw | I know a guy who had the AIM name 'file' He used to have logs of all the RANDOM crazy IM's he'd get from people screwing up |
| 03:35.23 | orlock | DrkShdw: hah! |
| 03:35.24 | file | I have directory also |
| 03:35.32 | orlock | DrkShdw: hear about the guy with the numberplate "NULL" |
| 03:35.33 | Qwell | file: on aim? |
| 03:35.38 | file | no, on here :P |
| 03:35.42 | Qwell | well..duh :p |
| 03:35.46 | DrkShdw | Qwell: yeah. his name is Shaun |
| 03:35.48 | russellb | i have drumkilla! |
| 03:35.53 | DrkShdw | orlock: nope |
| 03:36.04 | Qwell | I have...like...well...mine sucks |
| 03:36.21 | file | russellb: you kill drums :( |
| 03:36.26 | DrkShdw | he also has 'AIM User' lol |
| 03:36.27 | orlock | DrkShdw: whenever the cops were filling our a report for something vehicle related, if they couldent enter the plate number for some reason, they would enter "NULL" |
| 03:36.42 | russellb | heh, i used to register randoim aim names |
| 03:36.45 | DrkShdw | orlock: HAHA I bet that was fun as hell |
| 03:37.04 | orlock | DrkShdw: yeah, he started getting all sorts of weird and freaky infringment notices |
| 03:37.27 | file | russellb: oh and what are you doing on IRC, I totally banned you from it - nub! |
| 03:37.27 | Qwell | russellb: If you count up all the ones that I've owned over time...eeks |
| 03:37.35 | file | IAX! |
| 03:37.35 | Qwell | several hundred, at least :p |
| 03:37.36 | DrkShdw | "Your license is going to be suspended in 30 days" letters, 15,000 times a day |
| 03:38.05 | russellb | file: fine :'( |
| 03:38.16 | file | I have the power! |
| 03:38.17 | Qwell | file: now look what you've done |
| 03:38.39 | file | I couldn't let him waste his night on IRC |
| 03:41.27 | benjamin7062 | Eww.. Important question... are there cards that will 'feed' analog lines (for fax machines) that work with * |
| 03:41.28 | benjamin7062 | ? |
| 03:42.20 | DrkShdw | an FXS card (or module) will. I'e never personally done it, but I've read about it |
| 03:43.42 | benjamin7062 | I also know that you can get consumer crap,... like vonage boxes that can talk sip and spit out analog... just didn't know if anyone knew of one for * |
| 03:43.54 | benjamin7062 | I can read... or buy a pots for the few fax machines we have |
| 03:45.26 | DrkShdw | well, faxing over voip, is tricky at best. I would say: you really need an analog line for 911, and alarm systems.. just make it your actual fax line as well.. all independant of asterisk |
| 03:45.48 | benjamin7062 | Good call |
| 03:46.54 | SkramX | ccccc |
| 03:47.02 | SkramX | trixter around? |
| 03:47.34 | DrkShdw | speaking of 911. Anyone know right off hand, of how to configure the outbound routes to send 411, and 911 out through the POTS line? |
| 03:48.14 | DrkShdw | a guide would work, if you can point me in that direction. |
| 03:48.23 | Qwell | exten => 411,1,Dial(Zap/g1/${EXTEN}) |
| 03:49.05 | DrkShdw | oh lordy. I coulda found that in prolly 2 seconds of playing with the files. thanks qwell |
| 03:49.16 | Qwell | mmhmm |
| 03:52.12 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:53.12 | *** join/#asterisk bmg505 (n=leon@c1-70-15.rndf.isadsl.co.za) |
| 03:55.05 | *** join/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 03:55.52 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-70.cybersurf.com) |
| 03:56.21 | znoG | has anyone tried those USB phones that can work with DIAX, etc? |
| 03:59.21 | *** part/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 04:10.59 | *** join/#asterisk flujan (i=flujan@201-27-90-194.dsl.telesp.net.br) |
| 04:11.34 | flujan | hi guys... I'm getting this error message in my asterisk box: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 04:11.53 | flujan | but all zap channels are available... |
| 04:20.14 | *** part/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 04:21.01 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 04:21.08 | *** join/#asterisk sticks (n=bones@ip68-12-170-34.ok.ok.cox.net) |
| 04:21.16 | FuriousGeorge | anyone using snom phones here notice any problems with parking |
| 04:21.48 | FuriousGeorge | namely that they dont here what spot the call gets parked in on a blind tx and they get incomplete address when they direct dial? |
| 04:21.54 | *** part/#asterisk sticks (n=bones@ip68-12-170-34.ok.ok.cox.net) |
| 04:22.41 | dlynes_home | flujan: zap show status |
| 04:23.10 | *** join/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 04:23.32 | *** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 04:23.44 | n3glv | Hi guys |
| 04:23.50 | n3glv | konichi-wa |
| 04:23.57 | znoG | sayonara |
| 04:23.59 | dlynes_home | ni hao ma? |
| 04:24.04 | n3glv | quick question |
| 04:24.09 | flujan | dlynes_home, http://pastebin.ca/68829 |
| 04:24.22 | FuriousGeorge | ~lastspoke avi-bani |
| 04:24.25 | n3glv | I have two meetme conf's linked by transferring one to the other |
| 04:24.35 | FuriousGeorge | ~lastsponke av-bani |
| 04:24.36 | dlynes_home | ~seen avi-bani |
| 04:24.51 | jbot | i haven't seen 'avi-bani', dlynes_home |
| 04:24.51 | n3glv | how can I force a disconnect of a call (the conf link) |
| 04:24.51 | dlynes_home | ~seen av-bani |
| 04:24.52 | jbot | i haven't seen 'av-bani', dlynes_home |
| 04:25.14 | FuriousGeorge | ~seen avi-bani |
| 04:25.15 | jbot | FuriousGeorge: i haven't seen 'avi-bani' |
| 04:25.19 | flujan | dlynes_home, do you think this problem can occur because I'm using Slackware? This could be a distro problem? |
| 04:25.34 | flujan | dlynes_home, permissions or something? |
| 04:25.34 | dlynes_home | flujan: hell, no |
| 04:25.39 | dlynes_home | flujan: I use slackware, too |
| 04:25.55 | flujan | dlynes_home, cool... |
| 04:25.57 | dlynes_home | flujan: is there a call currently connected when you try to issue a call? |
| 04:26.15 | dlynes_home | flujan: actually, a good number of the participants in this channel are using Slackware |
| 04:26.25 | flujan | dlynes_home, :) |
| 04:26.51 | [TK]D-Fender | ~seen [av]bani |
| 04:26.52 | jbot | [av]bani <n=[av]bani@washuu.anime.net> was last seen on IRC in channel #asterisk, 66d 6h 1m 17s ago, saying: 'robin_sz: how are they?'. |
| 04:26.56 | dlynes_home | Even if slackware's a small %'age of the rest of the linux world, for whatever reason it's quite high in asterisk |
| 04:26.59 | FuriousGeorge | thats it |
| 04:27.04 | flujan | dlynes_home, I already tried a lot of things... and always have this problem... The guy who sold the card, tested the board in a centos an have no errors... |
| 04:27.10 | FuriousGeorge | he used snoms |
| 04:27.17 | [TK]D-Fender | Slacrware for the win! #1 in "just works" ! |
| 04:27.24 | FuriousGeorge | anyone using an snom and call parking around? |
| 04:27.37 | dlynes_home | [TK]D-Fender: yeah...but let's not saying anything about your uber typing skillz :p |
| 04:27.46 | flujan | [TK]D-Fender, |
| 04:27.54 | flujan | [TK]D-Fender, uncle Pat rulez |
| 04:27.57 | dlynes_home | flujan: how many lines do you have? |
| 04:27.59 | FuriousGeorge | noticing that i cant park a call right |
| 04:28.18 | flujan | dlynes_home, 30 channels in a Pri E1. |
| 04:28.54 | dlynes_home | flujan: can you pastebin your zapata.conf and zaptel.conf files? |
| 04:29.00 | dlynes_home | flujan: and lose all the commenting? |
| 04:29.51 | dlynes_home | heh...calgary's still getting snow and hail :) |
| 04:29.58 | flujan | dlynes_home, http://pastebin.ca/68833 |
| 04:29.59 | *** join/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 04:30.10 | flujan | dlynes_home, thanks for helping... :) |
| 04:30.44 | *** part/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 04:31.00 | dlynes_home | flujan: change bchan in zaptel.conf to bchan=1-15,17-31 |
| 04:31.03 | *** part/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 04:31.05 | dlynes_home | flujan: then shutdown asterisk |
| 04:31.12 | dlynes_home | flujan: and then type ztcfg -vvvvvvvvvvv |
| 04:31.18 | dlynes_home | flujan: and then reload asterisk |
| 04:31.39 | dlynes_home | flujan: and then let's see a log of you trying to dial |
| 04:33.08 | flujan | dlynes_home, |
| 04:33.11 | flujan | dlynes_home, http://pastebin.ca/68835 |
| 04:34.03 | flujan | dlynes_home, [TK]D-Fender http://pastebin.ca/68836 |
| 04:34.31 | Gamercjm | my garage door open when i dial an ext on my DID now :) |
| 04:34.41 | flujan | dlynes_home, so strange this error... isn't it? |
| 04:34.49 | Gamercjm | ive been so bored lol |
| 04:35.06 | dlynes_home | flujan: you need to Dial(Zap/g1/number) that's why |
| 04:35.12 | [TK]D-Fender | dlynes_home : SHUP YUO! |
| 04:35.19 | *** join/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 04:35.22 | dlynes_home | [TK]D-Fender: ? |
| 04:35.32 | FuriousGeorge | my parked calls context isnt supposed to have anything in it right? |
| 04:35.33 | [TK]D-Fender | <dlynes_home> [TK]D-Fender: yeah...but let's not saying anything about your uber typing skillz :p |
| 04:35.39 | dlynes_home | heh |
| 04:36.11 | FuriousGeorge | im looking at application park, says here its registered internally and doesnt need to be included in th DP |
| 04:36.32 | dlynes_home | FuriousGeorge: it is |
| 04:36.36 | FuriousGeorge | yet i keep getting address incomplete when i call my park extension, which is ** |
| 04:36.50 | flujan | dlynes_home, http://pastebin.ca/68838 |
| 04:37.08 | FuriousGeorge | and if i cant park myself, i cant at-xfer |
| 04:37.15 | FuriousGeorge | and so i cant use parking |
| 04:37.29 | FuriousGeorge | the wierd thing is that blind x-fer works |
| 04:37.32 | flujan | dlynes_home, I know this is strange |
| 04:37.50 | FuriousGeorge | but then of course i have to guess as to where the call went |
| 04:37.56 | dlynes_home | flujan: do you have the right signalling and that kinda thing? |
| 04:38.14 | flujan | dlynes_home, yeap... we are using ISDN here... |
| 04:38.21 | flujan | so... pri_cpe |
| 04:38.23 | flujan | :) |
| 04:38.27 | dlynes_home | flujan: yeah |
| 04:38.29 | mds2 | anyone know if cisco 7940/7960 phones can have DND on a per-line basis? |
| 04:38.33 | dlynes_home | flujan: but i meant the other stuff |
| 04:38.43 | flujan | dlynes_home, ? |
| 04:38.47 | Qwell | mds2: with the right firmware and channel driver, sure :p |
| 04:38.51 | dlynes_home | span=1,0,0,ccs,hdb3 |
| 04:38.59 | dlynes_home | and switchtype=euroisdn |
| 04:39.04 | dlynes_home | are those both correct? |
| 04:39.09 | mds2 | Qwell: is it documented somewhere? |
| 04:39.26 | Qwell | no.. |
| 04:39.31 | Qwell | You'll need to write code :p |
| 04:39.33 | mds2 | :) |
| 04:39.36 | mds2 | ah |
| 04:40.06 | mds2 | so it's not a feature of the cisco sip load? |
| 04:40.10 | Qwell | no |
| 04:40.13 | flujan | dlynes_home, just to make sure http://pastebin.ca/68840 |
| 04:40.21 | flujan | please, check my extensions.conf |
| 04:40.26 | mds2 | are there any opensource firmware efforts for 79xx? |
| 04:40.38 | Qwell | mds2: no |
| 04:41.03 | flujan | dlynes_home, yeap... at least is the configuration I see the guys using here in Brazil. |
| 04:41.27 | dlynes_home | flujan: turn off autofallthrough |
| 04:41.39 | mds2 | Qwell: you're suggesting I reverse engineer the phone, make a SIP load that has roughly the same feature set as Cisco's then add the per-line DND? |
| 04:42.00 | Qwell | mds2: no, I'm saying use skinny, and modify the asterisk channel driver to do what you want |
| 04:42.07 | dlynes_home | flujan: also after your dial command, add the following: exten => 5,n,Noop(${DIALSTATUS}) |
| 04:42.19 | dlynes_home | flujan: then call again, and paste the log |
| 04:43.30 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:43.37 | mds2 | Qwell: we're pretty well established on SIP. I thought the * Skinny code was pretty bare bones? |
| 04:43.56 | heison | anyone here used $AGI->verbose() ? |
| 04:44.39 | heison | i can no longer see characters sent to the console... last time i had it working was with Asterisk 1.0, I'm now on 1.2.7 |
| 04:44.50 | flujan | dlynes_home, http://pastebin.ca/68841 |
| 04:45.19 | flujan | dlynes_home, :'( |
| 04:46.23 | dlynes_home | flujan: hrm...not sure |
| 04:46.52 | dlynes_home | flujan: try using pri debug span 1 or whatever the command is, or pri intense debug to super duper debugging on the pri |
| 04:46.53 | flujan | dlynes_home, not sure about what? |
| 04:47.01 | dlynes_home | flujan: not sure about what th eproblem is |
| 04:47.11 | flujan | dlynes_home, me too... :P |
| 04:47.11 | dlynes_home | flujan: but i've gotta run....gotta install a new server |
| 04:49.23 | flujan | dlynes_home, http://pastebin.ca/68844 |
| 04:49.35 | flujan | after i use pri intense debug span 1q |
| 04:49.55 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 04:51.58 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-178-28.lsanca.fios.verizon.net) |
| 04:53.33 | flujan | dlynes_home, |
| 04:54.22 | flujan | dlynes_home, i dunno why nor how... but take a look: it works for a second... :) http://pastebin.ca/68846 |
| 04:54.38 | flujan | dlynes_home, we are almost there... at least I hope. :P |
| 04:56.04 | *** join/#asterisk lorinc (n=ang@caracas-4689.adsl.interware.hu) |
| 04:56.46 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
| 04:56.46 | *** join/#asterisk [hc] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 04:57.09 | phalacee | I was wondering if it is possible to use a 56k voice-modem in-place of a Digium (or equivalent) PSTN card ... |
| 04:57.21 | Qwell | phalacee: if you write zaptel drivers for it |
| 04:58.07 | [hc] | anyone have any suggestions for debugging/fixing dropped call problems when connecting to pstn using fxo devices (sangoma a200 in my case) ? |
| 04:58.42 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 04:59.32 | phalacee | oh fizzle-sticks ... I was hoping you'd either say yes, or a straight no ... cos my boss wants to set up a demo machine with a 56k modem in place of a PSTN ... |
| 04:59.49 | phalacee | any projects out there working on making generic drivers already? |
| 05:00.50 | dlynes_home | flujan: your d-chan is not functional (perhaps your driver didn't load correctly, check dmesg for details), and it looks like you might need a pridialplan and prilocaldialplan for your pri |
| 05:01.00 | dlynes_home | flujan: but i can't help you further...i'm heading out the door now |
| 05:01.03 | P-NuT | So, has anyone used SPA3000's with plain vanilla asterisk and NOT something like trixbox or something? |
| 05:01.11 | flujan | dlynes_home, thanks... |
| 05:01.12 | P-NuT | and is using it in AUS? |
| 05:01.18 | flujan | dlynes_home, I will check it |
| 05:01.34 | flujan | dlynes_home, thank you and have a good server install |
| 05:01.48 | Corydon76-home | phalacee: so you're going to spend $5,000 of billable time to write a driver, when you could just buy a card off ebay for $17? |
| 05:01.51 | SwK | p-nut haven't used them in .au area but have used them in the states and they are cool |
| 05:02.52 | phalacee | Corydon76-home: no, I'm going to do as my boss asks me to do, because he wants to demonstrate that Asterisk is capable of running on hardware you have lying around the house ... |
| 05:03.41 | Corydon76-home | capable, yes. The best idea, no. |
| 05:04.04 | Corydon76-home | For starters, the modem has to be full duplex, and most modems are only half duplex |
| 05:04.25 | P-NuT | SwK: Have you got the config parts for outgoing trunk to it, and to accept incoming calls? |
| 05:04.50 | *** join/#asterisk Cherebrum (n=jgarland@ares.jasongarland.com) |
| 05:04.56 | Cherebrum | look what I fond |
| 05:04.57 | Cherebrum | er found |
| 05:04.58 | Cherebrum | http://asteriskindy.com/media/asterisk.mov |
| 05:05.03 | phalacee | Corydon, is it feasible/possible to use two modems, and somehow split the work? |
| 05:05.22 | Cherebrum | it's a video of a bunch of people saying "Asterisk" here in Indianapolis |
| 05:05.31 | Cherebrum | but it has nothing to do with THIS asterisk |
| 05:05.39 | Cherebrum | Their logo even looks like the asterisk.org logo |
| 05:05.42 | Cherebrum | it's weird |
| 05:05.48 | SwK | p-nut well I'm not exactly sure of the "region" settings you need to adjust under advanced admin for aus telco's, but just register the FXO port like the FXS port and it will work just like the FXS side |
| 05:05.52 | Corydon76-home | phalacee: possible yes. Feasible, probably not. |
| 05:06.11 | SwK | p-nut: if its not registered it just bridges the FXO and FXS together for "failover pass thru" |
| 05:06.30 | SwK | p-nut: you might also want to check out this pdf http://www.jmgtechnology.com.au/spa_3000_guide.pdf |
| 05:06.40 | *** part/#asterisk Cherebrum (n=jgarland@ares.jasongarland.com) |
| 05:07.10 | SwK | p-nut that has the regional changes for aus support in it also |
| 05:08.26 | phalacee | Corydon: http://www.voip-info.org/wiki/view/X100P+clone |
| 05:09.01 | akant2 | Does anyone here have asterisk running with the provider quantum voice? |
| 05:09.29 | phalacee | Corydon: That page mentions that the Digium X100P FXO card is a cloned Intel v92 fav/data/voice modem |
| 05:10.34 | Corydon76-home | phalacee: yes, it is |
| 05:10.53 | phalacee | So I could just use one of those modems, and it would work? |
| 05:10.53 | Corydon76-home | phalacee: but as I said before, it's full duplex |
| 05:11.32 | Corydon76-home | It has to be that specific Intel chipset |
| 05:12.11 | phalacee | see the thing i am getting at here, is that my boss doesn't really care if its a specific modem that works, just so long as a 56k modem will work ... |
| 05:12.29 | Corydon76-home | So get some X100P's off ebay |
| 05:12.32 | phalacee | he doesn't care if it costs $25 for the modem, compared to $17 for the PSTN card ... |
| 05:12.53 | Corydon76-home | It's still just analog crap |
| 05:13.24 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 05:13.50 | TESTER2 | someone has festival installed and working OK? |
| 05:14.01 | Corydon76-home | phalacee: as you quite correctly pointed out, the X100P is a modem. So why not just get something that you know will work, rather than screwing around with cards that may or may not work? |
| 05:14.38 | phalacee | **nods** I intend to, I'm looking it up on ebay now |
| 05:15.41 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 05:17.17 | *** join/#asterisk erwinism (i=erwin@61.9.118.37) |
| 05:17.55 | erwinism | hello, do i need to compile asterisk so i can use speex codec? |
| 05:18.18 | Qwell | erwinism: I don't think I understand the question.. |
| 05:18.20 | P-NuT | SwK: I just want to use the SPA as a PSTN gateway. |
| 05:18.22 | P-NuT | That's all. |
| 05:19.12 | erwinism | Qwell, i have my asterisk working right now. I want to use the speex codec on it |
| 05:20.32 | P-NuT | and is it easier to do it through plain ol' asterisk? Or something like trixbox? |
| 05:21.03 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 05:21.49 | akant2 | What would cause a "metalic", light jitter sound when accessing music on hold? I just compiled Asterisk, and compiled mpg123 or whatever.. am I missing a piece? |
| 05:23.56 | drray | do you have ztdummy? |
| 05:24.07 | drray | or timing form a zaptel card? |
| 05:24.17 | akant2 | no |
| 05:24.45 | akant2 | all I have no now is a PAP2 with my analog phone |
| 05:24.45 | flujan | guys, I'm having the following error: http://pastebin.ca/68855 |
| 05:24.45 | akant2 | connected to the box |
| 05:24.49 | flujan | no d-channel available... |
| 05:24.59 | akant2 | and am testing by dialing an extension to access music on hold |
| 05:25.13 | akant2 | do I need to have ztdummy if I dont use a zaptel card? |
| 05:25.13 | flujan | to make calls, i make just a call the phone ring once and stop |
| 05:25.30 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 05:25.52 | flujan | then, to have asterisk dialing again I have to use ztcfg -v and restart asterisk |
| 05:26.34 | drray | akant2 - I'm not sure actualyl |
| 05:26.40 | drray | but that is where I would start |
| 05:26.43 | akant2 | ok |
| 05:27.02 | akant2 | last time I had this working I DID have a zaptel card for an analog line, perhaps it was timing off of tht |
| 05:27.13 | akant2 | Ill look at ztdummy and see |
| 05:27.33 | akant2 | any question, about t1 cards |
| 05:27.52 | drray | It may also be required with Music on Hold, i.e. to improve sound quality. |
| 05:27.59 | akant2 | I need to get one for testing.. and there cheaper alternatives to Digium? |
| 05:28.02 | drray | http://www.voip-info.org/wiki/view/Asterisk+timer |
| 05:28.11 | drray | akant2 - govarion.com |
| 05:28.25 | akant2 | awesome |
| 05:28.25 | drray | tehy ahve a 4 span tor2 card for $600 ish |
| 05:28.30 | drray | but they suck |
| 05:28.32 | akant2 | eeek |
| 05:28.37 | akant2 | lol |
| 05:28.43 | akant2 | I need a sing T card :) |
| 05:28.53 | drray | I use a govarion card for a property and like it |
| 05:28.58 | akant2 | no 100XP T1 cards :) |
| 05:28.59 | akant2 | lol |
| 05:31.24 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 05:40.22 | erwinism | what port the SIP uses? |
| 05:41.28 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 05:44.37 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.65) |
| 05:50.53 | *** join/#asterisk _omer (n=omer@gw3-fiberclient-37.brain.net.pk) |
| 05:52.28 | _omer | hello |
| 05:52.34 | _omer | any one? http://pastebin.com/724808 |
| 05:53.22 | *** join/#asterisk joelsolanki (n=jnsolank@202.160.161.94) |
| 05:55.59 | drray | pastebin is lagging for me |
| 05:56.11 | Qwell | drray: the .com tends to do that |
| 05:57.33 | _omer | drray ... its lagging for me too |
| 05:59.42 | joelsolanki | hi all |
| 06:00.00 | joelsolanki | i m using ser+asterisk from last 5 months without any problem. |
| 06:00.08 | joelsolanki | now i want to make following setup. |
| 06:01.07 | joelsolanki | users on ser should able to save / retrieve voicmail from asterisk. |
| 06:01.18 | _omer | any one? http://pastebin.com/724808 |
| 06:01.20 | joelsolanki | I heard this is possible but dont have idea. how ? |
| 06:02.52 | joelsolanki | any one ? |
| 06:08.22 | jmacz | Hi everyone |
| 06:08.49 | jmacz | I have a question regarding the b option of application MixMonitor |
| 06:09.27 | jmacz | I'm making some tests in voice recordings and have tried all the options (v(x), V(x), W(x), a and b) |
| 06:09.32 | P-NuT | Hi all, getting an outbound call out of the SPA3000 PSTN line.. |
| 06:09.42 | P-NuT | what do I have to set on the SPA to make that happen? |
| 06:09.48 | jmacz | all of them make sense to me, except "b" |
| 06:10.12 | jmacz | the show application MixMonitor shows the following: b - Only save audio to the file while the channel is bridged. |
| 06:10.12 | jmacz | <PROTECTED> |
| 06:10.30 | jmacz | sorry but english is not my native language |
| 06:10.49 | Pegasus_Epsilon | jmacz: what behavior are you trying to get |
| 06:10.50 | jmacz | what does exactly means "while the channel is bridged"? |
| 06:11.32 | jmacz | Pegasus_Epsilon, I want to record all the conversations made by some extensions through a PRI |
| 06:11.46 | jmacz | and put them into a single file |
| 06:11.47 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 06:11.53 | jmacz | option A is enough for this |
| 06:12.29 | Pegasus_Epsilon | i presume you have proper authorization or a warrant |
| 06:12.35 | jmacz | however, if I use or not the b option (I'm testing between 2 extensions first), doesn't make any difference :s |
| 06:12.49 | Pegasus_Epsilon | or maybe don't need one whereever you are |
| 06:13.19 | jmacz | Pegasus_Epsilon, of course, it's fot a Call Center and we will put a warning that "your call may be monitored for quality porpouses bla bla" |
| 06:13.49 | drray | yes the words "call center" and "quality"... |
| 06:13.51 | Qwell | porpoises? |
| 06:13.52 | jmacz | the point is I don't get what this b option is for |
| 06:13.57 | Pegasus_Epsilon | by "bridged" in that case i'm guessing it means when you have two extensions talking to one PSTN, or two PSTN lines talking to one extension |
| 06:14.13 | Pegasus_Epsilon | Qwell: ESL, he's doing well enough |
| 06:14.23 | rainkid | what is the proper way to configure asterisk to allow other extensions to pickup a call on hold? (without doing a transfer) |
| 06:14.30 | jmacz | drray, know what you mean :'( |
| 06:14.56 | drray | I actually enjoyed my time in the call center mines |
| 06:15.08 | Pegasus_Epsilon | jmacz: you probably don't want or need the b option for what you're doing |
| 06:15.26 | Pegasus_Epsilon | if it works without it, leave it off, if it doesn't work either way, you're having another problem |
| 06:15.41 | jmacz | Pegasus_Epsilon, you mean like when one transfers a call from the pstn to another extension? |
| 06:16.02 | Pegasus_Epsilon | jmacz: no, i mean when one person connects to a call in progress |
| 06:16.10 | Pegasus_Epsilon | so you have a three-way on the asterisk box itself |
| 06:16.22 | jmacz | drray, do you? :s |
| 06:16.27 | Pegasus_Epsilon | "operator break-in" |
| 06:16.51 | Pegasus_Epsilon | what i used to have the operator do to my girlfriend in highschool :b |
| 06:17.04 | Pegasus_Epsilon | sucks that they won't do that anymore |
| 06:17.44 | jmacz | he he he |
| 06:17.45 | Pegasus_Epsilon | operators used to be so willing to help you screw with people |
| 06:18.25 | Pegasus_Epsilon | "the number NNN-NNN-NNNN is currently in a call, can you tell me who they're talking to?" "sure, that's NNN-NNN-NNNN, do you need anything else?" "can you connect me to that call?" "sure, just a moment" |
| 06:18.42 | Pegasus_Epsilon | not anymore. so sad. |
| 06:18.47 | drray | back before they started selling call waiting |
| 06:18.52 | drray | and other "plus" services |
| 06:19.02 | jmacz | Pegasus_Epsilon, unfortunely yes :s |
| 06:19.10 | drray | I used to have the operator break the line so I could get on to BBS's |
| 06:19.19 | Pegasus_Epsilon | they did that for a while after call waiting, too, then they realized that the FCC had rules about that sort of thing, and knowing that it could be done doesn't make you a telco employee |
| 06:19.31 | jmacz | Pegasus_Epsilon, I get the point, thanks a lot for your explanation :) |
| 06:19.45 | Pegasus_Epsilon | no problem, jmacz |
| 06:20.37 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 06:21.22 | jmacz | Actually, I have only tried call waitting. I'd farly heard about 3-way calling but never needed to use it. |
| 06:21.37 | jmacz | Can't believe FCC has rules for it :s |
| 06:23.28 | drray | that was before the phone company was converted to a free market system |
| 06:23.41 | drray | ok, I know, it really wasn't |
| 06:23.52 | *** join/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 06:26.33 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
| 06:27.45 | jmacz | drray, tell me about it... |
| 06:28.30 | jmacz | one of the main "Public" Telcos in my country was sold this year to Telefonica (Spain) |
| 06:28.56 | jmacz | and there's more to come |
| 06:29.54 | dudes | what country? |
| 06:30.06 | jmacz | dudes, Colombia |
| 06:30.13 | jmacz | however, I'm really willing for the approval of less restrictive laws regarding VoIP |
| 06:30.41 | dudes | spain is less restrictive than Colombia? |
| 06:31.03 | jmacz | an advantage of Free Market is how it pushes on over this kind of demands of tech |
| 06:31.23 | jmacz | dudes, I really can't tell |
| 06:31.39 | dudes | So Columbia isn't capitialism? |
| 06:32.11 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 06:32.13 | dudes | err, capitalism |
| 06:32.32 | jmacz | what I was meaning is that Free Market makes things like Statal companies sold to private ones more likely |
| 06:33.01 | jmacz | dudes, Colombia is maybe the most capitalism country of South America at this time |
| 06:33.35 | drray | free markets don't build roads |
| 06:33.36 | dudes | right on |
| 06:33.44 | dudes | taxes do though |
| 06:33.45 | dudes | =[ |
| 06:33.48 | dudes | err, =p |
| 06:33.52 | drray | :) |
| 06:33.56 | jmacz | yep... |
| 06:34.28 | dudes | the difference in roads from South Dakota to Minnesota is insane |
| 06:34.36 | dudes | so you know which one has States taxes |
| 06:35.02 | jmacz | sadly we don't see our taxes on roads fixed and new ones build but buying weapons to sustain the internal war |
| 06:35.10 | Pegasus_Epsilon | dudes: SD can afford it |
| 06:35.21 | Pegasus_Epsilon | think about the geography, man |
| 06:35.25 | Pegasus_Epsilon | black hills vs lakes |
| 06:35.30 | dudes | maybe they can, however, their roads still suck |
| 06:35.40 | jmacz | I guess the US has a similar problem right? |
| 06:35.41 | Pegasus_Epsilon | oh, you're saying MN has better roads |
| 06:35.45 | Pegasus_Epsilon | nevermind, argument fails |
| 06:35.49 | dudes | hehe |
| 06:36.21 | jmacz | a lot of money invested in war, less in roads, health care, etc |
| 06:36.23 | dudes | I was driving to the casino for smokes the other day, and their was pot-holes six inches deep in about 15ft long |
| 06:36.40 | dudes | I doubt the war effort cobtributes to bad roads |
| 06:36.49 | jmacz | :s |
| 06:36.56 | drray | the war effort is self funding |
| 06:36.59 | dudes | the federal government only gives so much for funding of roads |
| 06:37.18 | drray | and there is a profit to it, whereas roads only get worse. |
| 06:37.21 | dudes | South Dakota doesn't have sales tax nor does it tax on cigarettes |
| 06:37.31 | jmacz | drray, that's very true |
| 06:37.44 | drray | nor does South Dakota have any sort of non gorvenment jobs |
| 06:37.57 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 06:38.02 | x86 | anyone around speak or understand spoken Urdu? |
| 06:38.05 | dudes | Minnesota doesn't have the issue with roads, but we have a 7% sales tax |
| 06:38.18 | x86 | dudes: you're lucky, here it's 8% |
| 06:38.20 | dudes | rather 6.5% rounded up |
| 06:38.29 | FuriousGeorge | anyone tried to patch 1.2.7.1 source with oel's meetermaid patch |
| 06:38.42 | drray | Texas (where I am from, originally) pulled a fast one and made most roads federal highways.. so they don't have to pay for them |
| 06:38.45 | FuriousGeorge | i get 1 out of 1 hunk failing |
| 06:38.45 | drray | hail LBJ |
| 06:38.46 | neilbags-work | 10% here, but it used to be 22% |
| 06:38.59 | jmacz | Is the "sales tax" the one applied to most of the things you sell or buy? |
| 06:39.16 | jmacz | Here -> 16% |
| 06:39.16 | drray | jmacz - sales tax is a consumption tax |
| 06:39.19 | dudes | when you buy certain items their is a tax |
| 06:39.21 | drray | similar to a vat |
| 06:39.26 | dudes | food in minnesota is extempt |
| 06:39.43 | neilbags-work | here in australia, its on anything that is not an 'essential' |
| 06:39.56 | dudes | beer, candy, and cigarettes are extra |
| 06:39.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 06:40.17 | dudes | but since i don't buy candy I'm safe on that tax =p |
| 06:40.29 | neilbags-work | some foods here are exempt, mostly fresh foods and such |
| 06:40.40 | neilbags-work | anything that the democrats don't think is a luxury |
| 06:40.50 | dudes | so true |
| 06:41.00 | drray | problem is sales taxes are regressive |
| 06:41.15 | jmacz | here even the food has that 16% tax (as well as the tooth paste, the soap, and almost all first need elements :s) |
| 06:41.19 | dudes | Minnesota has more taxes now than when Ventura was in office but that guy kind of screwed us with his refund that broke us |
| 06:41.43 | neilbags-work | drray: i don't think its regressive here |
| 06:41.44 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:42.31 | drray | regressive meaning that poor people pay more taxes than rich people when it is tied to consumption. |
| 06:42.39 | dudes | I'm glad a repub is in office here. Pawlenty has been doing a good job bring our economy back as well as ensuring border security |
| 06:42.53 | neilbags-work | drray: oh i missed your meaning sorry |
| 06:42.56 | jmacz | Here's regressive only when you are a Company of the "common" regime and you sell generating that tax |
| 06:43.16 | jmacz | almost all companies are part of the common regime, of course |
| 06:43.29 | drray | jnacz :) |
| 06:43.37 | dudes | I don't believe the poor man pays more than rich |
| 06:44.03 | jmacz | well, that's how it works in some places |
| 06:44.04 | drray | the poor carry a higher burden, percentage wise |
| 06:44.34 | drray | and poor people are less likely to be able to avoid taxes |
| 06:44.48 | jmacz | maybe the poor doesn't actually "pays more" but keeping the propotions, they do |
| 06:44.49 | dudes | you pay more the more you make |
| 06:44.49 | neilbags-work | well i guess here the democrats poked their heads in and got a whole bunch of things exempt in an effort to minimise the regression |
| 06:45.06 | jmacz | *proportion |
| 06:45.06 | drray | dudes - I was speaking of consumption (sales taxes) |
| 06:45.40 | dudes | maybe in sales taxes, but figure the property taxes as well as other taxes they pay via ownerhsip |
| 06:45.52 | drray | dudes - no doubt about that |
| 06:45.59 | jmacz | of course |
| 06:46.03 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:46.04 | drray | and they absorb the cost of renting |
| 06:46.04 | drray | etc |
| 06:46.41 | dudes | like dubya made the latter end of 600k last year and paid 187k in fed faxes (not including state, local, and such) |
| 06:47.01 | FuriousGeorge | anyway, as i was saying, im trying to apply the meetermaid patch to the asterisk 1.2.7.1 source, i got a file called meetermaid-1.2.7.1, used the normal patch syntax, and my hunk failed. how is that possible |
| 06:47.31 | dudes | maybe the file wasn't clean? |
| 06:47.36 | jmacz | what I said is that taxes like the "Added Value Tax" (16%) that we have here for sells, tends to hit more the poor people than the middle class or the rich |
| 06:47.55 | FuriousGeorge | dudes: i got it from asterisk bug tracker |
| 06:48.04 | dudes | I can see how it could affect the poor more than the upper classes |
| 06:48.24 | drray | and by poor I meant poverty line poor |
| 06:48.30 | drray | 20k family of 4 |
| 06:48.36 | drray | not 30k |
| 06:49.06 | dlynes_home | FuriousGeorge: just cause you got it on asterisk bug tracker doesn't mean it was clean |
| 06:49.09 | FuriousGeorge | anytax affects the poor more than the rich. 7% of a given item is a larger percentage of a poor man's total salary than a rich man's |
| 06:49.24 | FuriousGeorge | dlynes_home: i suppose anything is possible |
| 06:49.25 | dudes | My mom of a single mom of 3 working at hardees |
| 06:49.28 | FuriousGeorge | what are the odds you think though |
| 06:49.45 | dlynes_home | FuriousGeorge: which issue? |
| 06:50.06 | dudes | FuriousGeorge - have you tried appling the patch manually? |
| 06:50.24 | FuriousGeorge | there are two files one says meetermaid-v3.txt the other is meetermaid-1.2.7.1.txt |
| 06:50.44 | dudes | you have the "+" "-" and the find lines so it's not too difficult to make sense of them |
| 06:51.04 | FuriousGeorge | dudes: my knowledge of C is pretty redimentary |
| 06:51.09 | *** join/#asterisk mover (n=dlu@83.125.8.7) |
| 06:51.21 | FuriousGeorge | i see what you mean |
| 06:51.23 | mover | ola |
| 06:51.26 | dudes | like I said, their is a portion of code to follow |
| 06:51.29 | dudes | it's not too hard |
| 06:52.02 | *** part/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 06:52.30 | dudes | did you do "patch -p1 patch.txt < patch.c ??? |
| 06:52.51 | mover | anyone noticed about an inbound issue to new Nokia E60 Series in cause of missing stun support in this voip UA? |
| 06:52.52 | FuriousGeorge | err, i did a p0 |
| 06:52.55 | *** join/#asterisk dant (n=dan@2001:618:400:3f8d:204:76ff:fe1e:585e) |
| 06:53.04 | FuriousGeorge | dudes: ^ |
| 06:53.13 | FuriousGeorge | following the wiki |
| 06:53.19 | dlynes_home | ~book |
| 06:53.28 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:53.28 | dlynes_home | ~thebook |
| 06:53.31 | dudes | I don't know if p0 or p1 make a diff I just use p1 myself |
| 06:53.37 | Qwell | dudes: yes |
| 06:54.39 | dudes | what's going on Qwell |
| 06:55.02 | Qwell | about to head to bed |
| 06:55.02 | FuriousGeorge | dudes: when i use p1 it asks what file to patch |
| 06:55.27 | dudes | type the file and hit enter |
| 06:55.34 | dudes | dir includes |
| 06:55.53 | dlynes_home | ~mailinglist |
| 06:55.55 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 06:56.35 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 06:58.55 | FuriousGeorge | dudes: i think its features.c i wanna patch for meetermaid no? |
| 06:59.14 | dudes | I'm not sure what you're trying to patch |
| 06:59.21 | dudes | it should say in the patch |
| 07:02.37 | FuriousGeorge | http://pastebin.ca/68879 |
| 07:02.48 | FuriousGeorge | it says its failing to patch res_channels and res_features |
| 07:03.11 | dudes | paste the patch |
| 07:03.35 | dudes | or provide a link to the patch |
| 07:03.46 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 07:04.16 | FuriousGeorge | http://bugs.digium.com/view.php?id=5779 |
| 07:05.35 | dudes | the first is, res/res_features.c |
| 07:05.47 | dudes | second is channels/chan_local.c |
| 07:06.07 | dudes | third, include/asterisk/features.h |
| 07:06.16 | FuriousGeorge | yeah, i tried pointing it to both of those files individually |
| 07:06.24 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:06.25 | FuriousGeorge | its just not gonna happen, i dont think |
| 07:06.37 | dudes | see if it made a backup |
| 07:06.49 | FuriousGeorge | it made a .orig |
| 07:06.53 | FuriousGeorge | and a .rej |
| 07:07.03 | dudes | cp the orig in place of the current |
| 07:08.05 | dudes | then patch -p1 patch.txt |
| 07:08.16 | dudes | then choose the file in order |
| 07:08.19 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-215-217.telkomadsl.co.za) |
| 07:08.41 | *** join/#asterisk LH-euhost (n=LH-euhos@62.77.207.166.pool.invitel.hu) |
| 07:08.49 | *** join/#asterisk Libila (n=vye@ip68-6-130-59.sd.sd.cox.net) |
| 07:09.39 | Libila | I'm using a TDM04 digium card. When I'm on the phone I can hear "clicking" and static sounding noises in the background. How would I go about getting rid of that? |
| 07:10.10 | FuriousGeorge | dudes: dont you need a < in there after the p1 |
| 07:10.34 | FuriousGeorge | and before the .txt file |
| 07:10.47 | dudes | patch -p1 patch.txt |
| 07:10.56 | dudes | then enter the paths in order |
| 07:12.05 | FuriousGeorge | dudes: when i do it without the < i just get a new line. i can type the path to those three files one after another but i got a feeling somethings wrong... trying |
| 07:12.25 | dudes | don't have the < |
| 07:12.36 | dudes | it'll ask you for the file to use |
| 07:13.12 | FuriousGeorge | claudia asterisk-1.2.9.1 # patch -p1 metermaid-1.2.7.1.txt |
| 07:13.16 | FuriousGeorge | thats what happens |
| 07:13.18 | FuriousGeorge | thats it |
| 07:13.28 | FuriousGeorge | till i hit ctrl+c it sits there |
| 07:13.44 | FuriousGeorge | i can enter as many files as i want and it just gives me a new line every time i hit enter |
| 07:13.45 | dudes | well try with < to res/res_featues.c |
| 07:13.52 | dudes | make sure the file isn't NULL |
| 07:15.36 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 07:16.14 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:19.55 | FuriousGeorge | doesnt matter if i specify the file or not, hunks fail left and right |
| 07:20.14 | FuriousGeorge | even the one hunk thats supposed to patch features.conf.sample |
| 07:20.28 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 07:21.24 | dudes | I'd read the patch and search for the lines |
| 07:21.44 | dudes | otherwise I can't really help you since I don't use that patch and I'm pissed up |
| 07:22.07 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 07:22.10 | *** join/#asterisk JT (n=jon@CPE-138-130-2-5.nsw.bigpond.net.au) |
| 07:22.30 | FuriousGeorge | last thing i wanna do is manually start inserting these lines into code i dont really understand and seeing if it compiles |
| 07:22.42 | FuriousGeorge | i have a feeling if the patch were gonna work it would have worked already |
| 07:22.50 | FuriousGeorge | if oel were around i could ask him |
| 07:22.57 | *** join/#asterisk flujan (n=flujan@201-27-88-182.dsl.telesp.net.br) |
| 07:24.02 | *** part/#asterisk flujan (n=flujan@201-27-88-182.dsl.telesp.net.br) |
| 07:24.08 | dlynes_home | FuriousGeorge: that was 1.2 branch, not 1.2.7.1 |
| 07:24.16 | JT | anyone got some tips as to the cheapest FXS cards out there? |
| 07:24.19 | dlynes_home | FuriousGeorge: so it was against the code that was to become 1.2.8 |
| 07:24.48 | dlynes_home | JT: grandstream ata |
| 07:25.07 | JT | sorry, i should refine that query |
| 07:25.13 | JT | cheapest PCI based FXS cards |
| 07:25.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:25.33 | dlynes_home | JT: well, if you want pci specifically, and an external sip device won't do, you're not going to get cheap |
| 07:25.54 | JT | it's for the app_rpt module |
| 07:26.05 | JT | which as far as i know only accepts pci |
| 07:26.09 | dlynes_home | JT: you've got a choice of digium, digium clones, and sangoma |
| 07:26.25 | dlynes_home | app_rpt? you mean for alarm systems? |
| 07:26.30 | FuriousGeorge | oh shoot i got asterisk 1.2.9.1 and i want 1.2.7.1 |
| 07:26.42 | JT | nah, 2-way radio |
| 07:26.50 | JT | http://app-rpt.qrvc.com/ |
| 07:26.52 | dlynes_home | ah |
| 07:26.59 | FuriousGeorge | and of course svn checkout http://svn.digium.com/svn/asterisk/branches/1.2.7.1 doesnt work |
| 07:27.18 | JT | digium clones... what's out there? |
| 07:27.19 | s0lid | FuriousGeorge: you can download it on digiums ftp |
| 07:27.25 | dlynes_home | FuriousGeorge: ncftpget ftp://ftp.digium.com/pub/telephony/asterisk/asterisk-1.2.7.1.tar.gz |
| 07:27.38 | dlynes_home | JT: check out fleabay |
| 07:27.40 | s0lid | i have a problem with tdm2400p |
| 07:27.46 | s0lid | i have 20 fxs and 4fxo |
| 07:27.53 | s0lid | the problem is with 4 fxo |
| 07:27.59 | JT | ebay? |
| 07:28.03 | dlynes_home | yeah |
| 07:28.16 | dlynes_home | in my mind |
| 07:28.18 | s0lid | has anyone tried using the 2400? |
| 07:28.19 | JT | couldn't see much PCI FXS stuff when i last checked |
| 07:28.22 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 07:28.36 | s0lid | you can't find pci fxs nowadays |
| 07:28.40 | dlynes_home | it's not worth saving the money on the clones, when you probably get little or no after purchase support |
| 07:28.41 | s0lid | if you find one it's expensive |
| 07:28.50 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 07:28.57 | s0lid | just buy a tdm400p with fxs modules |
| 07:29.01 | FuriousGeorge | grr that link crashed kopete |
| 07:29.13 | JT | openvox.com.cn looks alright |
| 07:29.15 | dlynes_home | s0lid: you can get tdm400's with fxs modules, too |
| 07:29.17 | JT | 1yr warranty |
| 07:29.19 | FuriousGeorge | im having an opensource moment |
| 07:29.19 | JT | 30days support |
| 07:29.26 | s0lid | dlynes_home: why? |
| 07:29.31 | s0lid | i got mine with 2 fxs and 2 fxo |
| 07:29.37 | *** join/#asterisk Eggplant (i=No@dsl-72-19-44-253.cascadeaccess.com) |
| 07:29.49 | dlynes_home | s0lid: cause a tdm2400 costs a hell of a lot more than a tdm400? |
| 07:30.00 | s0lid | no |
| 07:30.08 | dlynes_home | s0lid: same price? |
| 07:30.08 | s0lid | dlynes_home: i have a tdm400p too |
| 07:30.13 | s0lid | cheaper |
| 07:30.17 | s0lid | in voipsupply |
| 07:30.22 | dlynes_home | really? |
| 07:30.22 | dlynes_home | damn |
| 07:30.24 | s0lid | voipsupply.com |
| 07:30.40 | s0lid | well that was 6 months ago i'll check just a sec |
| 07:30.41 | JT | but yeah |
| 07:30.56 | JT | app_rpt neads 2 FXS ports for every radio channel |
| 07:31.01 | JT | which makes it not cheap |
| 07:31.15 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:31.28 | JT | or you can buy their quad port radio interface cards, but they're only viable if you're actually running a few channels |
| 07:31.39 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:32.13 | s0lid | check it out here fxs modules |
| 07:32.14 | s0lid | http://www.voipsupply.com/index.php?cPath=99_103 |
| 07:33.00 | s0lid | $140 for the tdm400 with 1 fxs but at least you still have 3 modules you can use |
| 07:33.17 | s0lid | anyone experience using tdm2400p? |
| 07:33.23 | FuriousGeorge | i can use asterisk 1.2.7.1 with zaptel 1.2.6 and libpri 1.2.3 right |
| 07:34.32 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:35.04 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 07:36.23 | dlynes_home | s0lid: i've helped troubleshoot them, but never used one |
| 07:37.28 | JT | s0lid: thanks for that |
| 07:37.30 | JT | nice prices |
| 07:37.38 | FuriousGeorge | there are so many damn patches for asterisk, using one will generally prevent you from using another right? the line numbers will be all messed up |
| 07:37.41 | s0lid | no problem |
| 07:37.43 | JT | i will need a minimum of 2 FXS modules for my application |
| 07:38.00 | JT | duplicated by 2, for 2 different sites |
| 07:38.12 | s0lid | dlynes_home: you've troubleshoot a 2400p? |
| 07:39.18 | JT | openvox is about $100 cheaper for a quad FXS config, for what it's worth |
| 07:39.26 | JT | but i guess it has to be, to be a viable clne |
| 07:39.28 | JT | clone |
| 07:40.13 | s0lid | but that's a clone |
| 07:40.17 | s0lid | yes it will be cheaper |
| 07:40.30 | s0lid | but you'll take out digiums support |
| 07:40.32 | dlynes_home | s0lid: yeah...a couople of times on here when peeps have had troubles |
| 07:40.34 | JT | i wonder if there's cheaper still |
| 07:40.39 | s0lid | from buying openvox |
| 07:40.44 | JT | because remember i'm not connecting phones |
| 07:40.50 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 07:41.02 | FuriousGeorge | goddamnit, 12 of 12 hunks failed again. |
| 07:41.07 | s0lid | dlynes_home: i'm having a problem with my fxo quad module |
| 07:41.22 | dlynes_home | s0lid: what do you get from dmesg? |
| 07:41.39 | s0lid | dlynes_home: it's detected by the system and i configured it right but i get circuit-busy when i call it |
| 07:41.42 | s0lid | dlynes_home: i'll check that |
| 07:42.42 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
| 07:43.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:44.07 | QuAtRo[NL] | I'm trying to call from a dutch landline to a voipbuster account configured in my Asterisk |
| 07:44.39 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 07:44.52 | QuAtRo[NL] | But i get an error like: chan_sip.c: Auto destroying call '5b2e46f203bef9824256a93126b16b6a@127.0.0.1' |
| 07:45.24 | FuriousGeorge | this is all too frustrating. has anyone every applied meetermaid patch, or just happens to know syntax for the patch command it wants |
| 07:46.01 | QuAtRo[NL] | Full error is: Jun 22 08:40:33 DEBUG[6191] chan_sip.c: Setting NAT on RTP to 524288 |
| 07:46.01 | QuAtRo[NL] | Jun 22 08:40:33 DEBUG[6191] chan_sip.c: Stopping retransmission on 'de09d71c26004376b61b796f0817039b' of Response 14: Match Found |
| 07:47.20 | QuAtRo[NL] | Does someone know what to do? I already tried google and asteriskguru.com |
| 07:48.35 | dlynes_home | QuAtRo[NL]: that's not an error |
| 07:48.51 | dlynes_home | FuriousGeorge: patch < filename |
| 07:50.21 | FuriousGeorge | so it lists configs/features.conf.sample twice, then asks me file to patch, so i type in configs/features.conf... |
| 07:51.31 | FuriousGeorge | Skipping patch. |
| 07:51.31 | FuriousGeorge | 1 out of 1 hunk ignored |
| 07:51.31 | FuriousGeorge | can't find file to patch at input line 23 |
| 07:51.31 | FuriousGeorge | Perhaps you should have used the -p or --strip option? |
| 07:51.31 | FuriousGeorge | The text leading up to this was: |
| 07:51.38 | FuriousGeorge | etc |
| 07:52.00 | *** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:52.05 | FuriousGeorge | at this point i ctrl c b/c i know its not gonna get the other 3 files right either |
| 07:53.44 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 07:54.16 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 07:54.38 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 07:55.13 | dlynes_home | FuriousGeorge: maybe what you're looking at already has the patch applied? |
| 07:57.25 | FuriousGeorge | i can always make samples and tell right away |
| 07:57.45 | FuriousGeorge | i dont even have to do that, hold on |
| 07:58.59 | FuriousGeorge | no its totally not |
| 08:00.31 | dlynes_home | one sec |
| 08:01.55 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-182-66.dsl.pltn13.sbcglobal.net) |
| 08:05.14 | darkgamer20 | I am a little confused about asterisk and what it can do. one thing I am not clear about is if I can use my existing AT&T phone line with Asterisk and still have all that great functionality like autoattendant and stuff, without switching to a VoIP phone service? if that is possible how can I go about doing that? can someone give me advice or maybe a tutorial or guide to follow? |
| 08:05.24 | dlynes_home | FuriousGeorge: it'll only work on the cvs version |
| 08:05.32 | dlynes_home | FuriousGeorge: not on the ftp downloaded version |
| 08:05.51 | dlynes_home | FuriousGeorge: you can still go through it manually though, and apply the patches yourself to 1.2.9.1 or 1.2.7.1 |
| 08:06.17 | FuriousGeorge | svn version? |
| 08:06.34 | FuriousGeorge | you mean svn right? |
| 08:07.34 | FuriousGeorge | dlynes_home: ? |
| 08:07.39 | darkgamer20 | I dont want to be annoying but is there anything you guys know that can help me out? |
| 08:08.09 | FuriousGeorge | darkgamer20: asterisk does voip all by itself |
| 08:08.17 | FuriousGeorge | if you want to interface with a phone line |
| 08:08.23 | FuriousGeorge | which it soulds like you do |
| 08:08.28 | FuriousGeorge | you need special hardware |
| 08:08.53 | darkgamer20 | FuriousGeorge: what do you mean special hardware? |
| 08:09.38 | FuriousGeorge | there are devices that sit on your lan and are called ATAs (analog telephone adapters) or there are cards that go into your computer that you put modules on for phone company lines, or regular analog phones (fxo/fxs) |
| 08:09.43 | FuriousGeorge | ~fxofxs |
| 08:09.44 | jbot | [fxofxs] An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 08:10.03 | FuriousGeorge | ~docs |
| 08:10.04 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 08:11.25 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 08:11.39 | darkgamer20 | thanks FuriousGeorge, but one thing instead of an ATA cant i use a computer modem since that connects the computer and the phone too? |
| 08:12.10 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 08:12.19 | darkgamer20 | or is that a different function? |
| 08:12.26 | FuriousGeorge | darkgamer20: there is a modem that will work with asterisk as an FXO |
| 08:12.31 | FuriousGeorge | but ymmv |
| 08:12.49 | FuriousGeorge | i say drop the extra 50 bucks and get a real ata |
| 08:12.54 | FuriousGeorge | or tdm with an fxo module |
| 08:13.02 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
| 08:13.34 | darkgamer20 | im sorry but whats a TDM? |
| 08:13.57 | FuriousGeorge | dlynes_home: you meant svn version not cvs version, right? if so shouldnt svn checkout http://svn.digium.com/svn/asterisk/branches/1.2.7.1 asterisk acheive that |
| 08:14.17 | FuriousGeorge | darkgamer20: its a card that supports up to four fxo or fxs |
| 08:15.27 | dlynes_home | FuriousGeorge: i doubt it...one sec |
| 08:16.07 | FuriousGeorge | dlynes_home: its definately not working overhere, lemme mess with my syntax a bit |
| 08:16.10 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:16.16 | darkgamer20 | so the ATA and FXO or FXS make my regular phone use SIP? in other words do they make it like a wifi phone? |
| 08:16.42 | FuriousGeorge | ~tdm400p |
| 08:16.43 | jbot | i heard tdm400p is http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 08:17.39 | FuriousGeorge | if you have a phone you want to interface with asterisk, be it to make a call over sip, or iax or a regular land line you plug that phone into a tdm400p's fxs module. if you have a phone line from ma bell, you plug that into the FXO module |
| 08:20.29 | darkgamer20 | oh ok, so even if I have multiple phone around the house I have to connect them all to the TDM400P which will be in my computer? |
| 08:21.01 | x86 | hey guys, when people call into my IVR, none of thier DTMF is recieved... yet, if I call out from my IP phone through the same server, my DTMF is just fine |
| 08:21.05 | x86 | what could be causing this? |
| 08:21.20 | dudes | sip |
| 08:21.22 | FuriousGeorge | technically asterisk is a pbx and one module is for one phone, each phone gets its own extension. in practice one module can power a few phones |
| 08:21.37 | x86 | dudes: right, how do i fix it? ) |
| 08:21.38 | x86 | :) |
| 08:21.45 | FuriousGeorge | older phones maybe only 2 or 3, newer phones more per module |
| 08:22.05 | JT | it would vary depending on terminating device too, yeah? |
| 08:22.05 | FuriousGeorge | dtmc rfcblah setting |
| 08:22.13 | dudes | do you have port 5060 and 10k-20k |
| 08:22.13 | x86 | dudes: i've tried using various dtmfmode settings for it in sip.conf, but it still does not work |
| 08:22.22 | x86 | dudes: yep |
| 08:22.36 | dudes | then I have no clue |
| 08:22.37 | FuriousGeorge | isnt the point of svn for me to be able to go out and get a specific version of asterisk |
| 08:22.56 | dlynes_home | FuriousGeorge: just grabbing a specific version right now |
| 08:23.05 | dlynes_home | FuriousGeorge: but i don't htink it'll have the history with it |
| 08:23.24 | FuriousGeorge | im screwing around with svn ls and from what i can see my options are 1.0 1.2 and 1.2-netsec |
| 08:23.38 | dlynes_home | FuriousGeorge: is there a way you can apply tags to svn co? |
| 08:24.01 | FuriousGeorge | oh, duh |
| 08:24.07 | FuriousGeorge | its in the tags dir |
| 08:24.10 | dlynes_home | 1.2 that you see there is 1.2 branch |
| 08:24.12 | dlynes_home | FuriousGeorge: nod |
| 08:24.17 | FuriousGeorge | how perfectly obvious and intuitive |
| 08:24.27 | dlynes_home | FuriousGeorge: you sure? |
| 08:24.43 | FuriousGeorge | no, but i sure am sarcastiv sometimes |
| 08:24.48 | FuriousGeorge | *sarcastic |
| 08:24.51 | dlynes_home | svn doesn't seem to have a -z3 or -z9 option, either |
| 08:25.53 | darkgamer20 | FuriousGeorge: I dont understand how more than one phone can use one module since there is a limit to the number of phones you can connect to the TDM400P |
| 08:26.09 | QuAtRo[NL] | dlynes_home: When i call to my voipbuster number I hear the busy tone... |
| 08:26.15 | FuriousGeorge | each module provides an amount of power measuresd in REN |
| 08:26.25 | FuriousGeorge | i believe its 5.0 ren per fxs module |
| 08:26.41 | FuriousGeorge | a modern phone, with dc power, will cost .5 ren |
| 08:26.48 | drray | and the age of the phone dictates how much rens get used up |
| 08:26.54 | FuriousGeorge | an old phone maybe like 2 ren |
| 08:27.06 | FuriousGeorge | but thats an OLD phone |
| 08:27.15 | drray | crank phone |
| 08:27.23 | FuriousGeorge | anything that has its own power adapter will be around 1.0 ren or less |
| 08:27.39 | drray | there is also multiple phone devices on the same line |
| 08:27.49 | drray | fax/modem/tivo/and two phones |
| 08:28.00 | QuAtRo[NL] | And all Asterisk says (in the logs is) http://pastebin.ca/68912 |
| 08:28.20 | QuAtRo[NL] | And call from voipbuster to a dutch landline works fine |
| 08:28.22 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:28.34 | dlynes_home | QuAtRo[NL]: dood...talk about a late response |
| 08:28.58 | dlynes_home | QuAtRo[NL]: anyways..i see nothing wrong there |
| 08:29.04 | dlynes_home | QuAtRo[NL]: where's the rest of the log? |
| 08:29.11 | darkgamer20 | drray: do you mean that i can connect phone to a TDM400P in the similar fashion that i connect computers to a router or switch? |
| 08:29.15 | QuAtRo[NL] | This is what i says when i call.. |
| 08:29.21 | dlynes_home | QuAtRo[NL]: listen |
| 08:29.21 | FuriousGeorge | i got a clock radio that doesnt have its own power that is only 1.0 ren |
| 08:29.28 | dlynes_home | QuAtRo[NL]: you must have more than just debug logs |
| 08:29.39 | QuAtRo[NL] | Which log might be usefull? |
| 08:29.41 | dlynes_home | QuAtRo[NL]: the debug information that I see shows no problems |
| 08:30.01 | dlynes_home | QuAtRo[NL]: edit your logger.conf so you have full => error,warning,verbose,notice,debug,dtmf |
| 08:30.05 | FuriousGeorge | dlynes_home: ok here approaches the moment of truth |
| 08:30.09 | dlynes_home | QuAtRo[NL]: then do a logger restart |
| 08:30.56 | FuriousGeorge | of course all my hunks failed |
| 08:31.18 | dlynes_home | FuriousGeorge: wtf are you doing? |
| 08:31.23 | drray | what is a hunk? |
| 08:31.31 | dlynes_home | FuriousGeorge: i'm using hte same patch on the same svn checkout, and it's succeeding for me |
| 08:31.46 | darkgamer20 | FuriousGeorge: ok i got how the phones use the power and distribute it but what i dont get is how i can connect more than two phones (thats the number of ports for phones i see on the TDM400P) to one TDM400P? |
| 08:31.51 | *** join/#asterisk Elwell (n=Elwell@home.elwell.org.uk) |
| 08:32.00 | QuAtRo[NL] | What do you want to see? |
| 08:32.00 | FuriousGeorge | patch -p0 <meetermaid-1.2.7.1.txt from the asterisk-1.2.7.1 dir |
| 08:32.10 | QuAtRo[NL] | Everything since : Asterisk ready? |
| 08:32.14 | dlynes_home | FuriousGeorge: patch < meetermaid-1.2.71.txt |
| 08:32.16 | x86 | hmm ok |
| 08:32.20 | dlynes_home | FuriousGeorge: forget the -p0 |
| 08:32.30 | *** join/#asterisk lorinc (n=ang@caracas-1415.adsl.interware.hu) |
| 08:32.43 | FuriousGeorge | darkgamer20: i already told you there are up to 4 ports for four lines or extensions depending on the model |
| 08:32.47 | x86 | i changed it to rfc2833 for dtmfmode (i've tried auto, inband, and info before) and that seems to work for recognizing digits pressed |
| 08:32.57 | FuriousGeorge | but in reality one extension can go to more than one phone, i cant put it any other way |
| 08:33.06 | Snake-Eyes | Is having a choppy conversation due to jitter? As in jitter can cause a choppy call? |
| 08:33.09 | x86 | but when i call it with my cell phone, it rings the extension as it should, but when i hang the cell phone up it does not terminate the call |
| 08:34.12 | dlynes_home | FuriousGeorge: every chunk passed for me, anyways |
| 08:34.13 | QuAtRo[NL] | dlynes_home: What should I paste in pastebin? Everything since: Asterisk ready? |
| 08:34.20 | darkgamer20 | FuriousGeorge: well I got everything except that last part but I think I'll research that on my own, thanks alot really you've helped very much! thanks again |
| 08:34.28 | dlynes_home | QuAtRo[NL]: sure |
| 08:34.35 | FuriousGeorge | np |
| 08:34.45 | FuriousGeorge | dlynes_home: ok i forgot the p0 now its asking me one by one |
| 08:34.52 | QuAtRo[NL] | dlynes_home: http://pastebin.ca/68918 |
| 08:35.45 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:36.58 | FuriousGeorge | dlynes_home: http://pastebin.ca/68919 <-- thats where im at so far, ill look at yours now |
| 08:37.24 | QuAtRo[NL] | dlynes_home: If you need config files, let me know |
| 08:38.08 | FuriousGeorge | dlynes_home: so if you check out my pb, its already complainign about not finding the correct file |
| 08:38.09 | dlynes_home | FuriousGeorge: yeah...just type in the paths, manually |
| 08:38.16 | FuriousGeorge | ok |
| 08:38.25 | dlynes_home | FuriousGeorge: so like configs/blahblahblah.conf.sample |
| 08:38.44 | FuriousGeorge | yeah i know, im just making sure we are on the same page so far since you said you got it to work with the same code |
| 08:39.43 | dlynes_home | QuAtRo[NL]: didn't i ask you to give me full logs? |
| 08:39.49 | dlynes_home | QuAtRo[NL]: not debug only logs? |
| 08:40.53 | FuriousGeorge | http://pastebin.ca/68921 |
| 08:41.04 | FuriousGeorge | dlynes_home: failed again, are you sure you applied this patch |
| 08:41.14 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:41.27 | dlynes_home | yep |
| 08:41.45 | FuriousGeorge | where do you patch that file from? |
| 08:41.57 | FuriousGeorge | the asterisk-1.2.7.1 dir? |
| 08:42.02 | dlynes_home | FuriousGeorge: yep |
| 08:42.07 | dlynes_home | FuriousGeorge: where did you patch it from? |
| 08:42.21 | FuriousGeorge | i dont believe you, pastebin please :) |
| 08:42.23 | FuriousGeorge | same place |
| 08:42.59 | dlynes_home | http://pastebin.ca/68922 |
| 08:43.31 | dlynes_home | FuriousGeorge: i'd suggest doing a fresh svn retrieval |
| 08:43.44 | dlynes_home | FuriousGeorge: and patching it the old fashioned way, without the -p0 switch |
| 08:43.55 | dlynes_home | your -p0 switch probably screwed things up |
| 08:43.57 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 08:44.08 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 08:44.11 | Bert- | hello there |
| 08:45.18 | x86 | anyone gotten ztdummy to compile inside of UML? |
| 08:45.26 | Bert- | I have a issue with asterisk and my softphone. I tried X-lite and SJlabs, and when I make a call, I have one way audio. But if I clik on the 'hold' button, I can hear called party for a 1/2 second (I have to click again to hear something for another 1/2 second) |
| 08:45.33 | x86 | i need a timing source for mixmonitor, meetme, and musiconhold to work |
| 08:45.43 | FuriousGeorge | dlynes_home: i was in the parent directory of the svn checkout |
| 08:45.48 | QuAtRo[NL] | dlynes_home: Sorry.. Do you mean this: http://pastebin.ca/68923 |
| 08:45.50 | Bert- | Does someone ever had this pb ? |
| 08:45.59 | dlynes_home | FuriousGeorge: ah...that doesn't help :) |
| 08:45.59 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 08:46.48 | dlynes_home | QuAtRo[NL]: yes, exactly |
| 08:46.58 | dlynes_home | QuAtRo[NL]: and your error is plain as day there for why it won't work |
| 08:47.15 | adorah | Hi everyone.. |
| 08:47.32 | dlynes_home | QuAtRo[NL]: your extensions.conf is buggy as hell, and your username/password isn't correct for your sip phone |
| 08:47.40 | FuriousGeorge | dlynes_home: i take that back, you working dir is confusing me, you checked out into that asterisk-1.2.7.1-svn dir, right? |
| 08:47.48 | dlynes_home | FuriousGeorge: correct |
| 08:48.02 | adorah | How can I join in another extension to a call on a VOIP trunk? |
| 08:48.09 | dlynes_home | FuriousGeorge: because i've already got an asterisk-1.2.7.1 directory, and i didn't want to overwrite it |
| 08:48.33 | FuriousGeorge | dlynes_home: i see that was confusing me when i saw /asterisk/asterisk |
| 08:48.42 | dlynes_home | adorah: conference button on your phone |
| 08:48.53 | adorah | I don't have any.. |
| 08:48.58 | FuriousGeorge | we are actually in the same place, lemme try not patching from the same working dir as you, though i dont see how that would help |
| 08:50.05 | QuAtRo[NL] | dlynes_home: That last line (login errror) was known.. One phone isn't configured |
| 08:50.06 | adorah | Is there any code to dial or should I set the code to enable such a conference? |
| 08:50.44 | adorah | 2 extensions+1 VOIP trunk seems to me a very basic operation.. |
| 08:51.31 | adorah | Is there a way to add a VOIP trunk to a meet-me room? |
| 08:52.38 | dlynes_home | FuriousGeorge: i lost you |
| 08:52.50 | dlynes_home | QuAtRo[NL]: what's the exact problem again? |
| 08:53.03 | dlynes_home | QuAtRo[NL]: you posted the problem so long ago, I can't even scroll up there now |
| 08:53.30 | FuriousGeorge | dlynes_home: i just noticed the only difference between what you did and what i did is that your patch.txt file was not in the same dir as where you ran the patch command from |
| 08:53.35 | FuriousGeorge | so i tried that and same result |
| 08:53.53 | FuriousGeorge | where yours says (Stripping trailing CRs from patch.) |
| 08:53.55 | FuriousGeorge | mine says |
| 08:53.59 | dlynes_home | FuriousGeorge: I didn't run patch -p0 < patchfile.txt beforehand, either |
| 08:54.03 | FuriousGeorge | Hunk #1 FAILED at 72. |
| 08:54.06 | QuAtRo[NL] | dlynes_home: When i call from a dutch landline to a voipbuster account configured in Asterisk |
| 08:54.12 | darkgamer20 | can I have one FXO and two FXS (fax and phone) or do i have to have 2 of both? |
| 08:54.19 | QuAtRo[NL] | I hear the 'busy' tone all the time |
| 08:54.34 | FuriousGeorge | why in the world would a patch be applied correctly to the same code on your box and not on mine |
| 08:54.37 | dlynes_home | QuAtRo[NL]: yeah...i suspect it's the landline giving you the busy signal, not the voipbuster account |
| 08:54.52 | dlynes_home | FuriousGeorge: because you incorrectly applied it to start with |
| 08:54.56 | dlynes_home | FuriousGeorge: and so now it's confused |
| 08:55.16 | dlynes_home | QuAtRo[NL]: let's see your extensions.conf |
| 08:55.19 | FuriousGeorge | dlynes_home: nope, i constantly delete the dir and recheck it out from svn |
| 08:55.22 | QuAtRo[NL] | dlynes_home: Strange, because when i configure the voipbuster account in the voipbuster software it works fine... |
| 08:55.26 | dlynes_home | QuAtRo[NL]: scrub any passwords you have in there first |
| 08:55.42 | dlynes_home | QuAtRo[NL]: that's what i said...voipbuster's probably fine |
| 08:56.08 | dlynes_home | what's your checkout line? |
| 08:56.20 | QuAtRo[NL] | dlynes_home: http://pastebin.ca/68928 |
| 08:56.43 | FuriousGeorge | claudia src # svn co http://svn.digium.com/svn/asterisk/tags/1.2.7.1 asterisk-1.2.7.1 |
| 08:57.23 | FuriousGeorge | then i cd into asterisk-1.2.7.1 then i patch > meetermaid-1.2.7.1.txt |
| 08:57.45 | FuriousGeorge | then it asks me for the first file, so i type in configs/features.conf.sample |
| 08:57.48 | FuriousGeorge | and that hunk fails |
| 08:58.19 | dlynes_home | FuriousGeorge: that's why |
| 08:58.28 | FuriousGeorge | why? |
| 08:58.35 | dlynes_home | FuriousGeorge: you've got your redirection operator pointed in the wrong direction |
| 08:59.14 | dlynes_home | FuriousGeorge: wait a second...what did you download the patch file with? |
| 08:59.58 | FuriousGeorge | konqueror |
| 09:00.07 | FuriousGeorge | my operator is actually pointed the right way |
| 09:00.08 | Bert- | hmm |
| 09:00.13 | Bert- | nobody ever my pb so ? |
| 09:00.20 | Bert- | 'ever had' |
| 09:00.23 | dlynes_home | Bert-: ? |
| 09:00.38 | FuriousGeorge | dlynes_home: is there something wrong with downloading the patch from konqueror? |
| 09:00.43 | dlynes_home | FuriousGeorge: maybe konqueror isn't downloading the file correctly |
| 09:00.48 | dlynes_home | FuriousGeorge: do you have firefox? |
| 09:00.53 | FuriousGeorge | i do, ill try it |
| 09:01.05 | dlynes_home | FuriousGeorge: i'm wondering if maybe konqueror screws up the file somehow |
| 09:01.11 | Bert- | my pb is : I have two sip accounts. The first works fine. But about the other, Asterisk is connected to a Nextone softswitch |
| 09:01.17 | dlynes_home | FuriousGeorge: or maybe you downloaded the wrong patch or something |
| 09:01.33 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:01.33 | FuriousGeorge | i grabbed 1.2.7.1.txt |
| 09:02.04 | darkgamer20 | good night or good (whatever) guys see you later |
| 09:02.10 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 09:02.14 | QuAtRo[NL] | dlynes_home: My extensions.conf is on: http://pastebin.ca/68928 |
| 09:02.38 | dlynes_home | QuAtRo[NL]: yeah, i got it |
| 09:02.46 | dlynes_home | QuAtRo[NL]: but you need to seriously fix it |
| 09:03.09 | Bert- | And every time I try to call, with the 'nextone account', I have to push the 'hold' button |
| 09:03.10 | dlynes_home | QuAtRo[NL]: you've got numerous 's' extensions in your default context |
| 09:03.31 | QuAtRo[NL] | dlynes_home: Isn't that allowed? |
| 09:03.33 | dlynes_home | Bert-: yeah...i have no idea, and i'm too tired to think about it |
| 09:03.38 | dlynes_home | QuAtRo[NL]: of course not |
| 09:04.14 | Bert- | I suppose it is a bug from one of my conf files, as it works fine with another asterisk... But really don't understand why call is held by default |
| 09:05.21 | FuriousGeorge | dlynes_home: failed again |
| 09:05.38 | FuriousGeorge | i gotta finsih compiling what i got b/c they are gonna open in an hour and not have phones |
| 09:05.44 | FuriousGeorge | at least ill have 1.2.9.1 |
| 09:06.03 | dlynes_home | FuriousGeorge: pastebin your patch file |
| 09:06.14 | dlynes_home | FuriousGeorge: i'll compare against mine |
| 09:06.22 | dlynes_home | FuriousGeorge: my guess is it's the patch file |
| 09:06.55 | FuriousGeorge | http://pastebin.ca/68936 |
| 09:07.04 | FuriousGeorge | hope you are right |
| 09:07.07 | FuriousGeorge | but i doubt it |
| 09:07.18 | QuAtRo[NL] | dlynes_home: What should be on the place of the 's'? |
| 09:07.26 | dlynes_home | FuriousGeorge: well, there's gotta be some simple explanation |
| 09:07.41 | FuriousGeorge | i hope you are right |
| 09:08.00 | dlynes_home | QuAtRo[NL]: give them an extension number |
| 09:08.21 | dlynes_home | QuAtRo[NL]: nvm...you already gave them an extension...why include them twice? |
| 09:10.35 | FuriousGeorge | dlynes_home: notice any difference? |
| 09:11.10 | FuriousGeorge | i think im just gonna have to wait for 1.4 |
| 09:11.34 | QuAtRo[NL] | dlynes_home: removed that... |
| 09:12.23 | dlynes_home | FuriousGeorge: yeah...there's a difference |
| 09:12.26 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:12.42 | dlynes_home | FuriousGeorge: can i just dcc send you my patch file? |
| 09:12.47 | Bert- | <PROTECTED> |
| 09:12.53 | Bert- | what is that plz ? |
| 09:13.18 | Bert- | it is a call I made but what is hint ? |
| 09:13.34 | dlynes_home | Bert-: do you have any blf defined on that phone? |
| 09:13.45 | dlynes_home | Bert-: or do you have subscribecontext= specified in your sip.conf file? |
| 09:13.59 | Bert- | no, and it is a remote phone I called. |
| 09:14.08 | Bert- | No I don't use subscribecontext |
| 09:14.13 | Bert- | let me see what is it |
| 09:14.17 | dlynes_home | Bert-: then something's asking for blf |
| 09:15.03 | FuriousGeorge | dlynes_home: |
| 09:15.06 | FuriousGeorge | sure send away |
| 09:15.17 | Bert- | what means blf ? |
| 09:15.21 | FuriousGeorge | never tried recieving with this client though |
| 09:15.25 | Bert- | busy lamp file |
| 09:15.27 | Bert- | field |
| 09:15.36 | Bert- | but don't understand the aim o that thing |
| 09:15.48 | FuriousGeorge | dlynes_home: its gotta be quick though, i got 45 minutes before they open |
| 09:15.52 | FuriousGeorge | :) |
| 09:16.25 | FuriousGeorge | we must be getting the patch from the same place, im getting it from bugtracker |
| 09:17.43 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 09:18.19 | FuriousGeorge | dlynes_home: did you try to send it yet? |
| 09:19.04 | dlynes_home | FuriousGeorge: ah....one sec |
| 09:20.40 | dlynes_home | FuriousGeorge: i guess you're not able to receive it...one second |
| 09:21.18 | FuriousGeorge | hmm, i wonder why not, i thought only sending required port forwarding, maybe i got it backwards |
| 09:21.43 | dlynes_home | FuriousGeorge: try this instead: http://www.ancient-legacy.org/metermaid-1.2.7.1.txt |
| 09:22.01 | dlynes_home | FuriousGeorge: download it using firefox |
| 09:22.58 | Bert- | well I really don't understand |
| 09:23.13 | Bert- | all calls I try to make are placed in hold by default |
| 09:23.27 | Bert- | I have to 'unhlod' a call before being able to hear called party |
| 09:23.54 | dlynes_home | FuriousGeorge: did it work this time? |
| 09:24.38 | FuriousGeorge | http://pastebin.ca/68952 |
| 09:24.48 | FuriousGeorge | tell me what i could have possibly done wrong |
| 09:25.24 | *** join/#asterisk |oranjia| (n=kvirc@dsl-165-140-69.telkomadsl.co.za) |
| 09:25.24 | dlynes_home | wtf? |
| 09:25.34 | dlynes_home | well, if you want, i could ssh in, and try |
| 09:26.18 | FuriousGeorge | thanks, but how about you send the 4 files its trying to change as oppesed to the patch and ill just drop them in there |
| 09:26.29 | dlynes_home | ok |
| 09:26.29 | FuriousGeorge | that should last until 1,4 |
| 09:26.33 | dlynes_home | four files? |
| 09:26.37 | dlynes_home | I think it was three |
| 09:27.08 | FuriousGeorge | features.conf.samples res/res_features.c channels/chan_local.c and asterisk/features.h |
| 09:27.31 | dlynes_home | ah...i see a difference between yours and mine |
| 09:27.41 | FuriousGeorge | where? |
| 09:27.42 | dlynes_home | mine has cr/lf's; yours doesn't |
| 09:27.57 | dlynes_home | if anything though |
| 09:27.57 | FuriousGeorge | what the patch? |
| 09:28.05 | dlynes_home | that should make it not work, not make it work :p |
| 09:28.13 | FuriousGeorge | lol |
| 09:28.54 | X-Rob | Well. |
| 09:30.01 | dlynes_home | FuriousGeorge: just checking it out again |
| 09:30.38 | FuriousGeorge | see i ddint check out the patch, i just dl'ed it from the bugtracker page |
| 09:30.43 | dlynes_home | FuriousGeorge: you're using revision 35390 right? |
| 09:30.46 | FuriousGeorge | what command are you using to check out the patch |
| 09:30.46 | dlynes_home | FuriousGeorge: same here |
| 09:30.54 | FuriousGeorge | yeah |
| 09:30.58 | FuriousGeorge | 35390 |
| 09:31.01 | dlynes_home | FuriousGeorge: right click on the patch link from the web page, and click save as |
| 09:32.35 | FuriousGeorge | i did and same result |
| 09:32.50 | FuriousGeorge | i got 28 minutes to get this compiled and installed so its now or never on those files |
| 09:33.34 | dlynes_home | FuriousGeorge: ok...those four files are on my web server now |
| 09:33.48 | dlynes_home | FuriousGeorge: patch file's still there, too |
| 09:35.03 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:35.35 | dlynes_home | FuriousGeorge: i thought of one other thing, too |
| 09:35.38 | dlynes_home | FuriousGeorge: type patch -v |
| 09:35.46 | dlynes_home | FuriousGeorge: which version are you running? |
| 09:39.17 | Bert- | hmm why asterisk complains about hint when I try to call ? I mean Asterisk say I should add a hint for 087028xxxx, but that is the called number |
| 09:39.43 | Bert- | sipphone connected to my asterisk are 'hinted' |
| 09:40.00 | dlynes_home | Bert-: i thought you said you weren't using blf's? |
| 09:40.06 | Bert- | yep |
| 09:40.13 | Bert- | but now it is corrected :) |
| 09:40.14 | dlynes_home | blf's are hinted |
| 09:40.37 | Bert- | I made a context => [blf] |
| 09:40.46 | Bert- | in this context, I declared all my sipphones |
| 09:41.01 | Bert- | and I include [blf in the default context |
| 09:41.40 | Bert- | I'm pretty sure my issue about 'hold by default' state deals with that |
| 09:43.25 | Bert- | the way is, every time I launch my softphone (x-lite), I can see this line in asterisk : ERROR[11018]: chan_sip.c:10988 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 0556525138 in context blf |
| 09:43.33 | dlynes_home | anyways |
| 09:43.37 | Bert- | but this number is the one I tried to call |
| 09:43.38 | dlynes_home | i'm having trouble keeping my eyes open |
| 09:43.42 | dlynes_home | i'm hitting the hay |
| 09:43.48 | Bert- | I can't declare it in my blf context |
| 09:43.51 | *** join/#asterisk yunusyb (n=yunus@60.254.1.190) |
| 09:43.55 | Bert- | then should go to bed :) |
| 09:43.58 | dlynes_home | Bert-: I'll have to help you next time i see you, if you're still having the problem |
| 09:44.05 | dlynes_home | Bert-: not many people here use blf's |
| 09:44.06 | Bert- | I hope I'll find :) |
| 09:44.21 | dlynes_home | I'm one of the few that's had any amount of success with them |
| 09:44.23 | Bert- | I don't want to use it as I don't really understand what is it |
| 09:44.26 | FuriousGeorge | patch 2.5.9 |
| 09:44.35 | dlynes_home | FuriousGeorge: i'm using 2.5.4 |
| 09:44.43 | dlynes_home | FuriousGeorge: did those four files and the patch work for you? |
| 09:44.53 | Bert- | but as Asterisk complains about that, I'm sure all my issues comes from this blf error |
| 09:44.55 | FuriousGeorge | dlynes_home: compiling now |
| 09:44.58 | dlynes_home | if not, it might jsut be your patch tool version |
| 09:44.58 | FuriousGeorge | fingers crossed |
| 09:45.01 | FuriousGeorge | hard to type |
| 09:45.03 | dlynes_home | FuriousGeorge: oh...so the patch worked? |
| 09:45.17 | FuriousGeorge | so far i just compiled chan_sip |
| 09:45.23 | FuriousGeorge | so thats a big step |
| 09:45.28 | dlynes_home | FuriousGeorge: no errors on the patch though, right? |
| 09:45.29 | FuriousGeorge | chan_local just compiled |
| 09:45.42 | FuriousGeorge | err, the files you gave me werent patched? |
| 09:45.47 | dlynes_home | nope |
| 09:45.49 | FuriousGeorge | LOL |
| 09:45.59 | FuriousGeorge | ctrl-c |
| 09:46.01 | dlynes_home | i didn't realize you wanted the patched files :) |
| 09:46.24 | *** join/#asterisk wifi_guy (n=jdowe@h64-187-46-2.gtcust.grouptelecom.net) |
| 09:46.34 | FuriousGeorge | can you replace those files with patched ones ever so swiftly :) |
| 09:46.39 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-145-56-236.telkomadsl.co.za) |
| 09:46.43 | FuriousGeorge | i hate to go this far for nothing |
| 09:47.33 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 09:47.35 | Bert- | hmm Asterisk don't understand OPTION request |
| 09:47.38 | Bert- | ?? |
| 09:47.45 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-145-56-236.telkomadsl.co.za) |
| 09:48.02 | *** part/#asterisk wifi_guy (n=jdowe@h64-187-46-2.gtcust.grouptelecom.net) |
| 09:48.05 | dlynes_home | FuriousGeorge: same place and same files, but it's in www.ancientlegacy.org/patches/ |
| 09:48.11 | Bert- | it returns 404 not foun |
| 09:48.15 | Bert- | foudn |
| 09:48.53 | dlynes_home | it does....you've just got other issues |
| 09:49.40 | Bert- | well only error msg I can see is in 'sip debug', I see a packet from my host with OPTIONS request, then asterisk answer sith 404 not found or 405 method not allowed |
| 09:50.17 | dlynes_home | 404 means it's working |
| 09:50.25 | Bert- | ? |
| 09:50.36 | Bert- | 404 not found means ... not found for me |
| 09:50.40 | Bert- | :) |
| 09:50.43 | dlynes_home | if options didn't work, you wouldn't get a 404 |
| 09:50.50 | Bert- | okay |
| 09:50.52 | dlynes_home | 404 means it worked, but it couldn't find the resource in question |
| 09:51.09 | Bert- | ok |
| 09:51.28 | Bert- | but I don't understand why asterisk wnat me to register a remote number in my blf context |
| 09:51.34 | Bert- | I'm missing something |
| 09:51.36 | Bert- | but what |
| 09:51.51 | Bert- | register a local phone, ok. But a remote... |
| 09:52.16 | FuriousGeorge | dlynes_home: compiling again t-8min |
| 09:52.54 | *** join/#asterisk djtremors (n=newjacks@ppp121-90.static.internode.on.net) |
| 09:53.30 | djtremors | yay I'm on... been a while since going on irc that i've forgotten how to use it.lol |
| 09:54.11 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 09:54.27 | Bert- | anyway, no one here had the same pb as me |
| 09:54.42 | dlynes_home | pb? |
| 09:54.50 | Strom_C | pacific bell |
| 09:54.54 | Strom_C | obviously |
| 09:54.58 | dlynes_home | i guess |
| 09:55.05 | Bert- | on all calls (both incoming and outgoing), I have to push the hold button on my sipphone to hear my correspondant |
| 09:55.19 | dlynes_home | oh |
| 09:55.21 | dlynes_home | problem |
| 09:55.33 | Bert- | well yep b for problem :) |
| 09:55.37 | dlynes_home | Bert-: dood...spell out your words so we don't have to play guessing games about what you're trying to say |
| 09:56.16 | dlynes_home | Bert-: we're not literate in aol-speak |
| 09:56.30 | Bert- | you right, sorry |
| 09:56.39 | Bert- | hahaha :) |
| 09:57.15 | djtremors | hey all, anyone use softphones here (probably alot do). which one do you use? |
| 09:57.32 | Bert- | x-lite, sjphone, idefix |
| 09:57.35 | djtremors | i'm playing about with Express Talk, looks alright but having troubles with asterisk. |
| 09:57.39 | Bert- | (all with linux) |
| 09:57.47 | FuriousGeorge | dlynes_home: as far as i can tell we did good |
| 09:58.29 | QuAtRo[NL] | dlynes_home: If the logger only says 'simple logging enabled' on startup... Does that mean there are no syntax errors in the extensions.conf |
| 09:58.32 | FuriousGeorge | dlynes_home: thanks so much for all your help, if you handnt provided those files i would have had nothing to show for the last three hours i stayed up |
| 09:59.48 | *** part/#asterisk yunusyb (n=yunus@60.254.1.190) |
| 10:00.43 | Bert- | fabulous |
| 10:00.56 | Bert- | every time I call, call is on hold |
| 10:00.59 | Bert- | by default |
| 10:01.48 | hads|home | QuAtRo[NL]: 'simple logging enabled' doesn't have anything to do with extensions.conf |
| 10:02.01 | Strom_C | Bert-: what kind of telephone is it? |
| 10:06.05 | Bert- | a fuck*** softphone |
| 10:06.16 | Bert- | sorry but I have this issue for two days |
| 10:06.34 | Bert- | If it doesn't work in the evening, we will stop tests and buy a Cisco one |
| 10:06.36 | iDunno | it does what? ohhh, I want me one of those :) |
| 10:06.38 | Strom_C | Bert-: the solution, quite obviously, is to get a real telephone |
| 10:07.00 | Bert- | Strom_C : all softphones I tried have the same problem |
| 10:07.05 | Bert- | so it is from asterisk |
| 10:07.10 | Bert- | but don't know where |
| 10:07.33 | Strom_C | what version of asterisk are you running? |
| 10:07.37 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 10:07.45 | Bert- | 1.2.9 |
| 10:07.48 | Bert- | the last |
| 10:07.51 | Bert- | compiled from scratch |
| 10:07.53 | Bert- | on debian |
| 10:08.12 | Strom_C | are you running freepbx or anything? |
| 10:08.20 | Bert- | nothing |
| 10:08.22 | Strom_C | ok |
| 10:08.39 | Strom_C | where are you dialing out to? |
| 10:08.54 | Strom_C | and what kind of facilities are you using for termination? |
| 10:09.09 | Strom_C | er, sorry...origination /and/ termination |
| 10:11.24 | QuAtRo[NL] | dlynes_home: My log files don't show any incomming call... |
| 10:11.43 | QuAtRo[NL] | dlynes_home: Does that mean the server isn't reached |
| 10:11.55 | Bert- | well it is really simple |
| 10:12.06 | Bert- | I'm dialing my mother on his PSTN phone |
| 10:12.12 | Bert- | I doing nothing complicated |
| 10:12.17 | Bert- | it just doesn't works |
| 10:12.27 | Bert- | I can call, be called, all is okay |
| 10:12.40 | Bert- | except that I have to push this fucking hold button |
| 10:12.41 | Strom_C | are you calling out over a POTS line, an ISDN line, or VoIP? |
| 10:12.42 | Bert- | !!! |
| 10:14.31 | Bert- | VoIP |
| 10:14.39 | Strom_C | SIP or IAX? |
| 10:14.40 | Bert- | same is with my two accounts |
| 10:14.42 | Bert- | SIP |
| 10:14.45 | Bert- | all sip |
| 10:14.47 | Strom_C | which carrier? |
| 10:14.51 | Bert- | ... |
| 10:14.55 | Bert- | it is not a carrier issue |
| 10:15.08 | Bert- | I use free (french internet provider) |
| 10:15.21 | Bert- | and I use a nextone softswitch with A2Z destinations |
| 10:15.32 | Bert- | there is around 780 concurrent calls just now |
| 10:15.37 | Strom_C | Bert-: I'm just trying to help you. |
| 10:15.43 | iDunno | what's the "fucking hold button" do? |
| 10:15.44 | Bert- | the problem isme |
| 10:15.57 | Bert- | well Strom_Csorry :) |
| 10:16.07 | Strom_C | Bert-: what happens when you place an intra-switch call to another softphone? |
| 10:16.11 | iDunno | and what's the contents of extensions.conf? |
| 10:16.14 | Bert- | when I push the hold button, I can hear and be heard |
| 10:16.26 | Bert- | it is a stupid problem really |
| 10:16.36 | iDunno | hmm. maybe you've got the Dial command wrong in extensions.conf |
| 10:16.50 | Strom_C | Bert-: pastebin extensions.conf |
| 10:16.57 | Bert- | let me do it |
| 10:17.24 | *** join/#asterisk erwinism (i=erwin@61.9.118.37) |
| 10:18.16 | erwinism | hello, i have a little problem, i have X100P, if someone calls from PSTN, the asterisk wont hangup and leave the PSTN line open. How can i fix this? |
| 10:18.38 | Strom_C | erwinism: order disconnect supervision from your telephone company |
| 10:19.14 | erwinism | oh |
| 10:19.24 | *** join/#asterisk Wifi_Guy (n=Jdowe@h64-187-46-2.gtcust.grouptelecom.net) |
| 10:19.29 | *** join/#asterisk yacyac (n=yac@202.189.231.82) |
| 10:19.39 | Wifi_Guy | hi all |
| 10:19.57 | yacyac | hii Wifi_Guy |
| 10:20.53 | Wifi_Guy | can someone give me a basic idea of hardware requirements for running asterisk as a sip server for about 10 clients and 4 trunks? |
| 10:21.01 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 10:21.10 | Strom_C | Wifi_Guy: assuming no transcoding, any old PC should handle that just fine |
| 10:22.08 | Wifi_Guy | transcoding? that refers to the audio data in realtime? |
| 10:22.18 | benjamin7062 | If this is the wrong channel, just tell me to leave, but... Does 'anyone' use this in a call center environment; if yes, any of you guys have real time stats (queue's, what agents on a call, in bound, outbound, how long the duration is, etc?) |
| 10:22.33 | Strom_C | that refers to translating the audio data from one codec to another |
| 10:22.58 | djtremors | @wifi_guy.. there's a PDF on asterisk site which talks about system rewuirements.. not bad.. reading it atm actually. |
| 10:23.10 | erwinism | Strom_C, i mean if someone calls from PSTN and end the call, the asterisk wont hangup. example, "if they just dial and listen to IVR then hangup" |
| 10:23.12 | djtremors | it's called AsteriskTFOT.pdf |
| 10:23.14 | Wifi_Guy | I see... Well, all clients would be using 711u and the trunk would as well |
| 10:23.23 | Strom_C | benjamin7062: I'm afraid you're in the wrong channel. We discuss nothing but muffins here. |
| 10:23.41 | Strom_C | Wifi_Guy: then you should be fine with pretty much any PC made in the last three to five years |
| 10:23.47 | Wifi_Guy | muffins... lol |
| 10:23.57 | *** join/#asterisk locelavi (n=gd@172-135.240.81.adsl.skynet.be) |
| 10:24.06 | benjamin7062 | Strom_C, Dang, someone mentioned that -- I was mislead by the title... sigh, back to college for me |
| 10:24.21 | Strom_C | erwinism: yes, like I said, call your telephone company and order far-end disconnect supervision. |
| 10:24.22 | Wifi_Guy | thanks for the $0.02, I'll check that PDF as well! |
| 10:24.47 | benjamin7062 | Any of you guys a dev for asterisk? |
| 10:24.59 | Strom_C | benjamin7062: the devs all hang out in #asterisk-dev |
| 10:25.08 | erwinism | Strom_C, thank you very much |
| 10:25.12 | benjamin7062 | ahh |
| 10:25.25 | Strom_C | benjamin7062: but you'll have more success asking this question during daytime hours in north america |
| 10:25.38 | benjamin7062 | I'm in CST |
| 10:25.45 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-145-56-236.telkomadsl.co.za) |
| 10:25.47 | Strom_C | no, you're in CDT |
| 10:25.49 | benjamin7062 | If they were devoted like me... they would be up all night like me! |
| 10:25.55 | Strom_C | it's daylight time right now :) |
| 10:26.00 | benjamin7062 | right |
| 10:26.01 | benjamin7062 | good point |
| 10:26.13 | Strom_C | the actual time zone is "central time" |
| 10:26.29 | benjamin7062 | CSTCDT if you ask linux |
| 10:26.35 | benjamin7062 | ;-) |
| 10:26.42 | Strom_C | "standard time" and "daylight time" refer to specific offsets from GMT, not the physical zone |
| 10:26.46 | Strom_C | </pedantic> |
| 10:27.17 | benjamin7062 | yes master watch sir... I will keep that in mind when typing 'brief' references in the future. |
| 10:27.47 | benjamin7062 | heh |
| 10:27.55 | hads|home | benjamin7062: There is an app called queuemetrics that does something like that from memory. |
| 10:28.04 | Strom_C | tick tick tick tick U.S. Naval Observatory master clock tick tick tick at the tone, mountain daylight time tick tick tick... |
| 10:28.15 | hads|home | I know nothing of it though. |
| 10:28.20 | erwinism | hehehe |
| 10:28.51 | benjamin7062 | Strom_C, lol... nice I would spew some GPS timing data.. but it's look something like 101010001000111010101000011 |
| 10:29.04 | benjamin7062 | it'd* |
| 10:29.07 | benjamin7062 | sigh |
| 10:29.17 | benjamin7062 | hads|home, Thank you! |
| 10:29.53 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 10:30.21 | Strom_C | it's Qwell! |
| 10:31.04 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 10:32.27 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 10:32.28 | *** part/#asterisk Wifi_Guy (n=Jdowe@h64-187-46-2.gtcust.grouptelecom.net) |
| 10:33.32 | Strom_C | and then...catsex |
| 10:33.45 | FuriousGeorge | anyone using metermaid? i just parked a call and didnt get a light. if i configure the parkexten button to be a park orbit, as suggested in the wiki, i cant transfer there at all |
| 10:34.09 | FuriousGeorge | does your CLI say anything as to the status of the channel hint when you park a call? |
| 10:34.38 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 10:36.50 | *** join/#asterisk allocelavi (n=gd@172-135.240.81.adsl.skynet.be) |
| 10:44.20 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 10:48.39 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:48.43 | backblue | morning all |
| 10:48.48 | RoyK | mrnng |
| 10:50.33 | FuriousGeorge | ok when i shoe hints i see my parked extensions, which i suppose means the patch went ok |
| 10:50.39 | FuriousGeorge | my park orbit button is working now |
| 10:50.56 | FuriousGeorge | when i show hints, if someone is parked it shows up as inuse |
| 10:51.00 | backblue | ppl, i need any kind of special stat in zaptel drivers to receive fax on incoming line? |
| 10:51.00 | FuriousGeorge | BUT still no light |
| 10:51.06 | erwinism | Strom_C, i set echocancel as true, echotraining as true but the my voice got echo when i call to PSTN, (on voip, i have no problem) |
| 10:51.33 | erwinism | how can i solve that? |
| 10:51.45 | FuriousGeorge | exten => 701,hint,Local/701@parkedcalls |
| 10:52.24 | FuriousGeorge | then i configure the snom to see 701 as an extension, which allows for working presence with sip peers, but evidentally not for chan local |
| 10:53.26 | FuriousGeorge | which was the whole point of the patch |
| 10:58.52 | fnordian | hi |
| 10:59.09 | fnordian | is it possible to call a function like sip_header from an agi-script? |
| 11:00.05 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 11:01.49 | *** join/#asterisk muppetmaster (n=jasongoe@27.Red-213-97-53.staticIP.rima-tde.net) |
| 11:01.56 | muppetmaster | Hola |
| 11:02.06 | muppetmaster | Does a reload pick up changes for zapata.conf and zaptel.conf? |
| 11:02.11 | muppetmaster | Or does one need a full restart? |
| 11:02.42 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:03.32 | drray | ztcfg -vv |
| 11:04.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:04.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:04.36 | muppetmaster | Ah, and that reloads? |
| 11:04.44 | hads|home | muppetmaster: zaptel.conf will require a ztcfg. Some changes in zapata.conf are picked up by a reload and some aren't. |
| 11:04.52 | muppetmaster | Ok |
| 11:04.59 | muppetmaster | So a ztcfg -vv and then a reload? |
| 11:05.09 | joelsolanki | Hi all. i have ser+asterisk ..i want ser users to save/retrieve voicemail on asteris. |
| 11:05.19 | joelsolanki | how can i do that? any hints / docs |
| 11:05.23 | erwinism | reboot is more reliable hehe |
| 11:05.27 | hads|home | muppetmaster: Some changes in zapata.conf are picked up by a reload and some aren't. |
| 11:05.49 | muppetmaster | So, even with a ztcfg -vv I should do a restart of Asterisk to be sure? |
| 11:06.32 | erwinism | in zaptel.conf changes, ztcfg will do, in zapata.conf, reload will do |
| 11:06.55 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 11:06.59 | muppetmaster | erwinism, thanks |
| 11:07.45 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:09.12 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 11:11.17 | mover | anyone noticed about an inbound issue to new Nokia E60 Series in cause of missing stun support in this voip UA? |
| 11:11.21 | fnordian | do you know a way to access sip-header-fields from an agi-script? |
| 11:14.01 | mover | fnordian: show function SIP_HEADER |
| 11:16.41 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 11:16.51 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:16.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:16.57 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 11:17.02 | backblue | Jun 22 12:17:07 ERROR[11876]: chan_zap.c:2689 zt_hangup: What is wrong with you? You cannot use cause 17 number when in state 7! |
| 11:17.08 | backblue | ups, sorry :\ |
| 11:17.27 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:17.43 | fnordian | mover: Jun 22 13:09:48 WARNING[14815]: res_agi.c:1091 handle_exec: Could not find application (SIP_HEADER(Proxy-Authorization)) |
| 11:17.46 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 11:18.53 | mover | fnordian: what version of * us use? |
| 11:19.01 | fnordian | 1.2.7 |
| 11:19.30 | mover | use show function SIP_HEADER on cli it is there? |
| 11:19.34 | fnordian | i have SIP_HEADER in dialplans and on the cli |
| 11:19.46 | fnordian | but i cannot access it from an agi |
| 11:20.13 | mover | on cli?? |
| 11:20.25 | fnordian | yes, i have it there |
| 11:20.41 | *** join/#asterisk tRSS (n=tRSS@pk-isb-trg-sc01-019.speedcast.com) |
| 11:21.43 | mover | i use EXEC FUNCTION(name) |
| 11:22.36 | mover | if it not work on AGI you can use it in DIALPLAN all is ok. You must bypass the Variables |
| 11:23.53 | *** join/#asterisk sandos (n=sandos@83.233.97.253) |
| 11:24.31 | fnordian | AGI Rx << EXEC SIP_HEADER(Proxy-Authorization) -- AGI Script Executing Application: (SIP_HEADER(Proxy-Authorization)) Options: ((null)) |
| 11:25.01 | Modcuts | Is any work gonna be done to create a 64bit verison of asterisk? |
| 11:27.00 | fnordian | mover: i guess i have to do it that way, but i dont like that idea so much |
| 11:28.37 | tRSS | how can i run a sql query from extensions.conf to check if something is in the database or not and then process the call accordingly? I guess I just want to know how to run a sql query from extensions.conf? |
| 11:32.59 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 11:36.38 | *** join/#asterisk eivindtr (n=wingnut-@brmweb.barum.folkebibl.no) |
| 11:39.19 | Bert- | well |
| 11:40.22 | Bert- | It is impossible without licenses, to make a call from a sipphone (ulaw codecs) to a PSTN number, through a gateway only using G729, as asterisk can't do any codec translation ? |
| 11:40.44 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 11:40.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 11:41.41 | FuriousGeorge | anyone using metermaid in here? |
| 11:42.50 | FuriousGeorge | everything seems to be working except my device states |
| 11:42.57 | FuriousGeorge | which is kida the integral part |
| 11:43.12 | russellb | he updated something in that branch this morning |
| 11:43.31 | FuriousGeorge | russellb: im just using the patch for 1.2.7.1 |
| 11:43.37 | FuriousGeorge | not the trunk branch |
| 11:43.50 | russellb | ah, well then your patch probably isn't up to date :) |
| 11:44.14 | FuriousGeorge | russellb: the alternative then is to use trunk? |
| 11:44.24 | *** part/#asterisk eivindtr (n=wingnut-@brmweb.barum.folkebibl.no) |
| 11:44.36 | russellb | that would be best, yes |
| 11:44.41 | russellb | since it's still in development ... |
| 11:45.01 | FuriousGeorge | russellb: im kinda wary of that as this is for production. i thought if he patched 1.2.7.1 it would be guarenteed to be at least as stable as 1.2.7.1 |
| 11:45.03 | tRSS | how can I run a sql query from extensions.conf to check for something in the database? |
| 11:45.13 | FuriousGeorge | unless the patched code crashes it of course |
| 11:45.59 | russellb | tRSS: look up func_odbc |
| 11:46.18 | tRSS | alright thanks russellb |
| 11:46.19 | _problem_ | tRSS: there is a command available called system()..with which u can execute linux shell command try with that. |
| 11:47.54 | tRSS | _problem_: how to use the system command in my case? russellb: I have tried to look up func_odbc, can't find anything? |
| 11:47.55 | Bert- | what is Jun 22 13:47:20 WARNING[11932]: chan_sip.c:2555 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 8/8) please ? |
| 11:48.18 | russellb | tRSS: it's in trunk, but has been backported to 1.2 as well |
| 11:48.22 | russellb | let me see if i can find a link ... |
| 11:48.58 | _problem_ | tRSS: go to www.asteriskguru.com/tutorials and find system and trysystem command..there u can find their descrptions of use |
| 11:49.11 | russellb | tRSS: http://svncommunity.digium.com/view/func_odbc/1.2/ |
| 11:49.14 | russellb | that's it in svn ... |
| 11:49.30 | russellb | svn co http://svncommunity.digium.com/svn/func_odbc/1.2 func_odbc-1.2 |
| 11:49.41 | russellb | then check out the README on how to install it |
| 11:50.16 | russellb | then, once installed ... you should be able to type "show function ODBC" at the Asterisk CLI |
| 11:51.08 | *** join/#asterisk sandos (n=sandos@83.233.97.253) |
| 11:51.54 | tRSS | alright thanks. let me explore more into this |
| 11:52.45 | tRSS | oh another question: Does asterisk have the capability to allow a supervisor to barge into a call? I know that someone spy a call (chanspy) but does this allow a person to barge into this call as well? |
| 11:53.53 | russellb | if you're using chan_zap, there is an app called ZapBarge |
| 11:53.54 | *** part/#asterisk sandos (n=sandos@83.233.97.253) |
| 11:54.10 | russellb | otherwise, you'd have to emulate the functionality using MeetMe |
| 11:54.48 | *** join/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net) |
| 11:54.51 | tRSS | hmm, i thought of that too, but I wanted to keep things simple, rather then making a complicated mess, you know what I mean |
| 11:55.00 | russellb | yep |
| 11:55.07 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 11:55.55 | russellb | it wouldn't be bad with meetme, though ... |
| 11:57.04 | FuriousGeorge | russellb: is the metermaid branch of trunk at all different from regular old trunk? |
| 11:57.22 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 11:57.51 | russellb | it's different in that it has the metermain changes :) |
| 11:58.06 | FuriousGeorge | i was hoping it was based on 1.2.7.1 |
| 11:58.12 | FuriousGeorge | i mean .9.1 |
| 11:58.16 | FuriousGeorge | you know what i mean |
| 11:58.27 | QuAtRo[NL] | How can i test if a certain phonenumber is handled by my Asterisk |
| 11:58.32 | russellb | no, it is based on trunk |
| 11:58.36 | russellb | maybe he has a 1.2 branch, i don't know |
| 11:58.45 | russellb | http://svn.digium.com/view/asterisk/team/oej |
| 12:00.56 | FuriousGeorge | russellb: what about that multiparking |
| 12:02.00 | *** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk) |
| 12:02.09 | russellb | i don't know, take a look at his team directory |
| 12:02.16 | russellb | unless it specifies "1.2", it's trunk |
| 12:02.39 | MikeJ[Laptop] | slick move russellb.... |
| 12:02.45 | MikeJ[Laptop] | could have said somthing. |
| 12:02.51 | FuriousGeorge | i looked in the CHANGES file and multiparkings top entry is Changes since Asterisk 1.2.0-beta2: |
| 12:03.00 | FuriousGeorge | but that is the other side of the spectrum |
| 12:03.15 | russellb | MikeJ[Laptop]: heh, what are you referring to? |
| 12:03.21 | MikeJ[Laptop] | guess |
| 12:03.25 | russellb | lol |
| 12:03.25 | QuAtRo[NL] | russellb: Do you know how I can test if a certain phonenumber is handled by my Asterisk.. |
| 12:03.26 | russellb | :D |
| 12:03.36 | MikeJ[Laptop] | bad form |
| 12:03.42 | russellb | MikeJ[Laptop]: yeah, that was rough |
| 12:04.00 | russellb | MikeJ[Laptop]: i thought about sneaking the commit in with something else, but i just wanted to remove it before someone said something to me |
| 12:04.09 | MikeJ[Laptop] | all it takes is saying somthing... instead... you end up coming off like a real jerk |
| 12:04.09 | russellb | QuAtRo[NL]: call it? |
| 12:04.10 | joelsolanki | hi all. is possible to disconnect particular sip call in asterisk ? |
| 12:04.14 | MikeJ[Laptop] | you guessed wrong |
| 12:04.21 | QuAtRo[NL] | russellb: You're kidding me :P |
| 12:04.31 | FuriousGeorge | russellb: is there a specific file i should be looking for that gives the version number or is changes it? |
| 12:04.35 | MikeJ[Laptop] | banning somone from a channel, without bothering saying anything to them. |
| 12:04.41 | QuAtRo[NL] | It should be handled by Asterisk, but no phone is gonna ring.. |
| 12:04.51 | russellb | MikeJ[Laptop]: oh, we're talking about totally different stuff |
| 12:04.58 | MikeJ[Laptop] | yep. |
| 12:05.13 | MikeJ[Laptop] | just wanted to make sure you know that the way you handled that was lousy.. |
| 12:06.31 | russellb | nothing personal ... people just have a bad habit of talking about things they shouldn't in that channel |
| 12:06.40 | russellb | but sorry for not saying anything. |
| 12:07.10 | joelsolanki | can i disconnect particular call in asterisk ? |
| 12:07.23 | russellb | joelsolanki: "soft hangup" from the cli |
| 12:08.11 | MikeJ[Laptop] | russellb, that's the point.. by not saying anything.. you make it personal... remember.. you represent digium and asterisk... pulling stuff like that represents badly on the project. and yourself |
| 12:08.25 | joelsolanki | ok let me look at soft hangup |
| 12:08.29 | *** part/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 12:08.33 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:09.09 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:09.49 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 12:11.00 | *** join/#asterisk Henk (n=Henk@s5593c2e9.adsl.wanadoo.nl) |
| 12:12.56 | *** join/#asterisk nettux (n=Frank@62.123.91.224) |
| 12:13.08 | hads|home | bracket highlighting is cool. |
| 12:15.22 | Henk | Hi, i'm trying to make a call file the calls my cellphone. But it does not work. I've done "sip debug" and compared a call from a softphone with the call from asterisk. It seems asterist is getting the CallerID wrong. The softphone says ^^^ From: Me <sip:31137110134@budgetphone.nl>; ^^^ but astersi instead says: ^^^ From: "31137110134@budgetphone.nl" <sip:asterisk@81.171.111.77>^^^ and the server at budgetphones end responds with 404 unknown use |
| 12:15.22 | Henk | r. To i guess that is where I'm doing it wrong. I've tried many things but i cannot get it right. Can anyone help me ? |
| 12:15.51 | nettux | Hello everybody. I am experiencing a problem on a * machine serving about 150 SIP users (type friend) interconnecting them to another asterisk box via IAX. What happens is that many times the calls are dropped after 1 or 2 minutes and in the logs i found this message channel.c: Didn't get a frame from channel: SIP/XXXXX or sometimes IAX/xxxx whenever the call is dropped. What could be causing this? |
| 12:16.46 | tRSS | i have a PRI terminated in my asterisk box. I have agents answering calls. I want my agents to know what number was dialed and the caller id of the person calling? How can i achieve this? |
| 12:17.50 | tRSS | If my agents know what number was called, then only they can give a proper response to the caller. so how can i tell my agents about the called number? |
| 12:17.56 | *** join/#asterisk idpromnut (n=chris@modemcable157.119-82-70.mc.videotron.ca) |
| 12:19.10 | [TK]D-Fender | tRSS : Either give them a pop-up on their computer to relay that info, or you're stuck manipulating the CallerID to incorporate the DID info |
| 12:19.30 | nettux | tRSS maybe you could send a fake caller id forging it with the called number + the caller |
| 12:19.30 | [TK]D-Fender | tRSS : How many different numbers are you looking to track? |
| 12:19.49 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:20.29 | FuriousGeorge | so anyone using trunk now that can comment on its stability? |
| 12:20.36 | tRSS | [TK]D-Fender: i would say around 25. How can I give them a pop, I have seen some stuff using TAPI, but don't look that promising? |
| 12:20.43 | X-Rob | FuriousGeorge, it crashes NO MORE than once. |
| 12:20.45 | X-Rob | per second. |
| 12:20.53 | FuriousGeorge | lol |
| 12:20.59 | nettux | Hello everybody. I am experiencing a problem on a * machine serving about 150 SIP users (type friend) interconnecting them to another asterisk box via IAX. What happens is that many times the calls are dropped after 1 or 2 minutes and in the logs i found this message channel.c: Didn't get a frame from channel: SIP/XXXXX or sometimes IAX/xxxx whenever the call is dropped. What could be causing this? |
| 12:21.39 | [TK]D-Fender | nettux : Don't spam it, we heard you the first time. Be patient as not everyone who might be able to help you is here at any given time. |
| 12:21.55 | nettux | :) ok sorry |
| 12:22.02 | hads|home | FuriousGeorge: Trunk is stable for me on my home/test system |
| 12:22.18 | [TK]D-Fender | tRSS : Many ways. Auto-refreshing page on the agents PC, socket based program listing to AMI messages for agent callouts, etc. |
| 12:22.35 | FuriousGeorge | im gonna supress the urge to put trunk on this system even though metermaid would be perfect for these guys |
| 12:22.56 | FuriousGeorge | everything in the patch appears to be working except the device states, so oh well |
| 12:23.58 | QuAtRo[NL] | Is there someone where i can see a call come in my Asterisk? |
| 12:24.01 | FuriousGeorge | maybe when it goes to beta, should be any day nopw from what ive heard |
| 12:26.05 | Henk | can noone tell me how to get my called ID to say from: "henk <sip:1234@foo.nl>" instead of "henk <sip:asterisk@1.2.3.4> " ?? I've been at it for too many hours now |
| 12:26.27 | [TK]D-Fender | tRSS : But it'd be considerably easier to prefix the CID name with "XX:" to indicate the DID. |
| 12:27.06 | [TK]D-Fender | Henk : pastebin your call file |
| 12:28.34 | trelane | is there a way to show a list of queue's and how many callers are in them from the CLI? |
| 12:29.31 | Henk | [TK]D-Fender, pastbin is slow hang on |
| 12:30.08 | tRSS | [TK]D-Fender, please check my message that I have sent you in private. Would appreciate some help:) |
| 12:30.56 | [TK]D-Fender | trelane : "show queues" |
| 12:31.14 | trelane | [TK]D-Fender, I found it right after I asked, early start this morning and no caffeine |
| 12:31.19 | trelane | thank you :) |
| 12:32.12 | QuAtRo[NL] | [Question] Is there a logfile where i can see exactly when a call come in and how they are handled |
| 12:32.14 | [TK]D-Fender | trelane : My Polycom IP 600's have a MicroBrowser Idle page that shows queue stats @ 10s intervals using that output parsed for readability |
| 12:32.35 | Henk | [TK]D-Fender, --> http://pastebin.com/725252 |
| 12:32.38 | *** join/#asterisk mmmmmToop (n=mmmmToop@firewall.datapro.co.za) |
| 12:32.49 | trelane | [TK]D-Fender, you can't do that! that's useful! :/ |
| 12:32.56 | [TK]D-Fender | :D |
| 12:33.31 | [TK]D-Fender | trelane : I've also jsut set up my own phone for ACD login/out, but have yet to try Bweschke's SVN tree to implement it. |
| 12:35.01 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-84-159.nsw.bigpond.net.au) |
| 12:35.59 | P-NuT | hey all, has anyone overcome their echo problems on the SPA3000? |
| 12:37.08 | [TK]D-Fender | P-NuT : Echo issues are variable based on where you are and which firmware its running. YMMV |
| 12:37.39 | [TK]D-Fender | P-NuT : Its EC routine is less than stellar, but largely effective most of the time. |
| 12:38.42 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 12:39.50 | *** join/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 12:41.38 | P-NuT | yeah |
| 12:41.41 | P-NuT | ok. |
| 12:42.02 | stoffell | is thery any way to keep logging with debug 4 (or whatever) but 'not' logging the manager.c events ? |
| 12:42.41 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 12:42.58 | [TK]D-Fender | Henk : I suggest you let * register to your provider hand have it call a Local channel that will place the call through your registration and therefore gain authentication. |
| 12:44.55 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 12:45.04 | Henk | [TK]D-Fender, I guess I have registered with the provider (I can call my number and then get the demo-woman-voice), "sip show registry" shows that i'm connected. So that is ok right? How whould i make a local channel out of that? |
| 12:47.11 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 12:48.46 | SheriF_WorK | what softphone on linux supports G723 ? |
| 12:49.33 | mmmmmToop | ....you will certainly have to pay for it...& why not use g729? |
| 12:51.35 | SheriF_WorK | mmmmmToop: i'm using a voip device with asterisk network .. |
| 12:51.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:53.52 | [TK]D-Fender | Henk : Set up an exten to do your dialout and call it like "Local/5551212@ContextThathasADialCommandUsingMyProvider" |
| 12:54.25 | [TK]D-Fender | SheriF_WorK : G.723 is nasty on licensing. Good luck. And why that codec at all? |
| 12:54.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:57.14 | *** join/#asterisk alib80 (n=chatzill@196.35.242.16) |
| 12:57.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:58.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:59.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:59.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:59.58 | alib80 | hi all |
| 13:00.10 | SheriF_WorK | [TK]D-Fender: i have a multitech device... which is supports G723 G729 G711.1 U-LAW and A-LAW .. but i can't use ulaw and alaw cuz it's eating lots of bandwidth .. the idea is i want ot use g273 g272 when i call the mvp device .. and ulaw when i call using the ZAP channels .. any idea how to ? |
| 13:00.32 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
| 13:00.51 | [TK]D-Fender | G273? G272? huh?! |
| 13:01.36 | [TK]D-Fender | Just use G.729 |
| 13:01.41 | alib80 | has anyone had the problem whereby their phone start ringing randomly even though no asterisk channels are open? |
| 13:02.14 | alib80 | getting G723 to work legally is a major pain |
| 13:02.20 | [TK]D-Fender | alib80 : Mayeb you could be more specifig about what kind of phone and how it connects to * |
| 13:02.27 | SheriF_WorK | g723 when i call the device and ulaw when i call via zap channels :-D that what i want to do .. but don't know how i think asterisk should use periorites? |
| 13:02.38 | alib80 | swiss voice ip 10s |
| 13:02.55 | alib80 | connects to asterisk registering as a sip device |
| 13:03.01 | AltnTab | 2n analog gate -> TDM400P -> Asterisk with IVR. On incoming call from gsm_gate asterisk don't accept nothing dial to go further |
| 13:03.01 | nettux | has any of you experience with SER+Asterisk? |
| 13:03.21 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 13:03.50 | nettux | I would need a HOWTO or something |
| 13:04.10 | *** join/#asterisk jonnysupersonic (n=jonny@dsl-145-4-170.telkomadsl.co.za) |
| 13:04.13 | alib80 | originally i thought it may be a channel that doesn't close properly when you put the phone down |
| 13:04.26 | alib80 | and its some sort of call back taking place |
| 13:04.37 | [TK]D-Fender | SheriF_WorK : Set up seperate peers with seperate codec lists, or push everything through * and have it transcode |
| 13:04.43 | alib80 | but these phones hadn't been called out on for a few minutes |
| 13:06.31 | alib80 | i'm running asterisk 1.2.8 |
| 13:06.37 | SheriF_WorK | [TK]D-Fender: everyting should be passed through asterisk already . |
| 13:07.08 | [TK]D-Fender | SheriF_WorK : Then user seperate peers to call the gateway based on the codec desired |
| 13:08.38 | SheriF_WorK | [TK]D-Fender: have a link to read about this seperate peers ? |
| 13:08.49 | *** part/#asterisk dudes (n=dudes@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 13:08.49 | SheriF_WorK | i mean a how to :-) or the syntax to do it where i can find it? |
| 13:10.33 | [TK]D-Fender | SheriF_WorK : Nothing to say about it. just make another Peer entry in sip.conf |
| 13:10.43 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 13:11.10 | [TK]D-Fender | SheriF_WorK : know how you did it the first time? COPY & PASTE. Chane the peer name and use that to dial out changing the codec it uses. |
| 13:11.27 | *** join/#asterisk bernardovieira (n=bernardo@c911935d.static.bhz.virtua.com.br) |
| 13:11.57 | Henk | [TK]D-Fender, I'm trying to do what you said but im getting an error saying "chan_local.c:378 local_alloc: No such extension/context 0621232999@to-budgetphone creating local channel" I have a [to-budgetphone] block in the extensions.conf ans extensions reload seems to show it |
| 13:12.11 | iCEBrkr | blah |
| 13:12.16 | iCEBrkr | BLAH I SAY |
| 13:12.20 | iCEBrkr | oh.. and moosepenis! |
| 13:12.31 | _problem_ | [TK]D-Fender: do u have any information for me ? |
| 13:12.33 | [TK]D-Fender | Henk : Pastebin your dialplan. |
| 13:12.34 | Henk | [TK]D-Fender, --> http://pastebin.com/725295 |
| 13:12.35 | [TK]D-Fender | ~pb |
| 13:12.38 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 13:12.45 | [TK]D-Fender | _problem_ : Nope... stumped on your problem from the other day |
| 13:13.03 | iCEBrkr | Teamkillah |
| 13:13.05 | iCEBrkr | err I mean [TK]D-Fender |
| 13:13.11 | _problem_ | my mysql cdr is getting worst day by day bcuz of that |
| 13:13.24 | [TK]D-Fender | iCEBrkr : Someone guessed it right! ;) |
| 13:13.28 | iCEBrkr | :P |
| 13:13.31 | lunk | postgres never chokes like mysql does |
| 13:13.34 | [TK]D-Fender | iCEBrkr : First tim in YEARS |
| 13:13.42 | iCEBrkr | Really? |
| 13:13.50 | SheriF_WorK | [TK]D-Fender: how to tell it to use a current codec in dialling out ? i'm using disallow =all allow = g723 in the multitech peer section on sip.conf |
| 13:13.57 | iCEBrkr | I've always said that in my head when I see your nick |
| 13:14.19 | iCEBrkr | hrrm, freshair break. well, smoke break for others.. |
| 13:14.20 | iCEBrkr | BBIAB |
| 13:14.50 | [TK]D-Fender | iCEBrkr : Yup.... It started when a friend and I were playing a HL mod in clsoe quarters and were little "trigger happy" and took as much from friendly fire as enemy fire (See: US Army). So jokingly named our clan after it :) |
| 13:15.08 | [TK]D-Fender | SheriF_WorK : Can't use G.723 with * now can you? |
| 13:15.22 | [TK]D-Fender | SheriF_WorK : But yes, thats the method to do it. |
| 13:15.30 | SheriF_WorK | no can't :-) |
| 13:16.06 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.13 | [TK]D-Fender | Henk : I don't see an exten to handle 0621232999 in there either :) Time for coffee..... |
| 13:16.29 | *** part/#asterisk Meaty` (n=meaty3@66.254.41.11) |
| 13:18.25 | Henk | [TK]D-Fender, do i need to specify each number i want to call in the extensions file?? That is not possible could be anything. |
| 13:18.51 | Henk | http://pastebin.ca/69050 <-- much faster |
| 13:18.59 | [TK]D-Fender | Henk : You need to rethink things a little :) |
| 13:19.38 | [TK]D-Fender | Henk : You're so close but failing to realize the basics of contexts & extens ..... |
| 13:19.43 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 13:19.57 | [TK]D-Fender | Henk : tip : your problem is ...... "s" ;) |
| 13:20.33 | Henk | [TK]D-Fender, sorry i'm very new at asterisk its all pretty fuzzy to me. Let me explain: I want to record a message and have asterisk call a number (in this case my cellphone but could be anything) and than playback that recording to the person answering the line. |
| 13:20.55 | [TK]D-Fender | Henk : (and your dial command isn't using your peer entry, but thats cleary the NEXT problem you'll encounter) |
| 13:21.11 | [TK]D-Fender | Henk : *sigh* |
| 13:21.14 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 13:21.25 | [TK]D-Fender | Henk : exten => _X. <------------------------------------ |
| 13:22.08 | *** join/#asterisk speedwagon (n=Ariel@70.46.87.158) |
| 13:22.13 | [TK]D-Fender | REO! |
| 13:31.35 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 13:33.06 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 13:33.52 | eBody | what's the difference between a SIP and a HUD?? |
| 13:34.03 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.99.Dial1.SanJose1.Level3.net) |
| 13:36.25 | *** part/#asterisk joelsolanki (n=jnsolank@202.160.161.94) |
| 13:37.28 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 13:37.30 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:37.33 | [TK]D-Fender | eBody : Define "HUD". |
| 13:39.24 | eBody | on trixbox, they have the HUD manager |
| 13:39.32 | eBody | and i'm trying to get HUDlite to work |
| 13:39.41 | eBody | but it's just like a SIP softphone |
| 13:39.47 | eBody | what'd the difference between the two??? |
| 13:40.10 | Henk | [TK]D-Fender, --->> http://pastebin.ca/69061 <<< i think i'm getting closer |
| 13:40.43 | Henk | [TK]D-Fender, but still not there. you said something about not using the 'peer' |
| 13:40.46 | [TK]D-Fender | Henk : Yup, much. You sure you want to strip off that leading "0"? ;) |
| 13:41.20 | Henk | no i dont want to strip that of. (i've add one to stop it from loosing the 0) |
| 13:41.24 | [TK]D-Fender | Henk : In your last one the dial didn't say "@budgetphone.nl", it said @ and your DID number.... |
| 13:42.18 | Henk | was that better? |
| 13:42.31 | [TK]D-Fender | Henk : And your peer entry name is "budgetphone", not "budgetphone.nl" |
| 13:42.41 | [TK]D-Fender | Henk : ALMOST got it :) |
| 13:42.55 | Henk | ;) |
| 13:43.02 | [TK]D-Fender | eBody : What does "just like a SIP phone" imply? |
| 13:43.51 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.99.Dial1.SanJose1.Level3.net) |
| 13:44.03 | iq | Good Morning |
| 13:45.30 | Hmmhesays | woohoo my EQ shipped |
| 13:45.38 | eBody | [TK]D-Fender, that i installed this software phone called xlite and this HUDclient called HUDlite is just the same thing |
| 13:45.50 | alib80 | hi all does anyone know why sip phones would start ringing without a call being routed to them or maybe once a call has been put doen? |
| 13:45.53 | [TK]D-Fender | Henk : another tip : you made this new context to support your call file. Don't pass it a "padded # just to strip out the leading digit later. Thats masochistic and confusing :) |
| 13:46.11 | [TK]D-Fender | eBody : Well then I guess its just another SIP client. |
| 13:46.25 | Hmmhesays | http://www.musiciansfriend.com/product/Behringer-FBQ3102-Ultragraph-PRO-EQ?sku=182487 check that baby out |
| 13:46.49 | [TK]D-Fender | Hmmhesays : What a steal....\ |
| 13:46.56 | Hmmhesays | pretty good deal |
| 13:47.21 | [TK]D-Fender | Hmmhesays : Yeah, Behringer is a serious bang-for-buck company... |
| 13:47.37 | tzanger | Hmmhesays: can't see it |
| 13:47.43 | Hmmhesays | yep |
| 13:47.50 | Hmmhesays | i bought a bunch of behringer stuff yesterday |
| 13:48.09 | Hmmhesays | http://www.musiciansfriend.com/product/Behringer-Eurodesk-SL3242FXPRO-Mixer?sku=631246 that too |
| 13:49.09 | [TK]D-Fender | Hmmhesays : Holy crap, midi controllers have dropped through the floor... |
| 13:49.20 | Hmmhesays | oh? |
| 13:50.02 | Hmmhesays | ok I'll bite, what the hell do you need a midi controller for |
| 13:50.13 | [TK]D-Fender | Hmmhesays : Behringer has a really nice one that'd be perfect for rack FX processor control. |
| 13:50.35 | Henk | [TK]D-Fender, what is stripping of that 0 ? |
| 13:50.38 | [TK]D-Fender | Hmmhesays : I used to use a Boss VF-1 half-rack digital multiFX processor for stage work |
| 13:50.44 | Hmmhesays | ahh cool |
| 13:50.49 | [TK]D-Fender | Henk " {EXTEN:1} |
| 13:50.59 | Hmmhesays | i'm kind of a bare bones guy... I have distortion, more distortion and clean are what I use most |
| 13:51.02 | [TK]D-Fender | Hmmhesays : It has since fried, and I'm on a Boss GT-8 |
| 13:51.33 | Hmmhesays | yeah I remember talking about that one day, I was thinking of going with a podxt live... but i've found... when I have effects, I don't use them |
| 13:51.48 | [TK]D-Fender | Hmmhesays : I do more solo / intrumental work and I like my chorus / flanger / reverb / compression. I'm nearly as processed as Ashley Simpson ;) |
| 13:51.55 | Hmmhesays | LOL |
| 13:52.19 | Hmmhesays | yeah, I don't need to be very processed for what I do |
| 13:52.20 | [TK]D-Fender | Hmmhesays : One I've been looking at for about 2 years : http://www.musiciansfriend.com/product/MAudio-Keystation-Pro-88-MIDI-Controller?sku=709203 |
| 13:52.28 | Henk | [TK]D-Fender, ok made that a exten:0 |
| 13:52.40 | *** join/#asterisk sevard (i=kynan@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 13:52.43 | [TK]D-Fender | Henk : Just dits the : altogether |
| 13:52.48 | [TK]D-Fender | ditch* |
| 13:52.59 | sevard | [TK]D-Fender: Do you do line apperance |
| 13:53.11 | Henk | xten => _X.,4,Dial(SIP/${EXTEN}@budgetphone) << <like that ? |
| 13:53.15 | [TK]D-Fender | sevard : yup. My home has all 3 in use |
| 13:53.20 | Hmmhesays | [TK]D-Fender looks pretty sweet |
| 13:53.25 | sevard | [TK]D-Fender: how do you know? |
| 13:53.33 | Henk | or do i need the .nl ? |
| 13:53.36 | [TK]D-Fender | exten => _X.,4,Dial(SIP/budgetphone/${EXTEN}) |
| 13:54.02 | [TK]D-Fender | Henk : No, the peer entry in sip.conf takes care of the connection details just like ti does when you dial it from a phone on your system. |
| 13:54.11 | [TK]D-Fender | Henk : A call is a call is a call.... |
| 13:54.21 | [TK]D-Fender | sevard : What do you mean "how do I know"? |
| 13:54.36 | sevard | [TK]D-Fender: how do you know if all the lines are in use? |
| 13:54.47 | sevard | how does a SIP client know there's a trunk available? |
| 13:54.48 | Henk | chan_sip.c:1398 create_addr: No such host: budgetphone |
| 13:56.01 | [TK]D-Fender | sevard : You need to seperate the concept of "line appearances" from "trunk connections". |
| 13:56.16 | [TK]D-Fender | sevard : "line appearances" are seperate registrations to a SIP server. |
| 13:57.02 | sevard | [TK]D-Fender: I understand both. But if you have 2 registered trunks and 5 sip phones or atas how do they know that they can dial eachother but not out to the pstn until one of them hangs up |
| 13:57.33 | sevard | i was thinking about this problem and researched a bit but didn't find anything |
| 13:57.53 | file | the phones don't care, they call your server and then the server does whatever (ie: reject the call because all outbound lines are in use) |
| 13:58.22 | [TK]D-Fender | ... what file said. :) |
| 13:58.59 | sevard | file: right.. but a traditional pbx will show all the phones that line 1 is in use and you can use line two |
| 13:59.05 | sevard | file: at least PBXs that I know of. |
| 13:59.39 | file | that's a key system |
| 13:59.42 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 13:59.48 | sevard | file: learn me paw |
| 14:00.05 | file | key system does not a PBX make |
| 14:00.09 | [TK]D-Fender | sevard : Don't forget that typically you don't just "pull a line" on a SIP phone, you need to actually dial a number to get it. Its difficult to replicate "key system" methodology.... |
| 14:00.18 | Henk | [TK]D-Fender, i'm still not able to use 'just' budgetphone (although it says so in the sip.conf file) but budgetphone.nl does not get me anywhere either (i get the wrong sip-user error) |
| 14:00.49 | [TK]D-Fender | Henk : pastebin your new dial-plan, sip.conf entries, and CLI output of the attempt. |
| 14:00.50 | MattH | I've tried asterisk 1.2.6 and 1.2.9.1. I'm using native music on hold. When I use mp3s, the volume is nice an acceptable. When I use wave files, the volume is far too loud, even when the volume on the two files is the same. Thoughts? |
| 14:03.04 | X-Rob | MattH, yeah, it's an issue with the way they're decoded. There's no way to adjust the volume. use sox to reduce the volume of the wav files |
| 14:03.25 | sevard | [TK]D-Fender: Alright, I understand that. .. have you ever run into that problem though? I can only think how frustrating it would be to try and dial out in a 3 person household with two other people on the line but you can't and you won't know when they've hung up so you can dial out |
| 14:04.32 | *** join/#asterisk Borgon (n=l3orgon@host-69-59-103-160.nctv.com) |
| 14:05.05 | Borgon | when it comes to wireless adapters which is better for a good signal... low or high dm? i get 70 or 86 on left or right of window. |
| 14:05.12 | *** join/#asterisk __undef (n=jochum@213.30.245.34) |
| 14:05.14 | __undef | hi |
| 14:05.33 | Henk | [TK]D-Fender, do you want the version with .nl or without .nl (the second, cli complains about not finding the host) |
| 14:05.40 | Borgon | so -70 is better than 086 |
| 14:05.44 | Borgon | 086 |
| 14:05.48 | Borgon | -86 |
| 14:06.09 | sevard | -70db is better than -86db, but check your link quality, that matters more. |
| 14:06.51 | [TK]D-Fender | sevard : YOU CAN SCRIPTS A "CALL-BACK" FEATURE IF YOU REALLY FEEL LIKE IT... |
| 14:07.00 | [TK]D-Fender | Henk : wITHOUT |
| 14:07.04 | sevard | [TK]D-Fender: REALLY |
| 14:07.09 | sevard | [TK]D-Fender: TWO LINES NOW |
| 14:07.16 | Hmmhesays | I DON'T KNOW WHY WE'RE YELLING |
| 14:07.20 | Hmmhesays | LOUD NOISES |
| 14:07.23 | Borgon | appreciate it |
| 14:07.34 | sevard | BECAUSE [TK]D-Fender DOESN'T KNOW HIS CAPS LOCK LED EXISTED |
| 14:07.36 | [TK]D-Fender | sevard : I work in caps-lock. Forgive it or help yourself :) |
| 14:07.43 | sevard | S/DOESN'T/DIDN'T/G |
| 14:07.58 | sevard | why the hell on god's mother fucking green earth do you work in caps lock |
| 14:08.06 | [TK]D-Fender | sevard : my keyboard tray is actually usually pushed in just enough to hide them :) |
| 14:08.07 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 14:08.26 | sevard | are you the weirdo that uses MC? |
| 14:08.27 | [TK]D-Fender | sevard : Because I work a lot in fucking GREEN SCREEN (5250) |
| 14:08.49 | sevard | GREEN? |
| 14:08.54 | sevard | MC is blue iirc |
| 14:09.32 | [TK]D-Fender | sevard : 5250 <- Pay attention. I never said Linux. |
| 14:09.42 | Henk | [TK]D-Fender, ->> http://pastebin.ca/69073 |
| 14:09.53 | [TK]D-Fender | sevard : I work on an AS/400 running J.D.Edwards |
| 14:09.58 | sevard | what the crap is any of that |
| 14:10.00 | __undef | the same question as yesterday...can anyone tell me, which versions of misdn, the kernel and asterisk are known to work together? |
| 14:10.07 | sevard | 5250 and as/400 |
| 14:10.08 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 14:10.08 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:10.09 | [TK]D-Fender | sevard : Welcome to "The Real World" |
| 14:10.19 | MattH | x-rob: so basically I need to, when using wav or ulaw, decreaes the volume even more then when using mp3? |
| 14:10.25 | [TK]D-Fender | sevard : PC's aren't the entire universe you know.... |
| 14:10.30 | X-Rob | MattH, yeah |
| 14:10.40 | sevard | [TK]D-Fender: no shit, but i've never heard of this platform. tell me more |
| 14:10.45 | FaithX | __undef: What bri card? |
| 14:10.56 | _problem_ | __undef: kernel-2.6.12 and lastest install-misdn can work |
| 14:10.58 | __undef | FaithX: acer surfpci |
| 14:11.02 | *** join/#asterisk nothinman (i=shakey@aczs128.neoplus.adsl.tpnet.pl) |
| 14:11.06 | MattH | x-rob: is there any reason that asterisk might become unstable when doing native-moh mp3 as opposed to wav |
| 14:11.15 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 14:11.15 | nothinman | hello (-: |
| 14:11.24 | FaithX | __undef: What bri card? Not what brand. |
| 14:11.34 | X-Rob | not that I've noticed. freepbx has been using native moh for about 4 months now, I'm sure we would have had reports of it crashing * by now. |
| 14:11.37 | nothinman | more questions ;) |
| 14:12.03 | FaithX | __undef: chipset... |
| 14:12.08 | FaithX | Hi X-Rob |
| 14:12.10 | __undef | FaithX: cologne ;) |
| 14:12.26 | X-Rob | lo FaithX |
| 14:12.37 | nothinman | doesd anyone know wheter or not it is possible to check who answered the call when executing Dial(SIP/123&SIP/234&SIP/111)? |
| 14:12.49 | FaithX | __undef: then you should use zaphfc for sure |
| 14:13.23 | nothinman | Dial() probably sets some variable... |
| 14:13.48 | __undef | FaithX: hmm... will I be able to use the pri interface, too? |
| 14:14.04 | nothinman | but I don't know two things: what variable, and does it set it after the call is finished or before? |
| 14:14.23 | __undef | FaithX: someone wrote on a forum that it isn't possible to use pri and bri together when using zaphfc |
| 14:15.05 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:15.13 | Dr-Linux | what's ael module name? |
| 14:15.25 | FaithX | Hmmm.... I don't know about that |
| 14:15.43 | FaithX | did you try? |
| 14:15.49 | sevard | okayu then |
| 14:15.49 | sevard | bbl |
| 14:16.22 | ghenry | can you setip SIP users in any better way then listing lots of users in sip.conf, just getting a bit boring ;-) |
| 14:16.57 | [TK]D-Fender | Henk : Maybe do a "sip reload" not sure it took... it should find it. Something is very wrong |
| 14:17.11 | heath__ | ghenry, search for realtime |
| 14:17.17 | __undef | FaithX: i haven't been able to get zaphfc to compile, yet |
| 14:17.19 | [TK]D-Fender | sevard : Go to www.ibm.com and look up "iSeries mid-range" |
| 14:17.23 | ghenry | ah, will do heath__ |
| 14:17.47 | FaithX | X-Rob: see that! |
| 14:17.57 | X-Rob | see what? |
| 14:17.59 | Henk | [TK]D-Fender, could this be it: un 22 14:18:20 WARNING[30353]: acl.c:197 ast_get_ip: Unable to lookup 'budgetphone.nl' |
| 14:18.06 | X-Rob | I'm not watching the soccer yet, if that's what you're asking |
| 14:18.16 | Hmmhesays | i'm watching stargate |
| 14:18.20 | Hmmhesays | sg1 |
| 14:18.22 | FaithX | Can't get zaphfc to compile |
| 14:18.25 | [TK]D-Fender | Henk : Ok, time to fix your peer setup, your call file and dial plan seem good. |
| 14:18.30 | Henk | [TK]D-Fender, everything else seems ok from than on |
| 14:18.38 | [TK]D-Fender | Henk : Go verify with your provider as to what you should be using. |
| 14:19.07 | *** join/#asterisk Admin_OS (n=unix@porthos.sodisa.com.br) |
| 14:19.16 | Henk | (remember i CAN call this number and het the asterisk demo with these settings, and i am registered) |
| 14:20.20 | mover | how i can rewrite the TO header on an asterisk outbound call? |
| 14:20.21 | __undef | zaphfc aborts with "struct zt_chan has no member named bytes2transmit" |
| 14:20.51 | Henk | [TK]D-Fender, ok I changed it to "sip.budgetphone.nl" wich seems better i'm now getting different (non working though) results |
| 14:21.39 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 14:21.58 | [TK]D-Fender | Henk : you are almost there. keep at it. |
| 14:22.15 | [TK]D-Fender | Henk : You've never successfully dialed OUT before? |
| 14:22.39 | Henk | [TK]D-Fender, IT RINGING!!!! OMG |
| 14:22.49 | Hmmhesays | man i hate it when the hot chicks get taken over by the ga'ould |
| 14:24.27 | [TK]D-Fender | Hmmhesays : I never watched... a friend of mine has them ALL on XviD. Thats each series related to (and 100 other shows.... its kinda scary). Maybe I should start watching them :) |
| 14:24.40 | Hmmhesays | yeah its pretty good |
| 14:25.33 | Henk | [TK]D-Fender, it seems to have left a message in my voicemailbox ;) |
| 14:26.10 | Hmmhesays | ok trying to build buildroot one more time |
| 14:27.00 | [TK]D-Fender | heck : Success |
| 14:27.02 | __undef | okay, the bristuff 0.3 beta seems to compile... |
| 14:27.54 | Henk | [TK]D-Fender, no this was the first time to ever dial out. there are no phones connected to the server. I'ts just a server that one day will do system-to-phone stuff like calling me if nagios is complaining and i appear to be away from the computer etc |
| 14:28.13 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 14:28.18 | [TK]D-Fender | henk : Cool stuff. What generates that call file? |
| 14:28.26 | FaithX | __undef: you have the * source in place and you have and have patched it? I think * needs to be patched for bristuff |
| 14:28.36 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:28.38 | [TK]D-Fender | Henk : Oh, you mean this is a preliminary test for later use? |
| 14:28.55 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:28.57 | __undef | FaithX: i know...asterisk is compiling right now |
| 14:29.14 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 14:29.15 | stoffell | __undef, watch out with bristuff beta's, only the last one is a good one (some have an isdn hangup bug, could lead to $$$$ phone bills) |
| 14:29.47 | __undef | stoffell: whoops :) |
| 14:29.58 | stoffell | yeah, indeed:) |
| 14:30.17 | __undef | stoffell: the bri cards are internal, fortunately ;) |
| 14:31.17 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 14:31.19 | stoffell | yeah, most are, that doesn't matter :) the infamous *hangup bug* was introduced in a previous bristuff version :) |
| 14:31.51 | paryl | with an asterisk system behind a firewall, and a sip phone in a remote location behind a DSL router... i have both firewalls opened to port 5060 and 10000-20000. when i make a call from the sip phone, i can call extensions on the asterisk box, leave voicemails, check voicemail, and make outgoing calls. but ONLY calls to other sip phones connect without and audio |
| 14:32.31 | eBody | asterisk .net is used to create softphone apps right, not server apps?? |
| 14:33.15 | Hmmhesays | wow this duron 950mhz is rockin |
| 14:33.35 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 14:33.38 | paryl | any idea why only SIP-to-SIP calls would fail? |
| 14:34.03 | eBody | paryl, any error given?? |
| 14:34.08 | FaithX | paryl: external? |
| 14:34.31 | paryl | they're calls from a remote SIP phone to local SIP phone |
| 14:34.44 | FaithX | paryl: codec? |
| 14:34.57 | paryl | eBody: not so far... the phones ring, etc... it's just that when the call gets connected there's no audio |
| 14:35.00 | paryl | FaithX: gsm |
| 14:35.01 | FaithX | read the log |
| 14:35.13 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:35.19 | FaithX | paryl: no audio either way? |
| 14:35.50 | paryl | FaithX: right. but audio works fine calling voicemail, and calling out on the local exchange |
| 14:36.09 | eBody | paryl, that sounds like a sound card or headset prob |
| 14:36.20 | eBody | asterisk .net is used to create softphone apps right, not server apps?? |
| 14:36.25 | FaithX | What about remote sip to local exchange |
| 14:36.30 | paryl | eBody: it's not a softphone... it's a Polycom 501 |
| 14:37.01 | paryl | FaithX: yes, that's what i mean... it works perfectly. only calls to SIP phones are a problem |
| 14:37.18 | [TK]D-Fender | paryl : So You have * and one phone behind a router. They work together just fine. Its an OUTSIDE phone that doesn't get audio? |
| 14:39.42 | paryl | [TK]D-Fender: i have about 35 SIP phones on the local asterisk server's network, which all work fine. the remote SIP phone can call VM and make outgoing calls through the * box, and audio is fine. only calls between the internal SIP phones and the remote SIP phone are having issues |
| 14:39.44 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 14:40.12 | paryl | i hear the ring, etc... but once the call is connected, it's dead air |
| 14:40.16 | [TK]D-Fender | paryl : "canreinvite=no" for the outside phone. |
| 14:40.24 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 14:40.33 | paryl | [TK]D-Fender: in sip.conf? |
| 14:40.37 | [TK]D-Fender | paryl : Which I typically suggest globally, but I'll let it slide for now. |
| 14:40.39 | Borgon | paryl: not to be noisy but what type of work requires 35 sip fones |
| 14:40.40 | [TK]D-Fender | paryl : yup |
| 14:40.44 | funxion | has anyone seen this before? |
| 14:40.45 | funxion | Jun 22 08:22:16 WARNING[3477]: Identifier 1, identifier_type 1 not found in identifier list |
| 14:40.45 | funxion | Jun 22 08:22:16 WARNING[3477]: aMYSQL_query: Invalid connection identifier 1 passed in aMYSQL_query |
| 14:40.52 | *** join/#asterisk pnlarsson (n=niklas@c83-248-7-150.bredband.comhem.se) |
| 14:40.58 | funxion | this happens after a mysql query |
| 14:41.03 | paryl | [TK]D-Fender: ok, i'll give it a try! thanks! |
| 14:41.13 | paryl | Borgon: uuum... an office? :) |
| 14:42.14 | [TK]D-Fender | Borgon : OMGZ, * can be used in a corporate environment?!?!?! OH NOES! |
| 14:42.33 | *** join/#asterisk jpablo_ (n=jpablo@200.94.130.194) |
| 14:43.31 | paryl | [TK]D-Fender: hahahaha :) |
| 14:44.00 | paryl | [TK]D-Fender: when you say you recommend canreinvite=no globally... |
| 14:44.08 | paryl | what does that control? |
| 14:44.19 | jpablo_ | hey people, what would you recommend for a low bw codec, ilbc or gsm? |
| 14:44.24 | Henk | [TK]D-Fender, thank you for your help. I think I got the hang of it now. |
| 14:44.28 | [TK]D-Fender | paryl : means "yeah sure pass all the damned RTP through * box". Why not... its Gigabit uplink to my switches.... |
| 14:44.29 | Henk | bye |
| 14:44.34 | [TK]D-Fender | Henk : ywc |
| 14:45.13 | [TK]D-Fender | paryl : It means the audio stream passes through * since the phones by nature don't know to forge the return IP whichis why audio gets lost on the re-invite. |
| 14:45.43 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 14:46.32 | smackus | ok, so since i am new to svn, tell me if I am doing this right. |
| 14:46.37 | smackus | snv co http://svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/ |
| 14:46.39 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 14:46.52 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 14:46.58 | smackus | I did that from within my "/usr/src/ directory" |
| 14:47.00 | ptinsley | [TK]D-Fender, so what you are saying is that asterisk has real world applications? |
| 14:47.02 | paryl | hrmm... that worked beautifully... except now there's like a 5 second delay in the conversation? |
| 14:47.02 | ptinsley | :) |
| 14:47.28 | TESTER2 | how can I change ringtone of a fxs module (tdm400p) for one extension ? |
| 14:47.46 | Dr-Linux | what's ael module name? |
| 14:47.46 | smackus | then i cd into the polycom_acd_functions directory and do a make && make install? |
| 14:47.47 | [TK]D-Fender | smackus : When you spell SVN properly, sure :) |
| 14:47.54 | funxion | did anyone see my question above? |
| 14:48.02 | smackus | yeah... sorry. |
| 14:48.04 | [TK]D-Fender | ptinsley : I make no such claims! Tis HERESY! |
| 14:48.20 | [TK]D-Fender | paryl : can't account for that.... |
| 14:48.27 | smackus | so I cd into the polycom_acd_functions and do a make && make install, right? |
| 14:49.03 | [TK]D-Fender | smackus : its not jsut Polycom functions... its an entire build of *. |
| 14:49.08 | mover | how i can rewrite the TO header on an asterisk outbound call? |
| 14:49.32 | mover | is there a way? |
| 14:49.57 | ptinsley | [TK]D-Fender, what version did that polycom stuff branch off of? |
| 14:49.58 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 14:50.00 | [TK]D-Fender | smackus : And I'm not sure how uptodate the rest of it is. |
| 14:50.08 | [TK]D-Fender | ptinsley : not a clue. |
| 14:50.10 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 14:50.19 | funxion | anyone well versed with app_addon_sql_mysq |
| 14:50.54 | jbalcomb | Can the Cisco 7940G AutoAnswer via some SIP Header? |
| 14:51.53 | [TK]D-Fender | jbalcomb : I'm pretty sure I've seen it around somewhere. Checked the WIKI yet? |
| 14:52.35 | jbalcomb | [TK]D-Fender yeah, still looking. Just trying the multi-threaded search. ;) |
| 14:53.11 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 14:53.15 | smackus | has anyone here had experience with adding the acd features of the polycom phones using the svn branch? |
| 14:53.19 | TESTER2 | how can I change ring setting of a fxs module (tdm400p) for one extension (set a different ring). Is there a variable to set? |
| 14:53.25 | ptinsley | kinda off topic here but why does sourceforge suck so much? |
| 14:54.00 | [TK]D-Fender | smackus : Like I said its not "adding the features". Its an entire build. You do it jsut you would any other release. |
| 14:54.15 | smackus | ok, so clearify... |
| 14:54.17 | coppice | ptinsley: i can't tell you the reason, but i can confirm your view is accurate |
| 14:54.22 | [TK]D-Fender | smackus : Follow the notes in Mantis entry and just try it |
| 14:54.23 | ptinsley | :) |
| 14:54.43 | ghenry | why after Background, would I be autofallthroughing? |
| 14:54.59 | [TK]D-Fender | ghenry : "autofallthrough=yes" |
| 14:55.17 | ghenry | so it won't wait with that set? |
| 14:55.32 | [TK]D-Fender | ghenry : Thats a rather implicit instruction you know.... |
| 14:55.56 | stoffell | hm, is it good or bad practice to adjust the polycom gains? (in the manual it says "do not alter these values" ..) |
| 14:55.57 | [TK]D-Fender | ghenry : the "new" style of WaitExten does exist.... although WHY I don't know.... |
| 14:56.03 | smackus | [TK]D-Fender: looking for something resembling instructions, and not seeing what I think I should be.... I am on the mantis page. |
| 14:56.08 | ghenry | well, it's not waiting for an exten to be typed [TK]D-Fender |
| 14:56.14 | ghenry | [TK]D-Fender: And that's why? |
| 14:56.25 | [TK]D-Fender | stoffell : I'd suggest leaveing those alone, but allow each phone to remember its lsat settings on all 3 modes. |
| 14:56.38 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 14:56.42 | Bert- | hi again here :) |
| 14:56.42 | ghenry | [TK]D-Fender: If I add, exten => s,n,WaitExten(10), that should be fine? |
| 14:56.46 | [TK]D-Fender | ghenry : Does it still say "autofallthrough=yes" in your extensions.conf? |
| 14:56.57 | ghenry | [TK]D-Fender: aye. |
| 14:57.01 | stoffell | [TK]D-Fender, did that, but still the customer complains of 'low' volume (sometimes), even when set to maximum on the phone |
| 14:57.03 | __undef | hm. now i get ZT_SPANCONFIG failed on span 5: invalid argument (22) *sigh* |
| 14:57.04 | [TK]D-Fender | ghenry : THATS WHY |
| 14:57.21 | Hmmhesays | I think I am in awe |
| 14:57.25 | funxion | has anyone seen this before? |
| 14:57.28 | funxion | Jun 22 08:22:16 WARNING[3477]: Identifier 1, identifier_type 1 not found in identifier list |
| 14:57.28 | Hmmhesays | http://www.cracked.com/modules.php?op=modload&name=News&file=article&sid=445 |
| 14:57.29 | [TK]D-Fender | stoffell : If its the customer, you should tweak your pSTN interface, not the phone. |
| 14:57.31 | funxion | Jun 22 08:22:16 WARNING[3477]: aMYSQL_query: Invalid connection identifier 1 passed in aMYSQL_query |
| 14:57.34 | ghenry | [TK]D-Fender: OKAY!!!! ;-) But it's nice to have as it picks up any typos you do. |
| 14:57.45 | ghenry | [TK]D-Fender: will switch off ;-) |
| 14:58.27 | stoffell | [TK]D-Fender, highering the rxgain on the zapata that is? |
| 14:59.29 | [TK]D-Fender | stoffell : Yup. |
| 14:59.50 | [TK]D-Fender | stoffell : Or txgain. I forget which :) Just start playing and testing with the guy who whines the most :) |
| 15:00.02 | ghenry | thanks [TK]D-Fender works fine. |
| 15:00.10 | stoffell | hehe, okay [TK]D-Fender ;) |
| 15:00.26 | *** join/#asterisk salviadud (n=ralfalfa@201.145.29.99) |
| 15:01.11 | ptinsley | stoffell, ztmonitor can help some with this |
| 15:01.19 | ptinsley | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html |
| 15:01.31 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:01.44 | ptinsley | even if it doesn't sometimes it's just fun to look at :) |
| 15:02.03 | __undef | next problem...zaphfc works, except that it's spitting out lots of messages (zaphfc: dropped audio) |
| 15:02.10 | stoffell | ptinsley, yes, okay, will also play with it a bit ;) |
| 15:03.04 | Zeeek | afternoon |
| 15:03.07 | *** join/#asterisk PakiPenguin (n=AHMAD@linuxpakistan/admin/pakipenguin) |
| 15:03.10 | PakiPenguin | hello there |
| 15:03.26 | PakiPenguin | i have a problem with recording , the recording file i get is very very choppy |
| 15:05.10 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 15:05.20 | *** join/#asterisk notjason (n=notjason@ool-457183bb.dyn.optonline.net) |
| 15:05.28 | eBody | is asterisk using h323 by default?? |
| 15:06.07 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:06.11 | [TK]D-Fender | eBody : No, its in a seperate add-ons package |
| 15:06.35 | PakiPenguin | [TK]D-Fender, any idea about the choppy sound |
| 15:06.39 | jbalcomb | [TK]D-Fender Are you coded in VB, ASP, or Java? |
| 15:06.40 | [TK]D-Fender | eBody : * comes with support for a lot of different protocols in the base. |
| 15:06.49 | eBody | really, what protocol is defaulted?? |
| 15:07.14 | jbalcomb | eBody: TCP/IP |
| 15:07.36 | paryl | [TK]D-Fender: canreinvite=no was a lifesaver. thanks mucho |
| 15:08.04 | eBody | it's not running another voip protocol over tcp/ip? |
| 15:08.38 | [TK]D-Fender | jbalcomb : No, I'm coded in self replicating DNA/RNA |
| 15:09.09 | [TK]D-Fender | eBody : * can do a LOT of things. Depends what you want it to do. |
| 15:09.32 | [TK]D-Fender | eBody : The most popular VoIP protocol is SIP. |
| 15:10.38 | [TK]D-Fender | eBody: Quick lists of what VoIP protocols * can support : SIP, MGCP, SCCP, Unistim, H.323 |
| 15:10.50 | [TK]D-Fender | paryl : ywc |
| 15:11.40 | jbalcomb | [TK]D-Fender Hrmm.. sounds nice and impressive but obviously a bit unstable. |
| 15:11.53 | eBody | hmm, very interesting. then i'm quessing that SIP is the default |
| 15:12.02 | jbalcomb | MGCP stands for MeGaCraP |
| 15:12.03 | eBody | since i'm able to use these sip softphones. |
| 15:12.07 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 15:12.13 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 15:12.43 | jbalcomb | SCCP is silly because people call is skinny but it should be skippy |
| 15:12.56 | ptinsley | i never have figured that one out |
| 15:13.01 | *** join/#asterisk teknoprep (n=teknopre@c-68-83-86-17.hsd1.pa.comcast.net) |
| 15:13.04 | [TK]D-Fender | jbalcomb : No..... I'd never be able to eat PB&J again if they did that... |
| 15:13.04 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 15:13.08 | Bert- | does someone ever linked a Asterisk with a Nextone Softswitch ? |
| 15:13.08 | teknoprep | what is the name of the main .conf file? |
| 15:13.12 | teknoprep | for asterisk? |
| 15:13.22 | file | teknoprep: for what? |
| 15:13.26 | [TK]D-Fender | teknoprep : No such thing as "main" |
| 15:13.30 | TESTER2 | So no way to change the ringtone cadence on a fxs module of a tdm400p? |
| 15:13.35 | Zeeek | main.conf |
| 15:13.36 | teknoprep | configureing it to allow outbound and inbound phone calls |
| 15:13.39 | [TK]D-Fender | teknoprep : This isn't Apache you know.... |
| 15:13.45 | Zeeek | httpd?conf |
| 15:13.49 | [TK]D-Fender | teknoprep : That can take all sorts of files... |
| 15:13.58 | Zeeek | Mein.Kompf |
| 15:14.03 | *** join/#asterisk iq (n=iq@71-215-59-132.omah.qwest.net) |
| 15:14.07 | Zeeek | Ow |
| 15:14.28 | teknoprep | i need 2 things |
| 15:14.28 | Zeeek | there is an argument for saying extensions.conf is the "main" file |
| 15:14.37 | teknoprep | a windows dialer for voip connection to the asterisk box |
| 15:14.49 | teknoprep | sorta like a voip software phone for windows |
| 15:14.54 | file | this is already going down hill |
| 15:14.57 | teknoprep | and a how-to on setup of the conf files |
| 15:14.59 | ptinsley | teknoprep, you will spend most of your time in extensions.conf to route calls around and handle them, for setting up connections to the world your time based on what you are doing will be split between zapata.conf, zip.conf and iax.conf |
| 15:15.01 | Zeeek | most softphones *are* for windows |
| 15:15.13 | ptinsley | zip.conf = sip.conf |
| 15:15.15 | ptinsley | hehe |
| 15:15.19 | teknoprep | ok |
| 15:15.22 | Bert- | nobody here knows Nextone ??? |
| 15:15.26 | Bert- | strange |
| 15:15.27 | Bert- | :) |
| 15:15.59 | Zeeek | teknoprep look at htto=p://www.asterisk-docs.org |
| 15:16.10 | file | jbot: book? |
| 15:16.12 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:16.21 | Zeeek | s/htto=p/http |
| 15:16.27 | [TK]D-Fender | teknoprep : Go read .... THE BOOK |
| 15:16.29 | [TK]D-Fender | ~book |
| 15:16.31 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:16.46 | Zeeek | you can avoid reading the book though. The answer is 42 |
| 15:16.46 | file | I, I won't worry my life away! |
| 15:16.58 | [TK]D-Fender | ptinsley : iax.conf? largely irrelevent :) |
| 15:17.09 | ptinsley | ok, I spend time in there at least |
| 15:17.33 | ptinsley | pbx to pbx or voip provider type stuff |
| 15:17.35 | [TK]D-Fender | SIP = the protocol the rest of the world cares about. |
| 15:17.59 | teknoprep | sooo anyone know of a free softphone that works with asterisk? |
| 15:17.59 | [TK]D-Fender | ptinsley : Not saying it doesn't have its merits... just a question of relevence |
| 15:18.00 | *** join/#asterisk babyju___ (n=babyju@151.202.195.132) |
| 15:18.05 | ptinsley | but yes, I get your point, he will probably not touch it :) |
| 15:18.21 | Zeeek | teknoprep X-Lite IAXPhone SJPhone |
| 15:18.45 | jbalcomb | that book is crap |
| 15:18.49 | Zeeek | teknoprep DIAX |
| 15:19.26 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 15:19.46 | ptinsley | the book != what asterisk actually does in some cases, maybe what it was planned for it to do |
| 15:20.38 | Bert- | ~nextone |
| 15:20.40 | file | Asterisk changes. |
| 15:20.49 | Zeeek | for people who know nothing the book is great |
| 15:20.54 | ptinsley | Yes, but some of the things in the book have NEVER been in the source code |
| 15:20.56 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:21.03 | file | like what? |
| 15:21.05 | Zeeek | it explains the basics of dialplan which is where you spend the most time |
| 15:21.11 | ptinsley | well the way timeframes work for example |
| 15:21.22 | Zeeek | beginners don't care about shit like that |
| 15:21.34 | ptinsley | Zeeek, file asked for an example |
| 15:21.40 | coppice | ptinsley: you do realise it stocked in the fiction section, don't you? :-) |
| 15:21.45 | ptinsley | lol |
| 15:22.00 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:22.24 | __undef | asterisk with zaphfc and a tor2 card really is a PITA... |
| 15:23.05 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:23.08 | ptinsley | i do think the book has merit, don't get me wrong, it has just bitten me in the buttox a couple of times, just a tad bitter :) |
| 15:23.18 | *** join/#asterisk s0lid (n=s0lid@210.213.242.39) |
| 15:23.28 | coppice | they are doing a new one |
| 15:23.45 | Damin | ptinsley: Any book that is published about Open Source products is almost immediately out of date anyway.. |
| 15:24.01 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 15:24.09 | Damin | ptinsley: But * TFOT is an excellent reference to the 1.0 Asterisk releaseses... |
| 15:24.12 | ptinsley | True, a book on open source stuff is pretty much a unicorn in most cases |
| 15:24.26 | heison | does anyone know of a good Asterisk appliance? |
| 15:24.48 | Damin | ptinsley: There are changes between releases, which is why some things are different w/ the current 1.2 code.. |
| 15:24.59 | Damin | heison: Switchvox and/or Fonality |
| 15:25.01 | [TK]D-Fender | heison : Contradiction in terms. |
| 15:25.38 | coppice | maybe the new book will cover 1.2, now that 1.4 is coming out :-) |
| 15:25.50 | heison | thx |
| 15:26.04 | Damin | coppice: Hehehe.. |
| 15:26.15 | *** join/#asterisk apardo (n=apardo@87.217.144.109) |
| 15:26.33 | sticks | /h |
| 15:26.45 | [TK]D-Fender | coppice : Procrastination : The art of keeping up with yesterday. |
| 15:27.18 | coppice | I thought that was the work of a historian |
| 15:27.38 | Zeeek | one thing about books like that is unlike mainstream books, here you can contact the authors easily and tell them things you think are wrong or that need improvement. However, that book was meant as an into and it answers at least 80% of the questions asked here |
| 15:28.24 | [TK]D-Fender | Zeeek : Which is scary because it alreay out there and they jsut aren't reading... it should be far less. |
| 15:28.58 | Zeeek | well, if people say things like "that book is crap" the newbies are less inclined to get off their asses and read |
| 15:29.22 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 15:29.30 | TheCops | ps -aux |
| 15:29.32 | TheCops | oops |
| 15:29.34 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:29.56 | Zeeek | what disappoints me - and this is a general "asterisk community" observation, is that for the last 6 months or so, almost all asterisk channels, mailing lists, irc etc are so negative I have to wonder what' it's all about |
| 15:30.03 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 15:30.22 | coppice | don't most people read when they are on their asses? :-\ |
| 15:30.26 | Zeeek | if any of you ever look at -biz, it's become the worst kind of pissing contest |
| 15:30.36 | CunningPike | Zeeek: Really? I haven't picked that up - but I don't read -biz |
| 15:30.36 | ptinsley | ya, it's pretty rough, I have stopped reading it |
| 15:30.43 | Zeeek | coppice I don't know what we'd do without you :) |
| 15:30.57 | Zeeek | biz is crap :) |
| 15:31.04 | file | coppice: have you put a DSP into your toaster yet? |
| 15:31.10 | Zeeek | no kidding, it's hish noise to sig tho |
| 15:31.26 | coppice | file: not, but lots of muffins |
| 15:31.30 | CunningPike | Zeeek: I find -users good, and this channel is actually heaps better than it was 6 months ago |
| 15:31.32 | file | mmm muffins |
| 15:31.33 | Zeeek | rewrite: no, I'm kidding, but it's high noise-to-signal |
| 15:32.01 | Zeeek | this place was a dream a couple of years ago |
| 15:32.06 | sticks | /leave |
| 15:32.21 | *** part/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 15:32.30 | Zeeek | maybe because there weren't 7000 people asking questions that they'd know the answers to if they read a three page web site |
| 15:32.44 | Zeeek | you want /fart |
| 15:32.45 | CunningPike | Zeeek: Well, take comfort that, as a relative newbie, I don't get the negative vibe..... |
| 15:32.53 | Zeeek | good, glad to hear it |
| 15:33.00 | file | I just sort of block it out, except for the occasional good question |
| 15:33.21 | ptinsley | Zeeek, i think the biggest thing is it's hard for beginners to find the right resource off the bat |
| 15:33.37 | CunningPike | Zeeek: Any OSS project goes through this as it attracts more users from the "mainstream" |
| 15:33.49 | Zeeek | since I don't contribute code, I'm pretty patient with questions (as people were with me) but I won't tolerate someone not going to read at least a wiki page |
| 15:33.59 | Zeeek | CunningPike certainly so |
| 15:34.07 | ptinsley | when you first start digging you don't see alot of references to the book, alot of times all it takes is for someone to point out that they should look |
| 15:34.09 | [TK]D-Fender | ptinsley : Which is why we have it printed on a Louisville Slugger named the "ClueBat" with which we shall pummel them good and proper ;) |
| 15:34.28 | Zeeek | I know I needed some newbie info about FreeBSD once and I asked a stupid (but unfindable) question and they were very kind |
| 15:34.31 | *** join/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 15:34.39 | CunningPike | Zeeek: I tend to take a "fool me once' approach - no question's too dumb, but if I give you a link and you don't read it, have a nice day |
| 15:35.06 | Zeeek | I used to have a macro with five great links for beginners |
| 15:35.09 | ptinsley | i love the "read the archives" answers people give on alot of lists |
| 15:35.14 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:35.26 | Zeeek | by the way, if some one doesn't know what a dialplan is, they need to read something. |
| 15:35.53 | Zeeek | I used to have a direct link to The Dialplan is the heart of asterisk chapter which is easy to understand |
| 15:36.20 | Zeeek | yeah read the archives when the answer is simple (like 42) is a pisser |
| 15:36.42 | Zeeek | except when the asker has been around for two weeks jumping on new innocent people with the same dialplan question |
| 15:36.46 | [TK]D-Fender | CunningPike : Actually... no, plenty of questions are too dumb... means they didn't look ANYWHERE before opening their mouths :) |
| 15:36.49 | ptinsley | especially when you are talking about a list that has a bazillion posts per year |
| 15:36.57 | Zeeek | heh - yeah |
| 15:37.18 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 15:37.24 | Zeeek | Well asking for help on DISA when you don't know the name is legit, but once someone says "go read about DISA" there is no further excuse |
| 15:37.36 | Zeeek | you can't guess "DISA" |
| 15:38.04 | ptinsley | ya, not much chance on that one |
| 15:38.19 | CunningPike | [TK]D-Fender: Some people need to be pointed at what to look at - there's a lot of info out there - some of which is bollocks |
| 15:38.25 | Zeeek | true |
| 15:38.38 | Zeeek | there's also a lot of bollocks out there |
| 15:38.52 | Zeeek | (so few females in the asterisk world) |
| 15:39.07 | Zeeek | so I just came home from Astricon Paris |
| 15:39.43 | Zeeek | where all those same questions were answered... in French ;) |
| 15:40.06 | *** join/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu) |
| 15:41.24 | Zeeek | another tidbit about looking for info is that, because asterisk is still relatively marginal (i.e., eBay isn't grabbing searches with "Find asterisk on eBay") google works very well |
| 15:41.48 | Beighto | sticks: how do you do that generic action "sticks is back"? |
| 15:42.21 | Beighto | yes, that |
| 15:42.23 | ptinsley | man I hate those spam search sites |
| 15:42.28 | elg | can anyone clarify why asterisk is not detecting the peer properly based on the SIP INVITE here http://rafb.net/paste/results/brsDmF59.html |
| 15:42.30 | *** join/#asterisk bmg505 (n=leon@c1-105-2.rndf.isadsl.co.za) |
| 15:42.33 | Zeeek | join #irc |
| 15:42.39 | elg | also there is an INVITE that _does_ detect the peer correctly |
| 15:43.59 | CunningPike | Beighto: My ircproxy used to do it - until I got flamed for it. It's really annoying, apparently |
| 15:44.36 | Beighto | hMmm |
| 15:44.46 | Zeeek | the /me will insert your name at the beginning |
| 15:44.55 | Zeeek | <PROTECTED> |
| 15:45.20 | Zeeek | This tidbit fromt he -biz list support my case: "Honestly, who shit in your cornflakes this morning?" |
| 15:45.40 | Dr-Linux | here one of my client needs 12 DID's from a US sip provider. |
| 15:45.47 | Dr-Linux | here in pakistan |
| 15:45.57 | Zeeek | got several choices |
| 15:46.10 | Zeeek | mixnetworks comes to mind |
| 15:46.17 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 15:46.51 | CunningPike | Zeeek: Wow - that's bad |
| 15:46.56 | Zeeek | what is? |
| 15:47.03 | CunningPike | Zeeek: -biz |
| 15:47.11 | Zeeek | well in a nutshell - |
| 15:47.46 | Zeeek | someone introduced themselves in an awkward manner and sevean barracudsas startned kgnawing at him immediately |
| 15:48.32 | Zeeek | now I want to know, since many of the -biz people offer DID and all, shouldn't they be on the support phgones or developing instead of attacking people with poor expression skills ? |
| 15:49.05 | Zeeek | biz could be an interesting place but I find myself ignoring 99% of the posts |
| 15:49.27 | *** join/#asterisk EinsteinTaylor (n=dtaylor@216.243.100.29) |
| 15:49.36 | ptinsley | you know, the shit in cornflakes, not a good visual |
| 15:49.46 | CunningPike | Ya - any list with that low an SNR is a waste of time |
| 15:50.11 | Zeeek | you also see a lot of ${PROVIDER} is bullshit posts |
| 15:50.41 | Zeeek | for that matter there are a lot of bullshit posts from ${PROVIDERS} as well ;) |
| 15:50.52 | Spy000007 | There's so much shit on -biz because every provider wannabe talks about other providers instead of themselves |
| 15:51.12 | ptinsley | ya, i do like when some of the legit providers will actually get on there and respond to problems and help people |
| 15:51.14 | Zeeek | exactly and they think it makes them look good but the opposite is true, obviously |
| 15:51.17 | Spy000007 | No one just posts about what they're selling without shitting on someone else's product |
| 15:51.29 | Zeeek | yep |
| 15:51.41 | Zeeek | not good - this is what I was saying, what I regret now |
| 15:51.54 | coppice | its called a "product eco-system" :-) |
| 15:52.13 | *** join/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 15:52.18 | Zeeek | I will usually say "I have had good experience with $P" but unless I feel they really screwed me not "They suck" |
| 15:52.19 | EinsteinTaylor | can someone help me with the right terminology so i can read about something i'm trying to do? |
| 15:52.23 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 15:52.27 | EinsteinTaylor | just not sure what to look under yet |
| 15:52.31 | Zeeek | of course EinsteinTaylor |
| 15:52.34 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:52.38 | wasim | EinsteinTaylor: rubber blow up dolls |
| 15:52.42 | EinsteinTaylor | lol :) |
| 15:52.44 | Zeeek | wasim stop it! |
| 15:52.50 | EinsteinTaylor | i love it |
| 15:53.31 | EinsteinTaylor | i have one working server with an ISDN connection to our PSTN switch...i'm building a second asterisk server and i want the stuff to relay through the first one |
| 15:53.41 | EinsteinTaylor | back and forth to the PSTN |
| 15:53.54 | CunningPike | EinsteinTaylor: IAX trunking |
| 15:53.56 | EinsteinTaylor | just not sure what to research |
| 15:54.02 | wasim | EinsteinTaylor: Dial(IAX2/server1) |
| 15:54.03 | EinsteinTaylor | even if it is sip? |
| 15:54.11 | *** join/#asterisk wintix (n=tobias@pegel-neuburg.de) |
| 15:54.25 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:54.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:54.25 | Damin | Hmm.. does anyone have a soundfile of allison saying the name "Ryan"? |
| 15:54.27 | EinsteinTaylor | the actual phones are sip? but it's an IAX trunk between the two? |
| 15:54.33 | Zeeek | http://lists.digium.com/pipermail/asterisk-dev/2003-October/001927.html |
| 15:54.46 | CunningPike | EinsteinTaylor: Phone -> SIP -> Asterisk2 -> IAX trunk -> Asterisk1 |
| 15:55.12 | EinsteinTaylor | ahh...cool...thanks guys...that helps a lot... |
| 15:55.14 | CunningPike | EinsteinTaylor: We are doing this at our place |
| 15:55.20 | CunningPike | EinsteinTaylor: Works a treat |
| 15:55.35 | AltnTab | How to make Read() timeout last longer da a few seconds ? |
| 15:55.40 | EinsteinTaylor | we have about 1000 customers doing voicemail only on server 1 |
| 15:55.41 | Zeeek | EinsteinTaylor there is a short mention of trunking here: http://applications.linux.com/article.pl?sid=04/11/10/1632246&tid=115 |
| 15:55.52 | AltnTab | How to make Read() timeout last, longer than a few seconds ? |
| 15:55.53 | EinsteinTaylor | so i cant be screwing it up while i test other features on the other |
| 15:56.22 | EinsteinTaylor | u guys rock |
| 15:56.23 | CunningPike | EinsteinTaylor: That's exactly what we use our setup for - our test server trunks to our production one |
| 15:56.58 | CunningPike | Yes, we do |
| 15:57.05 | EinsteinTaylor | :) much thanks |
| 15:57.05 | Zeeek | and we will rock you |
| 15:57.13 | EinsteinTaylor | u gotta buy me dinner first |
| 15:57.27 | CunningPike | Settle |
| 15:57.50 | Zeeek | is anyone here using asterisk as a SOHO pbx? |
| 15:57.52 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:58.35 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 15:58.47 | Spy000007 | Zeeek: No, that's not what Asterisk is for. |
| 15:59.05 | Zeeek | that's what I use it for |
| 15:59.07 | CunningPike | Spy000007: ??? |
| 15:59.32 | Zeeek | I was asking so I could steal^H^H^H borrow some ideas |
| 15:59.52 | userdefined | Zeeek: i am (in addition to getting it working at work) |
| 15:59.52 | Zeeek | as a toolkit it is for whatever you want to use it for |
| 16:00.06 | userdefined | Zeeek: but just getting started on all of this, so likely not useful ;-) |
| 16:00.06 | CunningPike | Spy000007: Lots of people would disagree..... |
| 16:00.12 | Zeeek | userdefined what neat things have you done that an ordinary pbx can't do? |
| 16:00.16 | ptinsley | Spy000007, that seems like a pretty short sighted statement |
| 16:00.29 | Zeeek | userdefined OR, what would you LIKE to do that other... |
| 16:00.34 | Spy000007 | I'm just kidding... haha |
| 16:01.01 | Zeeek | IIRC, astersik was actually invented for exactly that |
| 16:01.01 | CunningPike | ~lart Spy000007 |
| 16:01.13 | Zeeek | shit, typo city |
| 16:01.13 | [TK]D-Fender | Zeeek : Several of my clients do |
| 16:01.20 | ptinsley | Spy000007, damn irc and it's lack of emotion ;) |
| 16:01.27 | CunningPike | lol |
| 16:01.40 | userdefined | Zeeek: i don't have any experience with an 'ordinary pbx' to compare with ... but i'd guess the biggest thing is managing sip. |
| 16:01.42 | Zeeek | so TK, what neat stuff do they do that an ordinary pbx can't? |
| 16:01.57 | Spy000007 | Asterisk was invented to prank call radio stations |
| 16:02.03 | userdefined | afik a normal pbx doesn't do sip ? |
| 16:02.04 | Zeeek | I'll be happy to tell you what we do |
| 16:02.14 | Zeeek | SIP isn't really a thing though |
| 16:02.19 | [TK]D-Fender | Zeeek : internal / external Follow-me, SpanDSP faxing. VM alerts sent to pagers, IVR callbacks. |
| 16:02.25 | Zeeek | it's just a way to word the data |
| 16:02.42 | mountainm2k | Polycom IP301 -- how to totally reset it? It gives me "Config file is error 0x4020" then reboots |
| 16:02.46 | Zeeek | TK so far I've done all that and yes, that's my answer too |
| 16:02.50 | CunningPike | Zeeek: Last Astricon, there was a chap who had hooked it up to Mister House, to control X10 devices by dialing in, and a whole bunch of other stuff |
| 16:03.15 | CunningPike | mountainm2k: Won't help - you need to fix the error in your config file |
| 16:03.18 | [TK]D-Fender | mountainm2k : sounds like you botched just a basica config file. Do other phones work? |
| 16:03.27 | Zeeek | CunningPike there is a guy now in Astricon Paris who has a lot of that stuff, even sells the controller cards |
| 16:03.33 | mountainm2k | Only have the one phone... |
| 16:03.33 | ptinsley | mountainm2k, is it pulling that from a boot server? |
| 16:03.46 | mountainm2k | It says "Can't find boot server, using previous config" |
| 16:03.52 | mountainm2k | even though the FTP server _is_ there |
| 16:03.54 | ptinsley | oh ya you did a good one |
| 16:03.57 | [TK]D-Fender | mountainm2k : Ok, then reset your provisioning files to the samples that came with your firmware and start over. |
| 16:04.01 | mountainm2k | and I checked in the bootrom the settings are rihg |
| 16:04.02 | CunningPike | Zeeek: Cool - Ed Guy? |
| 16:04.02 | Zeeek | mountainm2k - disconnect it from the network completely and reboot factory init |
| 16:04.03 | ptinsley | do you have the PlcmSpIp user/pass setup |
| 16:04.25 | Zeeek | CunningPike no he's a local but same idea - he's a ham radio operator |
| 16:04.32 | mountainm2k | tyes |
| 16:04.33 | Zeeek | as I was a hundred years ago ;) |
| 16:04.34 | CunningPike | Zeeek: As am I |
| 16:04.39 | mountainm2k | as am I |
| 16:04.40 | mountainm2k | heh |
| 16:04.43 | Zeeek | naw!!!! |
| 16:04.46 | wasim | we aren't allowed ham here |
| 16:04.48 | Zeeek | welll shit boys... |
| 16:04.50 | CunningPike | Zeeek: VA7IRL |
| 16:04.51 | Zeeek | CQ CQ CQ |
| 16:05.01 | mountainm2k | de KB0KZR |
| 16:05.01 | Zeeek | former W0DBJ |
| 16:05.10 | mountainm2k | at any rate -- how to factory INIT it? |
| 16:05.10 | Zeeek | join asterham |
| 16:05.16 | userdefined | Zeeek: my plans are to have * manage calls to/from FWD/iptel/INOC-DBA |
| 16:05.21 | mountainm2k | I pulled the network cable |
| 16:05.27 | Zeeek | look in the book - I forgot the three keys :) |
| 16:05.35 | CunningPike | mountainm2k: There's a key sequence - it's in the manual |
| 16:05.36 | mountainm2k | 4, 6, 8, and * |
| 16:05.41 | mountainm2k | but that doesn't seem to do it... |
| 16:05.47 | userdefined | Zeeek: additionally, once i switch from vonage to ${something-else} it'll manage PSTN via them |
| 16:05.54 | Zeeek | first of all turn it off for like 5 minutes |
| 16:06.11 | mountainm2k | it was unplugged all night |
| 16:06.20 | *** part/#asterisk smackus (n=smackus@63.149.122.94) |
| 16:06.31 | Zeeek | userdefined we have accounts with at least 15 providers and we switch around depending on whether we call cell, LD, inhouse etc |
| 16:06.35 | ptinsley | default password is 456 |
| 16:06.45 | Zeeek | mountainm2k in Norway that's 5 minutes! |
| 16:07.03 | mountainm2k | heh |
| 16:07.14 | Zeeek | actually isn't there a function key in the reboot sequence? |
| 16:07.24 | Zeeek | like 456 MESSAGE ? |
| 16:07.43 | Zeeek | you're gonna make me open the PDF, right? |
| 16:07.46 | mountainm2k | dunno -- everybody said the Grandstream sucked, which after using this, I agree |
| 16:07.55 | mountainm2k | heh, but I *HAD* to monkey with the provisioning stuff, |
| 16:07.56 | mountainm2k | lol |
| 16:08.01 | *** join/#asterisk mafkees (n=michiel@vanbaak.xs4all.nl) |
| 16:08.05 | userdefined | Zeeek: yep, that sounds pretty similar to my grand scheme also. |
| 16:08.12 | mafkees | heya all |
| 16:08.22 | ptinsley | it depends on the polycom model, for just reboot it's volume up + volume down + hold + DND |
| 16:08.31 | ptinsley | but he needs the admin interface which is 4 + 6 + 8 + * |
| 16:09.07 | mountainm2k | so is there any way to clear the config file without having the application running? |
| 16:09.18 | mountainm2k | because the application tries to load the config, and fails, and then it reboots again |
| 16:09.21 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 16:09.37 | ptinsley | i am trying ot remember if you can format the filesystem from the bootrom |
| 16:10.15 | userdefined | Zeeek: of course, if i ever get */SER/LCS working together here in the office i could also use my home * to forward calls to our corp. LCS host via our corp SER/* box (in theory) |
| 16:10.25 | Zeeek | upon further examination is is 4 6 8 * simultaneously |
| 16:10.37 | userdefined | like i said, still new, and likely quite naive in my expectations/theories =) |
| 16:10.54 | userdefined | s/naive/however it's really spelled |
| 16:11.11 | Zeeek | userdefined here's my main use: I can be working at home and watch the calls, make calls and receive calls as if I were in the office |
| 16:11.12 | mountainm2k | I see no way to format the filesystem from the bootrom |
| 16:11.32 | mountainm2k | also 468* only resets the "basic network config" |
| 16:11.43 | Zeeek | I can also have asterisk call me back free at a phone booth |
| 16:11.58 | Zeeek | mountainm2k the docs says Factory Reset |
| 16:13.10 | *** join/#asterisk Lino` (n=Lino@i577BFA3E.versanet.de) |
| 16:13.12 | mountainm2k | Yeah, "Reset to factory defaults" and the paragraph says "the basic network configuration referred to in the preceeding sections " |
| 16:14.04 | ptinsley | ya the 468* does work from bootrom on a 301 just checked |
| 16:14.04 | userdefined | Zeeek: my wife is pretty psyched about being able to set callerid to "answer me" selectively fwiw ;-) |
| 16:14.39 | Zeeek | callerid manipulation is a killer asset to asterisk |
| 16:14.49 | ptinsley | but there are some things you can mess up that that won't fix and I can't remember the way to fix it without the application loading |
| 16:15.29 | Spy000007 | haha this voicepulse "support services" is great -- i charge the customer $250 for a simple * setup and just pay voicepulse $99 to do it while i sit here and eat lunch |
| 16:15.31 | userdefined | i'll likely find it useful when responding to customers from home and not wanting to provide my real phone number/name |
| 16:16.02 | ptinsley | mountainm2k, if that doesn't work, do you have the default config files to put on your ftp server? |
| 16:16.27 | mafkees | userdefined: I use my home asterisk like that too |
| 16:17.03 | mountainm2k | No, I don't... |
| 16:17.12 | mafkees | it's my home system, but I can login any phone as "agent" and that will make that phone act as extension on work asterisk box |
| 16:17.20 | mountainm2k | I have the file it uploaded |
| 16:17.27 | mountainm2k | <mac>-phone.cfg |
| 16:17.31 | ptinsley | well, let me know if that doesn't work and i can get you a copy of the default files |
| 16:17.47 | mountainm2k | ...and since I can't download anything from Polycom's site |
| 16:17.50 | mountainm2k | well, the old stuff I can |
| 16:17.55 | mountainm2k | <grumps> |
| 16:17.55 | mafkees | too bad my iax provider wont let me spoof my callerid to something not billed on my iax user |
| 16:18.49 | Zeeek | change providers |
| 16:18.50 | ptinsley | hehe, I have been where you are, but I am a partner now so if you need files i can give them |
| 16:18.53 | Spy000007 | mafkees: what provider is that? |
| 16:18.53 | mountainm2k | ptinsley: if you could send me the defaults, that'd be cool... I can see it uploading the boot.log file, so I know it can get to my FTP server |
| 16:19.12 | mountainm2k | Or just the full SIP and bootrom ZIP files |
| 16:19.37 | mafkees | Spy000007: speakup |
| 16:20.25 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 16:21.02 | mafkees | I now setup a trunk to my work asterisk box, and as soon as my phone at home is logged in as agent on my home box all outgoing calls will be routed via the iax trunk to work box |
| 16:21.08 | ptinsley | mafkees, i'm sure thats a CYA on their part, i use teliax at the house and you can set it to whatever you want if you are looking for another provider, they aren't bad |
| 16:21.49 | mafkees | it's something on their part yeah |
| 16:21.52 | Spy000007 | never heard of them... i know voicepulse connect let's you set whatever you want |
| 16:21.58 | mafkees | it used to be possible |
| 16:22.01 | mafkees | but not anymore |
| 16:22.12 | mafkees | Spy000007: speakup is a dutch iax provider |
| 16:22.37 | Spy000007 | ah, ok, makes sense now |
| 16:24.30 | mafkees | I'm stuck with a specific queue config |
| 16:24.46 | *** part/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 16:24.48 | mafkees | I dont want to announce holdtime or position to the ppl waiting in the queue |
| 16:24.58 | mafkees | but I do want to play a soundfile every 25 seconds |
| 16:25.28 | mafkees | the file will tell the ppl in the queue: "please stay on the phone or press 9 to talk to operator" |
| 16:25.52 | mafkees | so I tried with announce-frequency = 0 and periodic-announce-frequency = 25 |
| 16:26.18 | mafkees | but on -users they told me I need the announce-frequency to something > 0 |
| 16:26.30 | mafkees | but that will make the holdtime/pos announcement to appear again |
| 16:26.35 | mafkees | but that's not what I want |
| 16:27.17 | _problem_ | mafkees: what do u mean by "but on -users they told me I need the announce-frequency to something > 0" |
| 16:27.25 | mafkees | mailing list |
| 16:27.30 | mafkees | I asked on the list first |
| 16:27.35 | mafkees | before going to irc |
| 16:27.42 | mafkees | at work I'm not allowed to irc |
| 16:27.50 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:28.08 | ptinsley | mafkees, try setting announce-holdtime = no |
| 16:28.30 | mafkees | ok, and how about the position ? |
| 16:31.23 | ptinsley | so you can't get it to play the periodic without the position is what you are saying? |
| 16:31.38 | mafkees | indeed |
| 16:32.19 | ptinsley | hmm, i have position turned on in all of my configs i think, i am not sure I have tried that setup |
| 16:33.00 | *** part/#asterisk bernardovieira (n=bernardo@c911935d.static.bhz.virtua.com.br) |
| 16:33.45 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 16:39.14 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 16:39.21 | *** join/#asterisk RoyK (n=roy@ti211310a080-6081.bb.online.no) |
| 16:39.47 | variable_office | what are some good incoming voip services that play nice with asterisk, voicepulses' $12/month for the number seems high |
| 16:40.00 | Zeeek | it is high |
| 16:40.18 | variable_office | Zeeek what do you use/like? |
| 16:40.21 | Spy000007 | it includes 4 channels and no per minute, but if you don't need it that much, you can probably get something cheaper |
| 16:40.40 | Zeeek | true it covers the channel issue that others do not |
| 16:41.07 | *** join/#asterisk Waverly360 (n=mirc@209.12.249.243) |
| 16:41.23 | Waverly360 | Good morning! |
| 16:41.42 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:41.43 | variable_office | Zeeek does that mean up to four people could be talking off you number at once on the incoming? |
| 16:42.35 | Waverly360 | CunningPike: Are you around today? :) |
| 16:42.40 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 16:42.53 | CunningPike | Waverly360: Yes :) |
| 16:43.01 | rainkid | so... what are some tricks i can use to minimize delay at the asterisk and ATA level? |
| 16:43.09 | Waverly360 | CunningPike: Mind if I pick your brain about queues some more? :) |
| 16:43.19 | CunningPike | Waverly360: Sure :D |
| 16:43.35 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 16:44.18 | variable_office | Zeeek what do you use for incoming? |
| 16:44.22 | Waverly360 | CunningPike: Is it possible to set up agents to be automatically logged into a queue? I don't want them to ever have to actually log in. |
| 16:45.13 | CunningPike | Waverly360: Certainly - just add their SIP UAs as queue members. In queues.conf, you can have member => SIP/whatever |
| 16:46.08 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:46.16 | CunningPike | Waverly360: But you will want your agents to logon and off - otherwise calls will get presented to agents who aren't there, especially with roundrobin |
| 16:46.46 | Waverly360 | CunningPike: Well, this is a special case. |
| 16:47.02 | Waverly360 | CunningPike: This queue will always be ringall. |
| 16:47.17 | CunningPike | Waverly360: Then member => will do it for you |
| 16:47.36 | Waverly360 | CunningPike: Awesome. I'm gonna play around with that for a bit. |
| 16:48.52 | s0lid | test |
| 16:48.53 | *** join/#asterisk Skarmeth (n=Skarmeth@201009012196.user.veloxzone.com.br) |
| 16:49.14 | CunningPike | Waverly360: What we do is set up our queue so that if an agent calls the queue from their appearance of the queue on their phones (the linekey/registration that queue calls are sent to), it calls AddQueueMember for that registration |
| 16:49.55 | kay2 | what's the way to convert a .ulaw or .alaw into .wav ? |
| 16:50.02 | kay2 | is there something doing that ? |
| 16:50.03 | tRSS | if I want to send a URL to an agent, I just simply put it in the Dial application. e.g. exten => 81XX,1Dial(SIP/user1,tT,http://www.google.com) , correct? |
| 16:50.12 | CunningPike | Waverly360: We don't actually use agents at all - you really only need agents if you want people to use different phones |
| 16:50.17 | Waverly360 | kay2: Look into a program called sox in linux. |
| 16:50.31 | CunningPike | kay2: You can use sox - there's a good page on the wiki about it |
| 16:50.43 | ptinsley | just don't install the rpm if you want mp3 support hehe |
| 16:50.53 | *** join/#asterisk gromm{CA} (n=me@xx081151026.cipherkey.com) |
| 16:50.54 | Waverly360 | Hah...yeah... |
| 16:51.40 | kay2 | thx |
| 16:51.50 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 16:52.21 | smackus | what is every ones preference for softphone on linux? |
| 16:52.28 | smackus | I have tried xlite. |
| 16:52.30 | smackus | not bad |
| 16:52.42 | ptinsley | kay2, sox with mp3 support is a good way to get around mp3 music on hold issues, just convert everything to gsm |
| 16:52.49 | *** join/#asterisk prodigy7 (n=prodigy7@p54A98F0A.dip0.t-ipconnect.de) |
| 16:53.02 | prodigy7 | hi |
| 16:53.25 | mountainm2k | hi |
| 16:53.27 | Bullseye_Network | smackus: We use sjphone |
| 16:53.46 | Bullseye_Network | sjlabs.com |
| 16:53.53 | wasim | moziax |
| 16:54.08 | CunningPike | smackus: SJPhone is my favorite |
| 16:54.18 | tRSS | wasim: i am in Lahore ;) |
| 16:54.20 | smackus | why is that? |
| 16:54.25 | mafkees | anyone here has a musiconhold file that plays ringing sound ? |
| 16:54.27 | mafkees | ;) |
| 16:54.40 | wasim | tRSS: woo hoo ;) |
| 16:54.56 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 16:55.00 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 16:55.01 | Bullseye_Network | smackus: We have 90+ Linux machines here running sjphone at our call centers. The quality is good and easy to seup and use |
| 16:55.02 | prodigy7 | i try to get asterisk work as voicebox for an sip number but the asterisk server doesn't answere calls ... i've one message which arrives on calls where i think, that this message shouldn't be ok -> chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) |
| 16:55.06 | ptinsley | mafkees, you have a pbx that won't ring? |
| 16:55.19 | tRSS | wasim: infact, i am at the empress road, right opposite the US Embassy/Consulate/whatever it is |
| 16:55.25 | nortex | [TK]D-Fender, Do you have a minute to answer a PRI question |
| 16:55.28 | prodigy7 | someone an idea what could be wrong? i think, i've done all neccessary portforwardings on my router... |
| 16:55.50 | TheCops | nortex I think he is not online right now |
| 16:55.56 | prodigy7 | and nat is enabled and type is peer |
| 16:55.58 | mafkees | ptinsley: I have, but with the help of _problem_ I found out announcement in queues wont work if you call it like this: Queue(name|tr) |
| 16:56.05 | CunningPike | nortex: Speak your question anyway |
| 16:56.15 | mafkees | ptinsley: I want ringing sound on the waiting ppl |
| 16:56.20 | mafkees | no fancy music |
| 16:56.26 | SplasPood | is 1.4 beta currently available, or where can I check out the latest tree headed towards 1.4 |
| 16:56.35 | ptinsley | i have a couple that won't ring anywhere except calling through queues but music on hold works fine |
| 16:56.37 | mafkees | so I provided the r flag to the Queue command |
| 16:56.41 | gromm{CA} | I have a digium TE110P T1 card that appears to be functional, but it doesn't detect the T1 signal. Is there any diagnostic stuff I don't know that could help me get this working? |
| 16:56.46 | CunningPike | SplasPood: SVN trunk? |
| 16:56.52 | kay2 | CunningPike: Is there any soft for Playing .ulaw ? |
| 16:56.54 | ptinsley | just dead air |
| 16:57.05 | mountainm2k | Thanks ptinsley, the phone actually works now again, lol |
| 16:57.08 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
| 16:57.12 | nortex | I'm having problems with long distance calling over PRI, I'm being told I need to switch the numbering from E.164 to unknown. How do I do that? |
| 16:57.14 | ptinsley | awesome, glad i could help |
| 16:57.24 | SplasPood | CunningPike: So the latest TRUNK is what's going into 1.4? |
| 16:57.26 | ptinsley | the polycom's are tricky but after you get them down, they are great |
| 16:57.49 | mafkees | musiconhold works great here too |
| 16:57.53 | mafkees | but I dont want music |
| 16:57.54 | CunningPike | SplasPood: #asterisk-dev would provide better details, but in general terms, yes |
| 16:57.58 | mafkees | I want ringing sound ;) |
| 16:58.01 | ptinsley | hehe |
| 16:58.09 | mafkees | but without musiconhold the queue announcements wont wokr |
| 16:58.13 | mafkees | work |
| 16:58.16 | RoyK | SplasPood: yes |
| 16:58.19 | prodigy7 | have maybe someone an working asterisk configuration for 1&1 behind an firewall ? |
| 16:58.23 | CunningPike | gromm{CA}: Do you have any PRI debug output? |
| 16:58.29 | Waverly360 | heh..dirty trick...just use a recording of ringing as your moh file... ;) |
| 16:58.38 | SplasPood | RoyK: Where does all the development for stuff that'll be post 1.4 release end up then? |
| 16:58.40 | RoyK | SplasPood: trunk is to be 1.4 beta and then 1.4 |
| 16:58.42 | gromm{CA} | CunningPike: besides 'pri show span 1' in asterisk? |
| 16:58.47 | mafkees | Waverly360: hehehehe |
| 16:58.51 | CunningPike | kay2: You mean outside of asterisk - just generally? |
| 16:58.58 | *** join/#asterisk _GiGi_ (i=gigi@disc.more.pl) |
| 16:59.03 | RoyK | SplasPood: on mantis. what sort of stuff are you talking about? |
| 16:59.05 | CunningPike | gromm{CA}: No - just that - does it show anything? |
| 16:59.11 | mafkees | Waverly360: I was hoping someone had that soundfile for me |
| 16:59.13 | gromm{CA} | CunningPike: Yup. |
| 16:59.19 | kay2 | CunningPike: yeahg |
| 16:59.21 | CunningPike | pastebin it |
| 16:59.22 | kay2 | CunningPike: outside |
| 16:59.26 | wunderkin | nortex, /etc/asterisk/zapata.conf pridialplan=unknown, prilocaldialplan=unknown |
| 16:59.27 | _GiGi_ | hello. |
| 16:59.31 | SplasPood | RoyK: I dunno, whatever anyone was working on.. I thought there'd be a branch for 1.4 and trunk would be "unstable" |
| 16:59.31 | CunningPike | gromm{CA}: Pastebin it |
| 16:59.32 | CunningPike | ~pb |
| 16:59.34 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 16:59.52 | CunningPike | kay2: What OS? |
| 16:59.56 | RoyK | SplasPood: email the -dev list about it. it's a good question |
| 17:00.00 | kay2 | CunningPike: windows |
| 17:00.11 | gromm{CA} | CunningPike: working on it. :) |
| 17:00.14 | _GiGi_ | im trying to run hylafax with sangoma A104 but i cant find any solutions how it make, someone can help me ? :) |
| 17:00.15 | mountainm2k | ptinsley: after all that, it still has the SIP info -- where the heck did it get that?!?!? |
| 17:00.20 | RoyK | kay2: running asterisk on windoze? |
| 17:00.22 | CunningPike | kay2: Oh - no idea - don't use it, sorry |
| 17:00.23 | SplasPood | RoyK: what, to ask why there isn'a a branch for 1.4 beta yet? |
| 17:00.25 | RoyK | _GiGi_: why hylafax? |
| 17:00.27 | kay2 | RoyK: NO |
| 17:00.31 | RoyK | :) |
| 17:00.32 | _GiGi_ | RoyK: why not ? :> |
| 17:00.37 | kay2 | CunningPike: linux then |
| 17:00.37 | kay2 | :) |
| 17:00.54 | Waverly360 | mafkees: Well..I'm sure someone does, but that's a really dirty hack. There's got to be some way to make it work like it should. We're having a similar problem here. |
| 17:00.56 | gromm{CA} | dammit. Pastebin is fuxxored. |
| 17:01.01 | _GiGi_ | RoyK: give me best solution for faxserver :) |
| 17:01.10 | Waverly360 | mafkees: It's just not important enough to focus on now..have other issues that take priority. |
| 17:01.23 | RoyK | SplasPood: yeah, or ask _why_ there isn't an 1.4 branch yet, since I agree with you that trunk should be trunk as soon as 1.4 is feature frozen |
| 17:01.34 | Bullseye_Network | try pastebin.ca |
| 17:01.34 | RoyK | _GiGi_: spandsp+app_rxfax+app_txfax |
| 17:01.42 | RoyK | ~pb |
| 17:01.44 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 17:02.00 | mafkees | Waverly360: I know, but this customer is getting on my nerves ;) |
| 17:02.19 | mafkees | hhmm |
| 17:02.22 | kay2 | CunningPike: under linux, you know anytything ? |
| 17:02.24 | _GiGi_ | RoyK: i have it now, 30% faxes get badrows |
| 17:02.29 | gromm{CA} | CunningPike: http://pastebin.ca/69192 |
| 17:02.31 | Waverly360 | mafkees: Hah..I know that feeling ;). Wish I could help more. If we figure ours out, I'll let ya know :) |
| 17:02.37 | RoyK | _GiGi_: we use spandsp with sangoma and te410p cards and what not, and it works like a dream |
| 17:02.38 | kay2 | CunningPike; that could read this .ulaw properly |
| 17:02.38 | kay2 | ? |
| 17:02.39 | CunningPike | kay2: Not really - I'm a Mac user :D |
| 17:02.41 | s0lid | i have an x100p |
| 17:02.45 | mafkees | Waverly360: please |
| 17:02.47 | RoyK | _GiGi_: what spandsp version? |
| 17:02.48 | gromm{CA} | CunningPike: keep in mind, that it's not hooked up to the T1. |
| 17:02.52 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 17:02.58 | kay2 | CunningPike: OK , under mac then |
| 17:03.00 | s0lid | i have an x100p how do i config it for incoming calls? |
| 17:03.07 | gromm{CA} | Unfortunately they're rather expensive and we've only got 1. :) |
| 17:03.11 | RoyK | s0lid: rtfm :) |
| 17:03.13 | RoyK | ~docs |
| 17:03.15 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:03.22 | mafkees | Waverly360: Queue(name|t) works |
| 17:03.23 | _GiGi_ | RoyK: (i got ~20faxes per second - active 20 channels) |
| 17:03.33 | RoyK | 20 faxes perr |
| 17:03.35 | RoyK | second? |
| 17:03.37 | mafkees | Queue(name|tr) works too, but wont play the announcements |
| 17:03.38 | kay2 | Waverly360: what u trying to do |
| 17:03.38 | RoyK | how is system load? |
| 17:03.42 | RoyK | cpu load |
| 17:03.43 | RoyK | etc |
| 17:03.59 | kay2 | Waverly360: I might be able to help |
| 17:04.03 | _GiGi_ | RoyK: 20 active channels everytime |
| 17:04.07 | Hmmhesays | what does res_crypto do in asterisk? |
| 17:04.10 | RoyK | _GiGi_: might be a load problem |
| 17:04.14 | s0lid | RoyK: can you give me one since you are so good |
| 17:04.15 | *** join/#asterisk mtaht4 (n=m@64-60-251-182.cust.telepacific.net) |
| 17:04.17 | RoyK | _GiGi_: what is the system load? |
| 17:04.28 | _GiGi_ | not 20 connection per sec :D |
| 17:04.31 | _GiGi_ | hmm |
| 17:04.31 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 17:04.34 | _GiGi_ | load ~1 |
| 17:04.38 | Waverly360 | kay2: I'm working on some queue stuff now..but CunningPike answered my question earlier. Also having some issues with the phone's ringing in the handset when you try to call someone within the office. |
| 17:04.50 | _GiGi_ | RoyK: but on iaxmodem and hylafax it works fine... |
| 17:04.52 | RoyK | s0lid: please, the docs are really good, and asking for 'how do I write an operating system' in #kernelnewbies is not a welcoming question |
| 17:04.53 | Waverly360 | kay2: mafkees is having a similar problem. |
| 17:05.00 | ptinsley | ya I would buy somebody a pizza if they can figure out why asterisk doesn't ring on some installs |
| 17:05.01 | _GiGi_ | but iaxmodem crashes in high load... |
| 17:05.16 | RoyK | i don't use iaxmodem.... |
| 17:05.17 | s0lid | RoyK: ok god of asterisk |
| 17:05.40 | mafkees | ptinsley: all voip channels ? |
| 17:05.44 | kay2 | Waverly360: well it's normal |
| 17:05.49 | mafkees | or pri/bri/landlines too ? |
| 17:05.50 | kay2 | Waverly360: I patched mine for that |
| 17:05.57 | RoyK | s0lid: not meaning to be rude, but please read some docs before asking. there's a really good book from o'reilly as well if you find the docs hard reading |
| 17:06.08 | _GiGi_ | RoyK: uhm, but when i connect it to hylafax i got ~2-3% bad faxes. |
| 17:06.08 | kay2 | Waverly360: because when it checks the status, it alows you to send the INVITE if you are In Use |
| 17:06.20 | _GiGi_ | RoyK: but sometimes iaxmodem crash :) |
| 17:06.23 | kay2 | mafkees ? |
| 17:06.27 | RoyK | _GiGi_: why iaxmodem? |
| 17:06.31 | CunningPike | gromm{CA}: Well, your PRI is down (I'm sure you knew that already :D) - you can't really do much until it's up. You'll need to work with your telco to that the signalling etc correct |
| 17:06.32 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 17:06.43 | Waverly360 | kay2: Wait..I'm not sure I understand. |
| 17:06.46 | CunningPike | gromm{CA}: What is the PRI connected to |
| 17:06.54 | kay2 | Waverly360: have a look in trunk |
| 17:06.56 | RoyK | _GiGi_: i don't know, really, we have quite a bit of a load on this system, 99% voice calls, though, but a fax here and there |
| 17:06.57 | mafkees | yes kay? |
| 17:07.05 | RoyK | _GiGi_: just app_rxfax, though |
| 17:07.10 | kay2 | mafkees: what's your problem with queue |
| 17:07.12 | CunningPike | kay2: I like Sound Studio |
| 17:07.13 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 17:07.18 | Dr-Linux | hi all |
| 17:07.28 | gromm{CA} | CunningPike: Actually, we have been. We've also got another asterisk box connected to that T1, and we're using the same settings on the new box as the old box. |
| 17:07.39 | mafkees | kay2: announcements wont play if I use the r option in the Queue() call |
| 17:07.42 | AltnTab | How can i make Read() timeout last longer ?! |
| 17:07.57 | RoyK | AltnTab: show function TIMEOUT |
| 17:07.59 | mafkees | kay2: I dont want musiconhold, I want it to provide ringing sound |
| 17:08.02 | kay2 | mafkees: all the rest will |
| 17:08.03 | _GiGi_ | RoyK: i have dedicated box for it. |
| 17:08.05 | CunningPike | gromm{CA}: The same PRI? :S |
| 17:08.06 | gromm{CA} | CunningPike: We're starting to think that there's something wrong with the TE110P, since we did get it at a bargain price. :) |
| 17:08.12 | Dr-Linux | one of my client in pakistan needs 12 numbers, so what's the prices for a US DID from a sip provider? |
| 17:08.18 | mafkees | kay2: I know, but still it's weird |
| 17:08.19 | RoyK | _GiGi_: but just using app_rxfax? |
| 17:08.24 | gromm{CA} | CunningPike: yup. We have to switch between the two. This is supposed to be an upgrade. |
| 17:08.28 | mafkees | none of the docs say the r option will kill announcements |
| 17:08.33 | _GiGi_ | RoyK: rx and tx. |
| 17:08.36 | kay2 | mafkees: have a look at what's going on on the rtp part |
| 17:08.45 | Hmmhesays | anyone know what res_crypto does? |
| 17:08.54 | CunningPike | gromm{CA}: Ah, so you unplug the PRI from one server and plug it into the other? |
| 17:09.03 | mafkees | Hmmhesays: It's for the keys you can use for md5 auth |
| 17:09.05 | kay2 | Hmmhesays: look into it |
| 17:09.06 | gromm{CA} | CunningPike: yes. |
| 17:09.14 | Hmmhesays | so I don't necessarily need it |
| 17:09.37 | kay2 | mafkees: what is it for at the end ? |
| 17:09.40 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 17:09.41 | kay2 | mafkees: a call center ? |
| 17:09.46 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 17:09.49 | mafkees | kay2: no, queue per phone |
| 17:09.52 | Waverly360 | kay2: Are we talking about the same problem? I'm curious about why I don't hear ringing in my handset when I'm calling someone else in the office. If I call an external number it rings fine. |
| 17:10.03 | gromm{CA} | CunningPike: It's also worth noting that when we load the kernel module for the card, sometimes it gives me some odd irq errors "wrote xx but read yy". |
| 17:10.13 | CunningPike | gromm{CA}: And you're absolutely positively sure that the zaptel and zapata.conf are the same - what hardware is in the older server? |
| 17:10.19 | kay2 | Waverly360: You are talking about app_queue ? |
| 17:10.45 | mafkees | kay2: customer has operator. This operator transfers calls to extensions, and wants to be able to put several callers in line for a phone |
| 17:10.49 | Waverly360 | kay2: I'm talking about a simple call from one phone to another. not involving queues. |
| 17:10.49 | CunningPike | gromm{CA}: Hmmm - sounds like an IRQ conflict. What does cat /proc/interrupts say? |
| 17:10.56 | mafkees | kay2: so I created a queue for every extension |
| 17:11.31 | gromm{CA} | CunningPike: we have a quad pri card in it... a TE405P |
| 17:11.40 | kay2 | Waverly360: are you sure you didnt do a answer before or something like this ? |
| 17:11.47 | kay2 | cuz what u saying is weird |
| 17:11.52 | mafkees | kay2: and while ppl are waiting for the phone to become available, they should hear normal ringing sound |
| 17:12.04 | CunningPike | gromm{CA}: Can you pastebin the zapata and zaptel files from _both_ servers> |
| 17:12.05 | CunningPike | ? |
| 17:12.05 | mafkees | kay2: and some commercial every 25 seconds |
| 17:12.27 | Dr-Linux | CunningPike: hi |
| 17:12.36 | CunningPike | gromm{CA}: And the output from cat /proc/interrupts on the newer server |
| 17:12.39 | kay2 | mafkees: what happens when you call from one phone to an other one |
| 17:12.41 | gromm{CA} | CunningPike: okay, will do. I'll include the /proc/interrupts output too |
| 17:12.48 | CunningPike | gromm{CA}: Perfect |
| 17:12.58 | CunningPike | Dr-Linux: Good morning to you |
| 17:13.15 | kay2 | hold on, rebooting |
| 17:13.17 | Dr-Linux | CunningPike: good moring to you, but good night to me |
| 17:13.29 | mafkees | kay2: ringing sound |
| 17:13.32 | CunningPike | Dr-Linux: Ah yes |
| 17:13.47 | mafkees | kay2: it's only an issue with the queue command |
| 17:13.57 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 17:13.59 | tdi | hi |
| 17:14.02 | tdi | s it possible to use hylafax with sangoma a104? |
| 17:14.57 | Dr-Linux | CunningPike: one of my client is starting new call center in pakistan, he will have 12 users inbound/outbound, so he will need 12 US and UK SIP DID's, |
| 17:15.10 | *** join/#asterisk flujan (n=flujan@201-43-210-40.dsl.telesp.net.br) |
| 17:15.13 | Dr-Linux | CunningPike: he wants me to configure his setup. |
| 17:15.26 | Dr-Linux | CunningPike: so what's the price for DID's? |
| 17:15.28 | flujan | dlynes_home, are you here? |
| 17:15.51 | CunningPike | Dr-Linux: I have no idea - we haven't really looked at SIP termination |
| 17:16.05 | flujan | guys, i trying to configure asterisk to make calls... |
| 17:16.11 | Dr-Linux | hhm.. |
| 17:16.16 | CunningPike | flujan: Aren't we all? :) |
| 17:16.21 | Dr-Linux | CunningPike: not sure how can guide me with this |
| 17:16.32 | tdi | i want to use 4xE1 sangoma in such way, that i do not have to use iaxmodem, chan_fax or rxfax |
| 17:16.36 | Hmmhesays | heh this acutually build for mipsel |
| 17:16.46 | flujan | CunningPike, just to make sure... last time i enter here dlynes_home just leave to install a new server. :P |
| 17:16.49 | CunningPike | Dr-Linux: Didn't someone suggest mixnetworks earlier - I saw them at Astricon and was impressed |
| 17:17.08 | flujan | well, I will describe the problem again. :P |
| 17:17.10 | *** join/#asterisk pdt-mobile (n=ptinsley@209.12.249.243) |
| 17:17.19 | Dr-Linux | CunningPike: what's there site? |
| 17:17.19 | flujan | first, i configure asterisk to work with a ISDN/PRI |
| 17:17.55 | CunningPike | Dr-Linux: http://www.mixnetworks.com/ - GIYF |
| 17:18.11 | gromm{CA} | CunningPike: Hmm... in /proc/interrupts, it seems to be sharing the IRQ... there's more than one device on that irq. |
| 17:18.12 | CunningPike | ~google |
| 17:18.13 | jbot | it has been said that google is a search engine found at http://www.google.com/ |
| 17:18.37 | CunningPike | gromm{CA}: Ah - that can cause issues - what's it sharing with? |
| 17:19.07 | CunningPike | gromm{CA}: Although usually they are timing issues, rather than not being able to get a PRI p |
| 17:19.16 | CunningPike | s/ p/up/ |
| 17:19.48 | gromm{CA} | CunningPike: see for yourself: http://pastebin.ca/69203 |
| 17:20.03 | nortex | When using E.164 and sending the caller id number to the telco how should my number beformated? |
| 17:20.21 | wunderkin | nortex, just set it to unknown.. |
| 17:20.26 | gromm{CA} | good god, jbot does regexes too eh? |
| 17:20.38 | mafkees | nortex: countycode + number |
| 17:20.43 | flujan | http://pastebin.ca/69204 here goes what I have to make a call. |
| 17:20.47 | CunningPike | gromm{CA}: Do you need USB on your server? |
| 17:20.51 | pdt-mobile | so has anyone here had a non ringing pbx problem that they have fixed. Sip to Sip don't ring and Sip to PSTN doesn't ring till the pbx downstream creates one |
| 17:20.58 | gromm{CA} | CunningPike: probably not. |
| 17:20.58 | pdt-mobile | but for some reason queues can make ring |
| 17:21.00 | mafkees | nortex: full international format without the digits you have to use to get international line |
| 17:21.03 | flujan | later, I cannot make other calls... I got this error: |
| 17:21.20 | CunningPike | gromm{CA}: Disable it then - it's bad to have it sharing an IRQ with your card |
| 17:21.20 | mafkees | pdt-mobile: yeah |
| 17:21.29 | mafkees | pdt-mobile: add the r flag to the dial command |
| 17:21.33 | pdt-mobile | it's there |
| 17:21.35 | gromm{CA} | CunningPike: okay, I'll give that a try. |
| 17:21.35 | pdt-mobile | makes no difference |
| 17:22.08 | CunningPike | flujan: Your PRI isn't working properly |
| 17:22.08 | CunningPike | flujan: No D-Channel |
| 17:22.12 | pdt-mobile | if i replace it with m, i get music on hold |
| 17:22.13 | Dr-Linux | CunningPike: i can't see there DID's price |
| 17:22.16 | pdt-mobile | so that part works |
| 17:22.21 | nortex | mafkees, So for US longdistance 10 digits should work? |
| 17:22.30 | CunningPike | flujan: What does 'pri show span 1' say? |
| 17:22.37 | flujan | CunningPike, but this is a problem of my configuration? |
| 17:22.43 | mafkees | nortex: I dont know how usa numbers are formatted |
| 17:22.44 | mafkees | sorry |
| 17:22.50 | CunningPike | flujan: Don't know yet |
| 17:22.50 | mafkees | <--- from the netherlands |
| 17:22.56 | mafkees | pdt-mobile: weird |
| 17:22.58 | userdefined | heh. a week of playing and *now* i discover the 'asterisk handbook' =) |
| 17:23.07 | nortex | mafkees, No prob |
| 17:23.40 | flujan | CunningPike, http://pastebin.ca/69206 |
| 17:23.50 | wunderkin | nortex, yes, 1NXXNXXX |
| 17:24.17 | gromm{CA} | userdefined: Heh. That's okay, we were running a live asterisk server for about a year before the O'Reilly book came out. :) |
| 17:24.35 | CunningPike | flujan: Your PRI is down - have you contacted your telco? |
| 17:24.38 | gromm{CA} | Mmm. Life on the bleeding edge. |
| 17:24.50 | pdt-mobile | mafkees: Dial(SIP/306,20,rt) but no ring when that goes through... i have seen some references to it on the mailing lists and other places but no resolutions |
| 17:24.55 | CunningPike | Dr-Linux: Well, I can - look under Services....... |
| 17:25.09 | gromm{CA} | CunningPike: so disable USB. Any other recommendations? |
| 17:25.54 | mafkees | pdt-mobile: did you fiddle around with indications.conf ? |
| 17:27.12 | pdt-mobile | we have a ban on files that start with i |
| 17:27.16 | CunningPike | gromm{CA}: Your card is on an IRQ by itself now? |
| 17:27.17 | flujan | CunningPike, but I have a gren light in zap show status |
| 17:27.32 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:27.33 | gromm{CA} | CunningPike: I'll see what I can do. It's down at the datacenter right now. :) |
| 17:27.42 | flujan | CunningPike, no alarms... |
| 17:27.43 | mafkees | pdt-mobile: smart move |
| 17:27.46 | pdt-mobile | hehehe |
| 17:27.47 | CunningPike | gromm{CA}: OK |
| 17:28.06 | CunningPike | flujan: You call progression says otherwise - what does 'pri show span 1' say? |
| 17:28.07 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 17:28.11 | mafkees | how can I record the ringing sound with asterisk ? |
| 17:28.16 | AltnTab | I've Set(TIMEOUT(response)=30), but still 5 seconds at max of timeout ... !? |
| 17:28.34 | NormSteel | installing a new copy of asterisk should i go w/ the asterisk-1.2? |
| 17:28.40 | gromm{CA} | CunningPike: one of the other odd things I've noticed is that zttool says "Alarms: Ok", even when it's not plugged into the PRI. :/ |
| 17:29.02 | flujan | CunningPike, http://pastebin.ca/69206 |
| 17:29.18 | CunningPike | gromm{CA}: That is odd - no errors in dmesg or ztcfg? |
| 17:29.46 | CunningPike | flujan: "Status: Provisioned, Down, Active" - don't know how else to say it |
| 17:29.51 | wasim | tRSS: isn't empress market in karachi? |
| 17:30.20 | wasim | or did they build one in her favour here too? |
| 17:30.21 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226) |
| 17:30.30 | *** join/#asterisk avpatel (n=patel455@c-69-142-70-121.hsd1.pa.comcast.net) |
| 17:30.32 | CunningPike | flujan: Have you spoken to your telco? |
| 17:30.32 | justinu | wasim lives |
| 17:31.13 | gromm{CA} | CunningPike: "wcte11xp: Unknown parameter `override'", that's it. Otherwise, it says "Found a Wildcard: Digium Wildcard TE110P T1/E1" and "TE110P: Span configured for ESF/B8ZS", and that's generally a good thing. |
| 17:31.24 | avpatel | any pointer to add support for T.38 passthrough in ooh323c |
| 17:31.50 | flujan | CunningPike, could it be a problem in my zaptel.conf or zapata.conf? |
| 17:31.52 | mafkees | avpatel: check the history in the sip channel |
| 17:32.15 | CunningPike | gromm{CA}: I'd suspect the card at this stage......... |
| 17:32.15 | pdt-mobile | mafkees: well... it would seem our ban on files that start with i screwed us |
| 17:32.16 | mafkees | avpatel: I think there's even a branch in svn for the t.38 stuff in the sip stack |
| 17:32.16 | pdt-mobile | it works now |
| 17:32.28 | gromm{CA} | CunningPike: So are we. :) |
| 17:32.33 | pdt-mobile | mafkees: where do you live, I am going to make good on my free pizza offer |
| 17:32.37 | CunningPike | gromm{CA}: :) |
| 17:32.40 | *** join/#asterisk trbldwine (i=trbldwin@adam.ur.northwestern.edu) |
| 17:32.41 | gromm{CA} | CunningPike: we're going to try the irq thing first ,and then give up. :) |
| 17:32.45 | avpatel | mafkees: looks to much complicated, that's the only chance |
| 17:32.45 | mafkees | pdt-mobile: The Netherlands |
| 17:32.50 | CunningPike | gromm{CA}: Where in CA are you? |
| 17:33.00 | gromm{CA} | CunningPike: Vancouver |
| 17:33.03 | pdt-mobile | do you have pizza places that take mastercard or visa? |
| 17:33.08 | pdt-mobile | i'll totally send you a pizza |
| 17:33.13 | pdt-mobile | that has been bugging me for WEEKS |
| 17:33.31 | CunningPike | flujan: Yes - it could be. But we'll never know until you talk to your telco, confirm that the PRI is up and confirm your signalling etc. |
| 17:33.54 | CunningPike | gromm{CA}: Get out! Me too |
| 17:33.59 | mafkees | pdt-mobile: ehm, not that I know |
| 17:34.00 | tdi | russellb: can i priv? |
| 17:34.01 | mafkees | :( |
| 17:34.04 | pdt-mobile | oh well |
| 17:34.08 | gromm{CA} | CunningPike: heh |
| 17:34.13 | mafkees | never mind dude |
| 17:34.15 | russellb | tdi: no |
| 17:34.17 | mafkees | glad I could help |
| 17:34.20 | wasim | pdt-mobile: a pizza hut here does |
| 17:34.33 | flujan | CunningPike, ok. I will talk with then... thanks... now I go home to watch Brazil vs. Japan... Actually, I'm from Brazil. :P |
| 17:34.39 | *** join/#asterisk dan42 (n=lung@24-148-96-186.ip.mhcable.com) |
| 17:34.49 | pdt-mobile | mafkees: now I just need to find you a ringing file, what style ring do you need? |
| 17:35.00 | CunningPike | flujan: Well, good luck with both :) Should be about 10-0 to Brazil |
| 17:35.02 | tdi | russellb: can i know the ban reason? |
| 17:35.04 | mafkees | pdt-mobile: just the normal ringing |
| 17:35.11 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:35.22 | mafkees | the one you get with dial(SIP/400,20,tr) |
| 17:35.55 | mafkees | I need it in asterisk native moh or mp3 |
| 17:36.07 | mafkees | as long as I can use it in MOH for a queue :) |
| 17:36.53 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 17:37.25 | eBody | what mod do i need, if at all, for sms text messaging?? |
| 17:37.35 | mafkees | none |
| 17:37.38 | mafkees | look at SMS |
| 17:38.00 | mafkees | food |
| 17:38.01 | mafkees | brb |
| 17:38.07 | flujan | CunningPike, maybe... ;) thanks for the help |
| 17:38.11 | flujan | CunningPike, see you. |
| 17:39.05 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 17:41.31 | [TK]D-Fender | mafkees : There is a queue option for ringing instead of MoH |
| 17:41.59 | [TK]D-Fender | mafkees : However callers would typically find it annoying |
| 17:42.05 | *** join/#asterisk burizaa (n=freeee@bb219-74-196-240.singnet.com.sg) |
| 17:42.06 | mafkees | [TK]D-Fender: yes, but that kills announcements |
| 17:43.22 | smackus | help!! my system crashed... I get this error when running asterisk -c |
| 17:43.23 | smackus | [root@asterisk ~]# Warning, flexibel rate not heavily tested! |
| 17:43.23 | smackus | Ouch ... error while writing audio data: : Broken pipe |
| 17:43.28 | [TK]D-Fender | mafkees : Does it? Oh well.... incessant ringing it is... |
| 17:43.49 | [TK]D-Fender | smackus : I'm betting a zaptel interface didn't come up |
| 17:44.05 | smackus | ok... where should I start looking? |
| 17:44.27 | mafkees | [TK]D-Fender: it's not my call. The customer wants ringing sound |
| 17:44.33 | tzafrir | _GiGi_, you had some questions about something with Asterisk? this may be the channel to ask them. (BTW: don't expect me to have the knowledge to answer...) |
| 17:44.38 | mafkees | so I did Queue(name|r) |
| 17:44.50 | mafkees | but the moment you do that, all announcements stop to work |
| 17:45.00 | _GiGi_ | tzafrir: im ask here, but i didnt get answer :/ |
| 17:45.03 | mafkees | so now I'm looking for a MOH file that plays the ringing sound ;) |
| 17:45.03 | smackus | here is my entire output: http://pastebin.ca/69221 |
| 17:45.27 | [TK]D-Fender | mmmm KINKY |
| 17:46.04 | smackus | ok, now I have this: |
| 17:46.05 | smackus | [root@asterisk ~]# Warning, flexibel rate not heavily tested! |
| 17:46.05 | smackus | Ouch ... error while writing audio data: : Broken pipe |
| 17:46.06 | smackus | Junk at the beginning 49443303 |
| 17:46.06 | smackus | Warning, flexibel rate not heavily tested! |
| 17:46.49 | justinu | smackus: asterisk -vvvvc to see where it died |
| 17:46.53 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:47.16 | file | oej: say it ain't so! |
| 17:47.37 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 17:47.54 | Zodiacal | anyone know if theres a way to switch lines on an analog phone? fxs? |
| 17:48.04 | smackus | http://pastebin.ca/69225 |
| 17:48.08 | Zodiacal | like to answer an incoming call |
| 17:48.13 | justinu | flash button |
| 17:48.20 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-5-22.slkc.qwest.net) |
| 17:48.33 | Zodiacal | justinu, coolness.. will that switch back to previous lines too? |
| 17:48.50 | justinu | yeah, i guess you never used call waiting on POTS line? |
| 17:48.52 | Strom_C | bah, flash button. real men flash the hookswitch |
| 17:49.07 | Zodiacal | justinu not in many years |
| 17:49.07 | tzafrir | _GiGi_, I see that you were answered. Maybe ask again, and be a bit more specific. |
| 17:49.08 | Zodiacal | :P |
| 17:49.08 | justinu | smackus: this is the problem: |
| 17:49.09 | justinu | Â [res_config_mysql.so]Jun 22 11:47:03 WARNING[4345]: loader.c:728 __load_resource: missing mod_data for res_config_mysql.so Segmentation fault |
| 17:49.18 | smackus | but it is latin to me.... |
| 17:49.23 | smackus | what do I need to fix? |
| 17:49.31 | justinu | don't use res_config_mysql.so |
| 17:49.41 | smackus | how do I not use it... |
| 17:49.54 | _GiGi_ | tzafrir: ok i ask after hour :) im ask on hylafax list and im waiting for answer :) |
| 17:49.54 | [TK]D-Fender | Strom_C : Real men use "immediate=yes" and have a hot chick patch the call with 1/4" phono plugs :D |
| 17:49.55 | smackus | some one else must have turned it on. |
| 17:50.03 | justinu | delete the file in /usr/lib/asterisk/modules? |
| 17:50.10 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 17:50.52 | *** part/#asterisk burizaa (n=freeee@bb219-74-196-240.singnet.com.sg) |
| 17:51.34 | userdefined | justinu: i believe it'd be better to add 'noload => res_config_mysql.so' to your modules.conf' |
| 17:51.48 | userdefined | but i'm a noob and could be wrong |
| 17:52.15 | justinu | either way |
| 17:52.40 | justinu | i figured rm would be easier for him |
| 17:52.41 | justinu | :P |
| 17:52.57 | twisted[asteria] | why does people talk to me when i'm not here? |
| 17:53.00 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:53.00 | twisted[asteria] | s/does/do |
| 17:53.05 | mafkees | lol twisted[asteria] |
| 17:53.13 | mafkees | they must be in love with you |
| 17:53.28 | twisted[asteria] | perhaps |
| 17:54.27 | userdefined | justinu: heh. i misread the lines and thought you were the one asking ;-) |
| 17:54.51 | smackus | ok, so now what does this mean? |
| 17:54.53 | smackus | <PROTECTED> |
| 17:54.53 | smackus | Jun 22 11:53:01 WARNING[5470]: manager.c:2457 init_manager: Unable to bind socket: Address already in use |
| 17:55.03 | twisted[asteria] | it tells you exactly what it means |
| 17:55.28 | justinu | yeah, something is already using that port |
| 17:56.21 | Spy000007 | hmm, i need to setup a call in number connected to my sirius satellite radio that includes changing channels... |
| 17:57.22 | justinu | sounds illegal to me :P |
| 17:57.25 | smackus | ok, so adding the line noload => did not fix it. |
| 17:57.39 | Spy000007 | for me, not for you :P |
| 17:57.59 | justinu | aiding and abetting? |
| 17:58.28 | Spy000007 | spys have immunity from such things |
| 17:58.32 | justinu | werd |
| 17:59.23 | Spy000007 | I think half the stuff being done in this channel is illegal. haha |
| 18:00.02 | justinu | i don't think that's true |
| 18:00.27 | *** join/#asterisk burizaa (n=freeee@bb219-74-196-240.singnet.com.sg) |
| 18:00.43 | burizaa | hi all, quick qn, what does "Unmonitored |
| 18:00.49 | burizaa | mean under SIP show peers ? |
| 18:01.00 | justinu | means you don't have a qualify configured for that peer |
| 18:01.37 | burizaa | and what does insecure settings for ? |
| 18:01.44 | mafkees | hi oej |
| 18:01.46 | justinu | that has to do with SIP authentication |
| 18:01.56 | burizaa | thnx justinu |
| 18:02.00 | justinu | np |
| 18:03.00 | tRSS | my res_odbc.conf is unable to connect. how can i check if I have the odbc drivers installed. i am using FC4 |
| 18:03.13 | burizaa | i put qualify=yes but then the status become unknown... any idea? |
| 18:03.30 | mafkees | it means the phone is not registered yet |
| 18:03.51 | burizaa | mafkees, i'm creating a peer trunk |
| 18:03.56 | mafkees | ah |
| 18:03.57 | mafkees | sorry |
| 18:04.21 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 18:04.34 | justinu | host=dynamic? |
| 18:05.01 | smackus | ok, suddenly i get the error Jun 22 12:02:57 ERROR[4335]: chan_zap.c:10702 setup_zap: Unknown signalling method 'pri_cpe' |
| 18:05.06 | [TK]D-Fender | justinu : Peer trunks would not be "dynamic". |
| 18:05.10 | justinu | i know that |
| 18:05.11 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 18:05.11 | smackus | i have always set it to pri_cpe |
| 18:05.35 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 18:05.42 | [TK]D-Fender | burizaa : Rather than dropping little breadcrumbs like you've been doing, pastebi the config as you're using it now. |
| 18:05.45 | [TK]D-Fender | ~pb |
| 18:05.49 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 18:05.49 | burizaa | justinu: i put the ip .. host=192.168.0.254 |
| 18:06.18 | justinu | burizaa: not sure why you'd get "unknown" in that situation |
| 18:07.05 | burizaa | http://pastebin.ca/69235 |
| 18:07.38 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 18:08.23 | [TK]D-Fender | burizaa : ALL OF IT |
| 18:08.27 | justinu | lol |
| 18:08.42 | [TK]D-Fender | justinu : That context name is a huge tip-off though... |
| 18:08.43 | burizaa | [TK]D-Fender: i'm using freepbx :( |
| 18:08.50 | [TK]D-Fender | justinu : SEE.... |
| 18:08.56 | justinu | lol |
| 18:09.13 | [TK]D-Fender | burizaa : And the next "commit" you do blows everything we suggest to you away |
| 18:09.38 | [TK]D-Fender | burizaa : Past it ALL anyways... |
| 18:09.55 | Strom_C | for estimating call traffic, is there a standard rough estimate of how many local and long distance minutes the average five-person office will use in a given month? |
| 18:10.06 | mog | Strom_C, you have the coolest junk |
| 18:10.12 | Strom_C | do I? |
| 18:10.19 | Strom_C | ooooh :) |
| 18:10.23 | mafkees | Strom_C: get the last several bills from the current telco |
| 18:10.29 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 18:10.35 | Strom_C | mafkees: this isn't for any clients I have already |
| 18:10.48 | Strom_C | these are hypothetical clients |
| 18:10.51 | Dr-Linux | question, is it for 1 SIP DID price? http://www.broadvoice.com/rateplans_unlimited_state.html |
| 18:10.53 | burizaa | [TK]D-Fender: what should i do? |
| 18:11.04 | justinu | <PROTECTED> |
| 18:11.06 | [TK]D-Fender | burizaa : .... are you not listening? |
| 18:11.15 | mafkees | Strom_C: ask bills to your current customers ;) |
| 18:11.25 | *** join/#asterisk kSquared (i=kkaitan@68-65-51-183.chvlva.adelphia.net) |
| 18:11.27 | Strom_C | groan |
| 18:11.30 | mafkees | sorry |
| 18:11.37 | burizaa | [TK]D-Fender: yes, i'm listeing... sorry for my bad english |
| 18:11.37 | Strom_C | that's not what I'm asking |
| 18:11.45 | [TK]D-Fender | burizaa : Pastbin your ENTIRE peer entry |
| 18:11.47 | kSquared | is there a way to play a sound from the Asterisk CLI? |
| 18:11.55 | [TK]D-Fender | burizaa : ALL OF IT. |
| 18:11.59 | mafkees | Strom_C: I dont think there is any golden rule |
| 18:12.00 | burizaa | okay |
| 18:12.16 | mafkees | it really depends on the size of the company, field they play in etc etc |
| 18:12.27 | Strom_C | of course there is no golden rule, but there is surely some sort of rough estimate based on type of industry |
| 18:12.39 | Strom_C | i.e. the average real estate office will make x calls of x duration in a given month |
| 18:12.50 | mafkees | I dont keep records |
| 18:12.58 | mafkees | because we call for free :) |
| 18:13.09 | mafkees | <--- working in a 5 ppl ict company |
| 18:13.23 | justinu | what's a free Mac OS irc client? |
| 18:13.33 | mafkees | justinu: xchat |
| 18:13.48 | justinu | cool |
| 18:13.48 | Dr-Linux | anybody knows SIP DID's price/mnt ? |
| 18:13.54 | mafkees | justinu: type: /version mafkees |
| 18:15.43 | burizaa | http://pastebin.ca/69242 |
| 18:15.51 | burizaa | thats all i got on my PEER |
| 18:15.53 | Dr-Linux | :S |
| 18:16.11 | Dr-Linux | any sip provider around? |
| 18:16.26 | Dr-Linux | who sell US/UK DID's |
| 18:17.58 | Spy000007 | voicepulse is the only one i see in this channel under an official name, there might be others, try sending a /msg or just look on the wiki under the "Cheapest Services and ATAs" page |
| 18:18.13 | Spy000007 | there's a section for "DIDs" |
| 18:18.41 | mafkees | Dr-Linux: http://www.voip-info.org/wiki/view/VOIP+Service+Providers |
| 18:19.03 | [TK]D-Fender | Dr-Linux : http://www.voip-info.org/wiki/ 1/3rd the way down the page is a GIANT FRIGGEN LIST. |
| 18:19.14 | [TK]D-Fender | <PROTECTED> |
| 18:19.29 | [TK]D-Fender | </rant> |
| 18:19.50 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 18:20.09 | kSquared | alright, so -- if I'm inside the asterisk CLI and connected, is there any way to use AGI through it? |
| 18:20.35 | mafkees | kSquared: to test the agi script ? |
| 18:20.37 | kSquared | specifically I'd like to use the "say number" command over a call to say a few numbers and test things out |
| 18:21.05 | mafkees | kSquared: agi scripts should run without error on the normal linux/bsd console |
| 18:21.09 | kSquared | I tried "show agi" to get a list of the commands (of which "say number <number>" is one of them) but there doesn't seem to be a way to actually execute them |
| 18:21.24 | kSquared | mafkees: I don't actually have a script written up |
| 18:21.26 | mafkees | to test it, create the agi and create a test extension |
| 18:21.40 | kSquared | I just wanted to dynamically execute that one line |
| 18:21.41 | mafkees | call to the test extension to see if the agi works |
| 18:22.27 | burizaa | [TK]D-Fender: have you take a look to my pastebin ? |
| 18:22.56 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 18:23.45 | [TK]D-Fender | burizaa : That isn't the full entry from sip.conf |
| 18:25.08 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 18:26.03 | burizaa | [TK]D-Fender: i dont have sip.conf ... i'm using freepbx |
| 18:26.20 | justinu | fender, you are truly masochistic |
| 18:26.32 | justinu | somebody buy this man a beer!! |
| 18:27.08 | Spy000007 | Wow, Broadvoice in PC WORLD today... |
| 18:28.01 | justinu | fender, got paypal? |
| 18:28.48 | Spy000007 | "Another VoIP provider, BroadVoice, is the eighth-most-complained-about company in eastern Massachusetts, Maine, and Vermont, according to the BBB office serving those areas." |
| 18:28.59 | [TK]D-Fender | burizaa : Yes you do. If you don't even know about the different config files, then you definately need to head to #freepbx |
| 18:29.05 | [TK]D-Fender | justinu : Yup |
| 18:29.09 | justinu | well what is it? |
| 18:29.13 | burizaa | http://pastebin.ca/69249 << i found it.. sip_additional.conf |
| 18:29.38 | CunningPike | smackus: Did you get sorted? |
| 18:31.16 | smackus | no |
| 18:31.20 | [TK]D-Fender | burizaa : And what is at that IP address? |
| 18:31.31 | smackus | just got forced to put new box into production early :-D |
| 18:31.37 | burizaa | [TK]D-Fender: quintum A800 |
| 18:31.48 | burizaa | sip configured |
| 18:32.19 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 18:33.53 | burizaa | now the SIP status become: UNREACHABLE |
| 18:33.59 | justinu | this is bullshit |
| 18:34.02 | justinu | The PayPal website is currently unavailable. We are actively working to restore access to the site as soon as possible. We apologize for the inconvenience. |
| 18:35.03 | Spy000007 | Doesn't Nufone's carrier-quality VoIP termination use Paypal? |
| 18:35.59 | file | justinu: can't win them all |
| 18:36.58 | Spy000007 | I usually charge for that outcome... |
| 18:38.32 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 18:39.12 | mafkees | anyone know how I can record the ringing sound on an asterisk box ? |
| 18:39.20 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 18:39.30 | CunningPike | mafkees: Use Monitor()? |
| 18:39.32 | TripleFFFF | question: is it possible to use userfield |
| 18:39.46 | mafkees | Monitor will record the ringing sound as well ? |
| 18:39.46 | TripleFFFF | i mean mmore hten one for a cdrrecord ? having more hten 1 user assigned field.. like.. |
| 18:39.50 | TripleFFFF | anything |
| 18:40.20 | *** part/#asterisk smackus (n=smackus@63.149.122.94) |
| 18:40.56 | mafkees | you can use the MySQL dialplan function or use an agi to write stuff in your own database/table |
| 18:40.57 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 18:42.19 | TripleFFFF | no |
| 18:42.22 | TripleFFFF | i cant use AGI |
| 18:42.29 | TripleFFFF | over 100 million cdr per day.. |
| 18:42.36 | TripleFFFF | i cant even think that |
| 18:42.49 | TripleFFFF | so i would need to ; seperate the userfiel.. and parse later ? |
| 18:43.05 | mafkees | that will be the easiest in a setup like that yeah |
| 18:43.25 | mafkees | because the MySQL command will be too heavy too, if agi is not holding it |
| 18:44.02 | mafkees | or use fastagi with some nice queueing script on the other side |
| 18:44.28 | CunningPike | mafkees: Don't know - try it! |
| 18:44.32 | CunningPike | :) |
| 18:46.13 | mafkees | actually, I think it will |
| 18:46.19 | mafkees | as long as you dont use the b flag to monitor |
| 18:48.00 | prodigy7 | have someone special experiences with fritzbox and asterisk behind it? |
| 18:49.34 | prodigy7 | and an other question too: is it possible to crypt the password in the sip.conf ? |
| 18:49.55 | *** join/#asterisk reza_ (n=reza@64-142-78-129.dsl.static.sonic.net) |
| 18:50.02 | reza_ | hey all |
| 18:51.12 | reza_ | what are those ethernet -> FXS devices called? |
| 18:51.16 | CunningPike | prodigy7: You mean like md5secret in iax? Don't think so |
| 18:51.22 | CunningPike | reza_: ATAs |
| 18:51.36 | reza_ | there's one that someone told me about that was good -- the s2000 or something like that |
| 18:51.43 | justinu | sipura 2000 |
| 18:51.55 | reza_ | excellent |
| 18:52.00 | reza_ | any recommendations for a good store? |
| 18:52.13 | CunningPike | reza_: Sears |
| 18:52.14 | vader-- | does ansterisk have a way to setup call forwarding by dialing an extension and entering the phone number you want to direct the calls to? |
| 18:52.15 | justinu | the guys at voipconnection.com are great |
| 18:52.24 | reza_ | i think sears sucks |
| 18:52.31 | Spy000007 | walmart |
| 18:52.38 | reza_ | vander- i just got that shit working |
| 18:52.41 | [TK]D-Fender | vader-- : It does after you invent it in extensions.conf |
| 18:52.43 | justinu | and no, i don't have any relationship with them, other than they ship when they say they're going to |
| 18:53.10 | justinu | i ordered a sipura 2100 for them, and had a tracking number in 20 mintues |
| 18:53.15 | mafkees | no, it doesnt record the ringing |
| 18:53.17 | mafkees | :( |
| 18:53.30 | mafkees | argh |
| 18:54.21 | vader-- | tkd i was using call forwarding on my cisco phones but the problem is the phones can handle two extensions but the phone will only forward for both not one of the other |
| 18:54.27 | [TK]D-Fender | justinu : THX :) |
| 18:54.34 | vader-- | i have a couple people who share offices and have two extensions on the phone |
| 18:54.37 | justinu | np |
| 18:54.44 | vader-- | if one person wants to forward their line it does it for both lines |
| 18:55.08 | mafkees | vader--: you use sip or sccp ? |
| 18:55.22 | vader-- | sip |
| 18:55.44 | mafkees | vader--: then you have to do it in the dialplan |
| 18:55.50 | vader-- | ya |
| 18:55.50 | mafkees | sccp can do it on the phone |
| 18:55.50 | *** join/#asterisk sevard (i=kynan@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 18:56.26 | reza_ | what sipura should i get |
| 18:56.32 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 18:56.33 | sevard | 2002 |
| 18:56.56 | _Sam-- | hey justinu, sorry to sound like the new guy...if im using CHanspy to listen in on my sales guys calls...what is the easiest way to setup something that would record them while i spy? |
| 18:57.07 | sevard | Monitor() |
| 18:57.20 | sevard | or is it Record() |
| 18:57.22 | sevard | one of the two |
| 18:57.24 | reza_ | hmm, dont see the 2002 on thevoipconnection.com store |
| 18:57.25 | _Sam-- | while im on a chanspy, if i just want to press a button and record? |
| 18:57.52 | sevard | if you want to just press a button and record things are messier, but if you run Monitor before you enter chanspy, that's easy |
| 18:57.56 | mafkees | monitor |
| 18:58.13 | sevard | reza_: no idea about them. the sipura 2002s are really nice though |
| 18:58.21 | _Sam-- | i just want to be able to press something while im in the middle of chanspy when i hear something i dont like...and then start recording |
| 18:58.25 | _Sam-- | not record every chanspy |
| 18:58.42 | mafkees | use the features.conf automon => |
| 18:58.47 | sevard | you could hook up a tape recorder :) |
| 18:58.49 | mafkees | dont know if it works for chanspy |
| 18:59.02 | reza_ | justinu / [TK]D-Fender -- which sipura's do you recommend |
| 18:59.14 | justinu | _Sam--: mixmonitor |
| 18:59.28 | justinu | reza: depends on what you want |
| 18:59.39 | justinu | i use sipura 2100... 2 lina ATA + nat router |
| 18:59.40 | reza_ | an ata that works |
| 18:59.57 | justinu | if you just need a single line, try the spa-1000 |
| 18:59.57 | reza_ | what's the diff w/ that and the 2002? |
| 19:00.05 | justinu | i think the 2002 is just an ata |
| 19:00.05 | reza_ | i need two lines, but no nat |
| 19:00.06 | justinu | no nat router |
| 19:00.15 | reza_ | ah; what do you want a nat router for? |
| 19:00.25 | justinu | home broadband |
| 19:00.28 | justinu | it does qos too |
| 19:00.52 | reza_ | hmm, i'm using a linux box as a router/nat -- i should figure out how to implement qos on it... |
| 19:01.03 | justinu | diffserve precedence queing |
| 19:02.03 | [TK]D-Fender | reza_ : reza what do you expect out of it, and in what kind of envirnment? |
| 19:02.18 | mafkees | I use openbsd for qos :) |
| 19:02.44 | reza_ | TK - you actually suggested I get one a few weeks back; forogt the exact model --- thogh 2002 seems right |
| 19:03.10 | [TK]D-Fender | reza_ : Quite likely, but I'd want to confirm its working envinment before I confirm that blind. |
| 19:03.28 | [TK]D-Fender | reza_ : You have a router alread right? |
| 19:03.53 | prodigy7 | is it possible to crypt the password in the sip.conf ? |
| 19:04.07 | reza_ | i'm running a linux router w/ asterisk and a tdm400 w/ 3 fxs no fxo -- a did from voxbone, and one voip phone |
| 19:04.16 | reza_ | you suggested the polycom ip 601 which i'm also about to order |
| 19:04.50 | [TK]D-Fender | reza_ : Hold that though. What are you going to DO with that IP 601? |
| 19:05.29 | reza_ | all calls go to the receptionist at the front desk |
| 19:05.34 | reza_ | she gets the 60 |
| 19:05.35 | reza_ | 501 |
| 19:05.36 | reza_ | 601 |
| 19:05.40 | sevard | dlynes_home: wake up |
| 19:05.55 | [TK]D-Fender | reza_ : I presume you meant just the 601 |
| 19:06.18 | reza_ | yeah |
| 19:06.20 | reza_ | cant type |
| 19:06.47 | reza_ | then she can route calls to the various phones via the spa-2002 |
| 19:06.53 | sevard | learning to type is a good thing to do before you go on irc |
| 19:07.26 | reza_ | servard - these nuts. |
| 19:07.34 | reza_ | hmm |
| 19:07.38 | reza_ | japan vs brazil has started |
| 19:07.46 | [TK]D-Fender | reza_ : so the SPA is for 2 internal extensions? |
| 19:08.01 | reza_ | well, i need 2 of them for 3 internal extentions |
| 19:09.18 | reza_ | am i cool with that order? |
| 19:09.50 | reza_ | hmm -- it's interesting that you can tell if you've got a runnaway process by the sound your computer fan makes |
| 19:09.58 | sevard | reza_: you want two CDs? |
| 19:10.23 | reza_ | CDs? |
| 19:10.27 | sevard | DEZ NUTS BEOCH |
| 19:10.45 | mafkees | lol reza_ |
| 19:11.34 | [TK]D-Fender | reza_ : OK, then that confirms it. Get the SPA-2002. You do NOT want the one with the built in router. |
| 19:12.02 | sevard | that's what I said to begin with |
| 19:12.04 | sevard | i'm freaking GOD |
| 19:12.26 | reza_ | anyone know what state discountvoipoutlet.com ships from? |
| 19:12.34 | sevard | call them |
| 19:12.37 | reza_ | i cant find it anywhere on their storefront, |
| 19:12.40 | reza_ | no hpone number |
| 19:12.41 | reza_ | ghay |
| 19:12.42 | *** join/#asterisk key2 (n=ashdown@sd-420.dedibox.fr) |
| 19:12.45 | Spy000007 | brazil has the hottest groupies |
| 19:12.56 | mafkees | wb key2 |
| 19:13.12 | [TK]D-Fender | then DIE for us :) |
| 19:13.22 | mafkees | ok, time to cleanup |
| 19:13.28 | mafkees | <--- away |
| 19:13.36 | sevard | i have two phones on speaker phone and i'm beeping my computer speaker |
| 19:13.42 | sevard | listening to the echo |
| 19:13.53 | sevard | i bet this is what being on acid is like |
| 19:13.58 | kSquared | lol |
| 19:14.12 | kSquared | would probably be more efficient just to run the echo demo :p |
| 19:14.24 | sevard | I AM THE GOLDEN GOD |
| 19:14.28 | *** join/#asterisk saftsack (n=oliver@p54A7FB70.dip.t-dialin.net) |
| 19:14.31 | saftsack | hi |
| 19:14.33 | sevard | ECHO DEMOS ARE FOR NUBS |
| 19:14.40 | sevard | hi saftsack |
| 19:14.42 | kSquared | speakig of which |
| 19:15.10 | TripleFFFF | !tell us about noobs |
| 19:15.14 | kSquared | I still can't believe there isn't a one-shot way to just play an arbitrary sound from the CLI >:| |
| 19:15.18 | TripleFFFF | oups.. |
| 19:15.22 | saftsack | are some hylafax experts here? i have a question. if faxgetty is written in the inittab as restart will it automatically started after rebooting the pc? |
| 19:15.34 | ptinsley | i am a big fan of calling from one polycom to the other on speakerphone and pretending i am a stadium announcer :) |
| 19:15.37 | TripleFFFF | softsack i dont know |
| 19:15.42 | TripleFFFF | saft better lol |
| 19:16.08 | sevard | ptinsley is my new friend |
| 19:16.10 | saftsack | ^^ |
| 19:16.18 | ptinsley | hehe |
| 19:16.25 | sevard | kSquared: in what since of the word, there is if you have a sound card |
| 19:16.42 | saftsack | and why do i need other TAE pluggers for fax devices as for normal analog telephones? |
| 19:16.52 | reza_ | ok, order is off |
| 19:17.17 | reza_ | and finally, they show thier address -- it's in florida -- glad i got 2nd day shipping |
| 19:21.48 | Bullseye_Network | Easy Question: I Dont need ztdummy just for a voicemail system correct? |
| 19:21.54 | Qwell[] | Bullseye_Network: correct |
| 19:21.57 | *** join/#asterisk knight_ (n=root@blackhole.phunc.com) |
| 19:21.58 | Bullseye_Network | thanks |
| 19:22.05 | knight_ | hey driz |
| 19:22.19 | knight_ | twisted |
| 19:22.49 | knight_ | hey can anyone recommend cheap and affordable desktop ip phones that offer extensive feature sets (like web content on the display, etc) |
| 19:23.08 | Strom_C | define "cheap and affordable" |
| 19:23.10 | Qwell[] | 17 days?!@ |
| 19:23.14 | file | omg omg omg |
| 19:23.21 | Qwell[] | heh |
| 19:23.25 | Qwell[] | my boss just pwned himself |
| 19:23.46 | Qwell[] | basically tore the desk out of the wall..heh, moron |
| 19:23.47 | Bullseye_Network | If anybody is interested awhile back I make all the software on quadrasoftware.com opensource. The software was never updated to work with 1.2 but some of you might be interested in some of the code. Its all in Visual Basic. |
| 19:23.52 | knight_ | Strom, home use |
| 19:23.56 | knight_ | so cheap |
| 19:23.57 | knight_ | :) |
| 19:24.06 | file | Qwell[]: ooh |
| 19:24.06 | Strom_C | knight_: I was looking for a dollar value |
| 19:24.07 | existx | i had a boss that stomped on a mobo once |
| 19:24.11 | knight_ | I have 5.8ghz portables connected to TDM400's right now |
| 19:24.16 | knight_ | Strom, $100-200 |
| 19:24.25 | Strom_C | knight_: used cisco 7940/7960 |
| 19:24.29 | knight_ | yeah |
| 19:24.34 | existx | the metal spoke that's used to clip the cpu went into his foot |
| 19:24.35 | knight_ | i have a 7960 on my desk here at work now |
| 19:24.37 | existx | it was funny |
| 19:24.44 | Qwell[] | existx: ouch |
| 19:24.46 | knight_ | but lots of people here said they'd never buy these ones again |
| 19:24.53 | Strom_C | knight_: I quite like mine |
| 19:24.55 | existx | Qwell[]: that'll teach him :) |
| 19:24.58 | Qwell[] | knight_: feel free to ship yours here |
| 19:25.08 | Strom_C | knight_: or send them to me :) |
| 19:25.08 | knight_ | Qwell, I'm not complaining. |
| 19:25.13 | Qwell[] | :p |
| 19:25.14 | knight_ | Besides, they're not mine. |
| 19:25.17 | ptinsley | Qwell[], how did he manage that |
| 19:25.22 | vader-- | is this valid |
| 19:25.22 | vader-- | exten => s,1,Set(temp=${DB(data/${EXTEN})}) |
| 19:25.23 | Qwell[] | ptinsley: he's a doofus |
| 19:25.29 | file | Qwell[]: muffintastic! |
| 19:25.46 | Qwell[] | file: skinnytabulous! |
| 19:26.01 | Strom_C | dogballsandcheeseoriffic! |
| 19:26.15 | Qwell[] | Strom_C: Why's everything always gotta end up "dog balls"? |
| 19:26.23 | Strom_C | because dogballs are amusing |
| 19:26.43 | file | according to you. |
| 19:26.47 | Qwell[] | indeed |
| 19:27.28 | ptinsley | i prefer monkey |
| 19:27.34 | justinu | to each his own, i guess |
| 19:27.41 | sevard | yout mom |
| 19:28.05 | knight_ | any alternatives to the 7960? |
| 19:28.08 | knight_ | that is comparable? |
| 19:28.09 | Qwell[] | knight_: 7940 |
| 19:28.32 | vader-- | anyone know why this line exten => s,1,Set(temp=${DB(data/${EXTEN})}) would cause this output in the asterisk console Executing Set("SIP/001759E558CE-02-6369", "temp=") in new stack |
| 19:28.47 | Qwell[] | vader--: because that entry doesn't exist |
| 19:28.47 | knight_ | qwell, isnt that a step down? |
| 19:28.51 | Qwell[] | knight_: only barely |
| 19:28.54 | knight_ | also, any other brands? |
| 19:29.10 | Qwell[] | people here like their polycom 601s |
| 19:29.23 | Strom_C | i think the polycom phones are a pain in the ass to set up |
| 19:29.28 | Qwell[] | probably |
| 19:29.48 | jake1932 | the new Sipura ones actually look good |
| 19:29.57 | justinu | if you hate XML, don't buy a polycom |
| 19:29.58 | jake1932 | haven't tried em yet though |
| 19:30.30 | ptinsley | after you figure out the quirks of the polycom phones they are pretty good |
| 19:30.40 | ptinsley | my big complaint so far is the directory functions on the phone |
| 19:31.07 | Strom_C | Qwell[]: http://www.stromcarlson.com/misc/balls.png |
| 19:31.09 | ptinsley | it supports a global directory but there is no good way to update it without a reboot of every phone in a company when you add/edit someone |
| 19:31.11 | Qwell[] | no thank you |
| 19:31.16 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 19:31.20 | Strom_C | Qwell[]: it's safe for work |
| 19:31.52 | file | it's safe |
| 19:32.04 | sevard | file is full of shtie |
| 19:32.05 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 19:32.10 | heison | `seen bkw |
| 19:32.17 | heison | ~seen bkw_ |
| 19:32.20 | jbot | bkw_ is currently on #asterisk (1d 5h 1m 1s). Has said a total of 2 messages. Is idling for 1d 4h 48m 27s, last said: 'Jun 21 06:24:33 NOTICE[16882]: chan_iax2.c:3123 iax2_read: I should never be called!'. |
| 19:32.33 | sevard | Strom_C: that's sfw :P |
| 19:32.46 | Strom_C | sevard: ? |
| 19:32.55 | sevard | safe for work. |
| 19:32.56 | sevard | bunghole |
| 19:33.55 | Spy000007 | Worst Logos Ever -- http://www.manic.com.sg/blog/archives/000305.php |
| 19:33.56 | TESTER2 | Where can I enable MWI on my zap channel (fxs module of a tdm400p)? |
| 19:34.03 | Spy000007 | sfw |
| 19:34.06 | smackus | ok, need help with my zapata.conf |
| 19:34.15 | justinu | Spy000007: lol |
| 19:34.16 | smackus | http://pastebin.ca/69283 |
| 19:34.26 | smackus | Extension '6406' in context 'allcallsinbound' from '8015582352' does not exist. Rejecting call on channel 0/8, span 1 |
| 19:34.38 | smackus | it is not doing multiple contexts. i have in my extensions.conf [progrexion] and [evolution] |
| 19:34.39 | file | smackus: I think that's self explanitory |
| 19:34.44 | smackus | its not |
| 19:34.46 | smackus | i am frazzled |
| 19:34.48 | smackus | i need help |
| 19:35.09 | smackus | i need one t1 to dial each |
| 19:35.12 | sevard | Spy000007: that's pretty awesome |
| 19:35.23 | file | smackus: it searched in the allcallsinbound context and found no extension... if you want it to search other contexts, include them |
| 19:35.36 | smackus | ok |
| 19:35.46 | Spy000007 | scroll down for animation |
| 19:35.47 | file | if Asterisk just searched every context in existence it would be very very insecure :) |
| 19:35.49 | smackus | someone had turned me away from that yesterday. |
| 19:35.53 | sevard | yeah, that's the best part |
| 19:35.58 | smackus | how do i do more than one context in one group |
| 19:36.00 | Qwell[] | file: how silly |
| 19:36.08 | file | well, you have to figure out how you want it to work... I can't read your mind |
| 19:36.12 | file | I can only tell you how things work |
| 19:36.14 | sevard | asterisk? secure? haha |
| 19:36.19 | ptinsley | smackus, you do an include in the extensions.conf not in the zapata |
| 19:36.24 | ptinsley | leave the interface in your inbound context |
| 19:36.34 | ptinsley | but include the numbers you want to be able to be called from there |
| 19:36.40 | Bullseye_Network | y |
| 19:36.41 | ptinsley | just be careful what you include |
| 19:36.43 | Bullseye_Network | ops |
| 19:36.52 | Bullseye_Network | wrong window |
| 19:37.05 | Idle | Qwell[] smells |
| 19:37.23 | smackus | file: so in the extensions.conf do I need to create a [allcallsinbound]? |
| 19:37.31 | justinu | yes |
| 19:37.32 | smackus | or can i just do the include |
| 19:37.42 | file | a context has to exist, in order to search it |
| 19:37.46 | ptinsley | yes, create that context and include what you want to be accessable |
| 19:37.55 | vader-- | is there a console command to check the asterisk db? |
| 19:37.56 | smackus | so what would go in it... nothing? |
| 19:38.03 | justinu | all your DIDs? |
| 19:38.04 | file | I would also highly suggest learning this stuff |
| 19:38.07 | Idle | :D |
| 19:38.15 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 19:38.17 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 19:38.25 | Idle | Qwell[]: I still need a machine with PCI 2.2.... gimme your desktop |
| 19:38.29 | Qwell[] | k |
| 19:38.34 | smackus | ok, i had that before, i think i am too frazzled since my other system crashed, i am just f'ing it all up now. |
| 19:38.38 | smackus | sorry to bother |
| 19:38.38 | Idle | you pay the shipping too |
| 19:38.42 | Strom_C | smackus: for the sake of you and everyone else in #asterisk, I would strongly recommend that you at least read the asterisk book :) |
| 19:38.42 | Qwell[] | Idle: k |
| 19:38.42 | ptinsley | or you can do goto's in the inbound context like this: exten => 6404,1,Goto(phones,6404,1) |
| 19:38.58 | Qwell[] | Idle: but I'm shipping via UPS |
| 19:38.59 | justinu | smackus: chill out, take a deep breath, smoke'm if you got'm |
| 19:39.02 | Qwell[] | SUCKER |
| 19:39.02 | *** join/#asterisk Meaty (n=meaty3@66.254.41.11) |
| 19:39.07 | Idle | just get jj to carry it on his back... so that when he gets here, I can kick him in the nuts, for everyone on efnet |
| 19:39.11 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 19:39.35 | justinu | rochambeau him for it |
| 19:39.55 | mountainm2k | ptinsley: another question on the provisioning here? |
| 19:40.02 | ptinsley | mountainm2k, sure |
| 19:40.23 | mountainm2k | I set up address, etc in sip.cf... phone1.cfg a few global things... |
| 19:40.29 | Idle | how does everyone find an old box that someones throwing away, but all I can find are p120's and 486's |
| 19:40.46 | mountainm2k | MAC.cfg is where I'm trying to specify the auth, display name, etc... |
| 19:40.55 | ptinsley | right |
| 19:40.57 | mountainm2k | that way I only have MAC.cfg, phone1.cfg and sip.cfg |
| 19:41.08 | Qwell[] | Idle: ebay.ca? |
| 19:41.16 | mountainm2k | the manual indicates I should have MAC.cfg which points to EXTENSION.cfg, and sip.cfg |
| 19:41.17 | Idle | I dont want to PAY for one |
| 19:41.23 | Idle | well, even ship one |
| 19:41.30 | Qwell[] | go rob somebody.. |
| 19:41.33 | mountainm2k | which would mean each phone has a copy of the global config, which kindof defeats the purpose |
| 19:41.33 | Idle | I bet that 120 woulda run awesome |
| 19:41.40 | Qwell[] | but? |
| 19:41.40 | Idle | stupid PCI 2.0 |
| 19:41.57 | kSquared | sevard: the server doesn't have a sound card, unfortunately |
| 19:41.59 | ptinsley | well, not really, it just tells you to reference them |
| 19:42.20 | ptinsley | and the polycom way is to have a MAC.cfg which points at phoneMAC.cfg and have the guts in phoneMAC.cfg |
| 19:42.36 | ptinsley | but you don't have to do that, you can just have MAC.cfg if you want |
| 19:42.37 | Qwell[] | Idle: Did you at least snag the ram? :D |
| 19:42.39 | justinu | mmm guts |
| 19:42.39 | mountainm2k | yeah, but then I have two files for each phone instead of only one |
| 19:42.50 | justinu | mountainm2k: it's the way |
| 19:42.59 | Idle | ooh, I should ask the local LUG |
| 19:42.59 | mountainm2k | I'd rather have only one file for each phone, and then the generic ones that apply to all phones |
| 19:43.01 | CunningPike | mountainm2k: sip.cfg _is_ your global config - it sets stuff that is the same for every phone. <phone>.cfg is your set-specific config. <mac-address>.cfg links them to your set's MAC address |
| 19:43.02 | Idle | Qwell[]: no |
| 19:43.04 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 19:43.05 | Qwell[] | lame |
| 19:43.08 | Idle | its still sitting in my basement |
| 19:43.13 | Idle | SDRAM |
| 19:43.15 | Idle | SIMMs |
| 19:43.16 | Idle | w00t |
| 19:43.19 | Qwell[] | oh, so you snagged the whole thing |
| 19:43.24 | ptinsley | an example of MAC.cfg is something like this: <APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone0004f2029a3f.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="/log/" OVERRIDES_DIRECTORY="/overrides/" CONTACTS_DIRECTORY="/contacts/"/> |
| 19:43.44 | CunningPike | mountainm2k: When you're looking to add a line appearance to extension 2348, it's a lot easier to find phone2348.cfg and edit it than look for 00043479287492.cfg |
| 19:43.52 | Idle | uhm, not really |
| 19:43.53 | Idle | its mine |
| 19:43.57 | Idle | was my first computer ever |
| 19:44.01 | Qwell[] | I see |
| 19:44.12 | Idle | that thing was the shit for 5 years |
| 19:44.18 | Idle | well, sorta |
| 19:44.26 | justinu | what computer is "the shit" for 5 years? |
| 19:44.32 | Idle | it was the shit for the first 2, then it was just shit for the later 3 |
| 19:44.36 | justinu | most of them suck after 6 months |
| 19:44.53 | mountainm2k | ptinsley: So can I add to MAC.cfg anything from phone1.cfg? Like, say: <phone1> |
| 19:44.54 | mountainm2k | <PROTECTED> |
| 19:44.54 | mountainm2k | </phone1> |
| 19:44.59 | Idle | justinu: this thing was cutting edge when we first got it |
| 19:45.03 | mountainm2k | because it doesn't seem to work that way... |
| 19:45.04 | Idle | had a Mach64 video card :D |
| 19:45.25 | ptinsley | does it work with two files? |
| 19:45.37 | Bullseye_Network | How would I disable SIP all together on an asterisk box? noload => chan_sip.so ? |
| 19:45.39 | mountainm2k | Havn't tried, trying to avoid that... :-P |
| 19:45.47 | mountainm2k | I'll give it a shot |
| 19:45.54 | ptinsley | what isn't it doing? |
| 19:46.01 | CunningPike | mountainm2k: I recommend it anyway, for the reason I gave earlier |
| 19:46.13 | *** join/#asterisk Assid (i=assid@PPP-219.65.7.68.mum1.dialup.vsnl.net.in) |
| 19:46.30 | mountainm2k | It doesn't seem take the items in MAC.cfg -- the userid (extension), display name, etc... |
| 19:46.40 | mountainm2k | but it _does_ have the SIP server, which is coming from phone1.cfg |
| 19:47.25 | ptinsley | there are some scripts floating around out there that can help you with phone provisioning for polycom stuff it might be worth a google if you are going to do alot of them |
| 19:48.01 | ptinsley | if the concern is the two files that would cut down on your work a great deal |
| 19:48.03 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 19:48.13 | TESTER2 | I get a special dialtone when MWI is on but the MWI ligth (on the analog phone) stay close... any special option? (FXS module on a tdm400p with mailbox= in zapata.conf) ? |
| 19:49.01 | mountainm2k | Well, my concern is for starters two files instead of one, but additionally I want to keep everything as global / generic as possible... and phone1.cfg contains a _lot_ of stuff... |
| 19:49.18 | mountainm2k | I wanted to get it down to, for each phone, specify the extension/user/etc, then load the global config |
| 19:49.54 | mountainm2k | it _does_ work when I put it in phone1.cfg -- so maybe I just need to give up and do it that way... |
| 19:50.18 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 19:50.22 | justinu | resistance is futile |
| 19:50.28 | mountainm2k | heh |
| 19:51.06 | mountainm2k | still wish I could globalize it a bit more... |
| 19:51.38 | justinu | just write/steal a script to gen the files for you |
| 19:51.39 | mountainm2k | so I guess the next question is this -- can I edit the "dialplan" functionality to make it provide a second dialtone after they hit "9" ? |
| 19:51.58 | ptinsley | well you can't really put it all in phone1.cfg if you have more than one phone ;) |
| 19:52.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-91-78.red.bezeqint.net) |
| 19:52.40 | mountainm2k | Well, like I said, the goal was to have a single file with the per-phone info, not a single file that loads another single file that is fairly large |
| 19:52.43 | justinu | my polycoms to HTTPS provisioning, theoretically you could even write a CGI to gen the files |
| 19:52.50 | ptinsley | my phoneMAC.cfg file only contains the <phone1><reg.... /> and <msg .../> sections |
| 19:53.05 | mountainm2k | Interesting... |
| 19:53.14 | mountainm2k | Well, there's a lot of other crap in the template |
| 19:53.39 | ptinsley | justinu, the only reason I haven't gone that direction is for log files and overrides on the handset that need to be saved to the server incase the phone forgets something |
| 19:53.46 | syzygybsd | i have a Sangoma a104u. at idle it has 4000 Interrupts a second, and under about a 10 call load it drops to 3000. Does anyone know if this will cause issues under heavyer load? |
| 19:54.00 | justinu | ptinsley: you can do that as well, you just need mod_put in your apache server |
| 19:54.24 | ptinsley | ah, cool, I hadn't looked into it that deeply, the ftp stuff has been working, no reason to fix it if it isn't broken :) |
| 19:54.58 | ptinsley | it would be optimal for me long term though, all of my configs are generated from database and written to the ftp server so the phones can pick them up. Would be easier to just have them generated on the fly when they were needed |
| 19:55.07 | vader-- | does anyone see anything wrong with this? |
| 19:55.08 | mountainm2k | Thinking.... (ouch)... There's no reason I can't have more than just phone.cfg and sip.cfg, etiher... |
| 19:55.09 | vader-- | exten => s,1,GotoIf($DB_EXISTS(data/$EXTEN)}?2:7) |
| 19:55.11 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 19:55.18 | mountainm2k | ;blinks |
| 19:55.41 | ptinsley | if you had multiple ftp servers, otherwise sip.cfg is sip.cfg to every phone that goes there |
| 19:55.42 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 19:55.57 | sevard | aaaaaaaaaaaaaaaaa |
| 19:56.03 | sevard | no more tetris please |
| 19:56.12 | mountainm2k | but I could have a generic.cfg and a phone.cfg and load them in the right order from MAC.cfg I would guess? |
| 19:56.30 | ghento | Hi there. I want to use asterisk to call a number just to check if the phone number is working, i don't need the other phone to ring & answer. Can this be done? |
| 19:56.46 | ptinsley | actually yes for sip you could do that but phone1 i think is a special one |
| 19:57.01 | [TK]D-Fender | vader-- : exten => s,1,GotoIf(${DB_EXISTS(data/${AVariableNothTheExtenBecuaseItsS}}?2:7) |
| 19:57.29 | ptinsley | but your best bet is to treat sip.cfg and phone1.cfg as global and do everything from MAC.cfg and phoneMAC.cfg or by phoneEXT.cfg as was suggested for easy admin |
| 19:58.01 | justinu | ghento: how will you determine it's working? |
| 19:58.43 | ghento | justinu: just if a connection is made i guess..i'm not exactly sure :) |
| 19:58.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 19:59.29 | *** join/#asterisk the_beginner__ (n=toasty@user-314.l2.c2.dsl.pol.co.uk) |
| 19:59.49 | ghento | if the correct dial-tones are returned or something |
| 20:00.21 | justinu | ghento: that's generally unreliable unless you use something with some real supervision protocols, like PRI |
| 20:02.08 | vader-- | thanks fender |
| 20:02.49 | mafkees | ptinsley: you found the soundfile for me ? |
| 20:02.50 | mafkees | ;) |
| 20:03.04 | mafkees | I tried with monitor, but that is not recording the indications ~( |
| 20:03.14 | ptinsley | well I made one but I haven't had time to convert it to gsm because I am in the middle of something |
| 20:03.20 | ptinsley | just wondering if I shoudl finish it when I get done |
| 20:03.32 | mafkees | ok |
| 20:03.38 | mafkees | if it's .wav |
| 20:03.43 | mafkees | I can convert it too |
| 20:04.00 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 20:04.01 | mafkees | do that kindda stuff all the time for customers that send me .wav for their ivr |
| 20:04.15 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 20:04.20 | mafkees | hey oe |
| 20:04.23 | mafkees | hey oej |
| 20:04.55 | ptinsley | any sangoma experts in here? |
| 20:05.15 | justinu | i dunno about experts, but there's some collective knowledge |
| 20:05.16 | mafkees | I'm not an expert |
| 20:05.19 | mafkees | but I use them |
| 20:05.30 | mafkees | only PRI though |
| 20:05.42 | ptinsley | ok, have you ever had one of the a200 analog cards wtih echo canceling hardware play no audio? |
| 20:05.49 | *** part/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 20:05.58 | mafkees | nope |
| 20:06.00 | justinu | sorry, no experience with the analog cards |
| 20:06.04 | sticks | Hi. I am brand new to asterisk. Does anybody have any idea why, when I run "asterisk -cv" to get the asterisk console, I get continuous white noise out my speakers? I am running it on NetBSD. |
| 20:06.04 | eBody | anyone not able to use the text messaging w/ xLite?? |
| 20:06.05 | mafkees | me neither |
| 20:06.14 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 20:06.14 | Hmmhesays | ok something is wrong with my Makefile |
| 20:06.16 | Hmmhesays | hrm |
| 20:07.12 | mafkees | here in .nl analog is only for home use |
| 20:07.34 | mafkees | we have ISDN BRI |
| 20:07.34 | sevard | <PROTECTED> |
| 20:07.36 | sevard | shite |
| 20:07.48 | ptinsley | i have two pbx's with the same hardware config 4 port fxo a200 echo canceler |
| 20:08.01 | ptinsley | and the EXACT same software install, one has audio with echo canceler enabled, one doesn't |
| 20:08.25 | Qwell[] | ptinsley: call sangoma? |
| 20:08.36 | ptinsley | well I was just wondering if anybody had seen it |
| 20:08.47 | Qwell[] | I'm sure they have :) |
| 20:08.50 | ptinsley | hehehe |
| 20:11.27 | ghento | justinu: okay thanks, will look into PRI |
| 20:11.53 | mafkees | sangoma is really linux minded |
| 20:12.03 | mafkees | I have this S518 dsl nic |
| 20:12.07 | justinu | their wanpipe stuff is in the kernel distro |
| 20:12.10 | justinu | so yeah |
| 20:12.15 | [TK]D-Fender | ptinsley : Yes, you can lock up the audio if the PID is desynched on the EC module. |
| 20:12.19 | mafkees | first thing they asked was: what linux |
| 20:12.25 | mafkees | so I replied: OpenBSD |
| 20:12.35 | [TK]D-Fender | ptinsley : Effectively all audio goes in, but never gets out. |
| 20:12.45 | mafkees | they had to patch me through to some longhaired guy in the basement |
| 20:12.47 | mafkees | lol |
| 20:13.14 | mafkees | support couldnt help me |
| 20:13.37 | mafkees | and the new version of that nic is not supported on openbsd :( |
| 20:13.48 | mafkees | the nic I have is borked |
| 20:13.51 | sevard | try linux |
| 20:14.00 | sevard | it's like bsd |
| 20:14.03 | sevard | except less greif |
| 20:14.10 | ptinsley | [TK]D-Fender, it's like a roach motel |
| 20:14.11 | sevard | grief |
| 20:14.23 | mog | linux is the coolest.... |
| 20:14.28 | mafkees | sevard: uhhuh, when linux comes with realtime nat failover and trafficshaping in iptables I'll look into it |
| 20:14.49 | sevard | whatever that crap is, my linookz makes me happy |
| 20:14.56 | mafkees | mine too |
| 20:15.01 | mafkees | but not for border firewalls |
| 20:15.07 | sevard | use cisco for that. |
| 20:15.12 | mafkees | no |
| 20:15.16 | mafkees | I prefer bsd |
| 20:15.27 | sevard | i tried bsd |
| 20:15.45 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 20:15.47 | sevard | the only thing i ever liked that ran bsd was that little mac notebook my school borrowed to me for two years |
| 20:15.54 | sevard | pretty |
| 20:19.00 | Hmmhesays | fuck |
| 20:19.14 | [TK]D-Fender | ~sex |
| 20:19.18 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 20:20.07 | Hmmhesays | I think that is the lamest thing i've seen today |
| 20:20.21 | sevard | i thought it was pretty funny |
| 20:20.36 | sevard | forgot to check uptime before remove condom |
| 20:20.40 | sevard | erm, unmount |
| 20:22.44 | mitcheloc | why'd they still the ~sex from #freepbx! (or did jbot know it first?) =/ |
| 20:23.29 | *** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
| 20:23.31 | podzap | hi |
| 20:24.15 | podzap | is it possible to send a fax from a real fax machine -> sipura spa-2002 -> asterisk, and have asterisk save it as e.g. a tiff file? |
| 20:24.16 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 20:24.50 | podzap | my use case is that i need to sign a document and email it to somebody, but i don't have a scanner. |
| 20:25.18 | mountainm2k | OK, back to my Polycom issue -- I told MAC.cfg to load three files, phone201.cfg, genericphone.cfg, and sip.cfg |
| 20:25.29 | brad_mssw | podzap: spandsp / rxfax |
| 20:25.32 | mountainm2k | the line1 key now shows "...t IP" |
| 20:25.36 | mountainm2k | but I don't know what that means |
| 20:25.47 | Spy000007 | they sell scanners for $20 at best buy on black friday |
| 20:26.20 | podzap | brad_mssw: thanks, man |
| 20:26.24 | sevard | Spy000007: that's if you get past the 3000 screaming people who all got there 3 days before at 3:30 a.m. |
| 20:26.33 | *** part/#asterisk sticks (n=sticks@ip68-12-170-34.ok.ok.cox.net) |
| 20:27.22 | podzap | Spy000007: what makes you assume that i live in the USA? |
| 20:30.02 | mountainm2k | What's it mean when I get a fast-busy when dialing out, but incoming calls work? |
| 20:31.45 | mafkees | mountainm2k: your extensions.conf section for outbound calls is wrong |
| 20:32.05 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:32.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:32.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:32.45 | smackus | ok |
| 20:32.56 | ptinsley | Qwell[], thanks for pointing out the obvious, support says it's a bad module on the card, i need smacking every once in awhile |
| 20:32.56 | smackus | I need some help with my T1s and zaptel |
| 20:33.00 | smackus | http://pastebin.ca/69324 |
| 20:33.07 | smackus | I have two t1s |
| 20:33.16 | smackus | the first t1 is working fine. |
| 20:33.18 | smackus | the second one is not |
| 20:33.57 | smackus | if i swap the cables from port 1 to port 2 and from port 2 to port 1, the one that was working, does not work, and the other one that was not, starts. |
| 20:34.01 | smackus | can someone gimme a hand? |
| 20:34.44 | mafkees | smackus: did you call your provider ? |
| 20:34.57 | mafkees | sounds like there's something wrong with the line |
| 20:35.33 | smackus | it works if I put it into channels 1-23 on the card. |
| 20:35.42 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 20:35.56 | smackus | it seems like only the 1st port on the card is configured correctly, because no matter what t1 i put in there it works |
| 20:36.09 | smackus | if i take a working one and put it into port 2, it stops working. |
| 20:36.24 | smackus | and it all worked in the other system |
| 20:36.32 | mafkees | are the jumpers on the card correct |
| 20:36.56 | smackus | let me verify. |
| 20:37.04 | smackus | for a pri they should be set to e1 right? |
| 20:39.10 | mafkees | if you are having a PRI with 30 channels, yeah |
| 20:39.18 | mafkees | e1 == european |
| 20:39.28 | mafkees | t1 == us style, it's 24 channels |
| 20:39.50 | mafkees | and j1 == ??? |
| 20:39.55 | smackus | ok |
| 20:39.59 | mafkees | japanese PRI |
| 20:40.10 | smackus | when i do a zap show channels it only shows 1-23 |
| 20:40.20 | smackus | shouldnt it have 1-95 minus my dcahn? |
| 20:40.22 | smackus | chan |
| 20:40.33 | mafkees | 1-23 == T1 |
| 20:40.40 | mafkees | it's the first port |
| 20:40.51 | smackus | right...shouldnt it show all the ports with that command? |
| 20:41.03 | mafkees | yes, if all is configured correct |
| 20:42.02 | smackus | ok... so now i know that it is misconfigured. |
| 20:42.07 | mafkees | your zapata is correct |
| 20:42.20 | mafkees | can you pastebin your /etc/zaptel.conf ? |
| 20:42.29 | smackus | yep, i was just gonna ask if you wanted it. |
| 20:42.38 | userdefined | so, it turns out that even if i use SER as a front to * i still need to figure out how to send all call to LCS from asterisk clients to SER ... |
| 20:43.22 | smackus | http://pastebin.ca/69361 |
| 20:44.05 | mafkees | hhmm |
| 20:44.19 | userdefined | is it correct that * would need to register to the SER as a user ? (similar to the FWD config in the sample .conf ?) |
| 20:44.22 | mafkees | what does ztcfg -vvv tell you ? |
| 20:44.39 | smackus | hang on. |
| 20:44.55 | userdefined | then set up extensions.conf such that anything to @lcsdomain is sent to SER? |
| 20:45.16 | mafkees | userdefined: I have no idea |
| 20:45.21 | mafkees | I never played with cer |
| 20:45.21 | smackus | mafkees: i run that at the prompt, right? because it gives me absolutely nothing |
| 20:45.26 | mafkees | yes |
| 20:45.30 | mafkees | it's a linux command |
| 20:45.42 | mafkees | run it as root |
| 20:45.45 | smackus | yeah, i just returns the next line. |
| 20:45.46 | smackus | did |
| 20:46.08 | smackus | [root@localhost etc]# ztcfg |
| 20:46.08 | smackus | [root@localhost etc]# |
| 20:46.34 | mafkees | ztcfg -vvv |
| 20:47.14 | Zodiacal | anyone know of phones that have the concept of normal line buttons. i.e. bob is on line one. etc? (not parking tho) |
| 20:47.24 | Zodiacal | or is this something asterisk can't really do |
| 20:47.42 | mafkees | Zodiacal: snom, cisco, polycom |
| 20:47.53 | Zodiacal | mafkess cisco? |
| 20:48.00 | smackus | mafkees: http://pastebin.ca/69365 |
| 20:48.01 | Zodiacal | 7960's show calls in a list |
| 20:48.21 | mafkees | but it has those 6 softkeys on the right |
| 20:48.30 | Qwell[] | line keys |
| 20:48.33 | Zodiacal | mafkees those can be used as lines? i thought just speed dials |
| 20:48.35 | mafkees | they can be configured to be lines, or speeddials |
| 20:48.39 | Zodiacal | oic |
| 20:48.42 | Qwell[] | or, on skinny, anything you like |
| 20:48.47 | Zodiacal | interesting |
| 20:48.55 | Qwell[] | or, rather |
| 20:48.59 | Qwell[] | anything you pay me for :P |
| 20:49.02 | Zodiacal | :P |
| 20:49.11 | Zodiacal | i have chan_sccp running, but they are speeddials |
| 20:49.15 | Zodiacal | how would i get them lines off hand? |
| 20:49.19 | Qwell[] | well, chan_sccp sucks |
| 20:49.21 | mafkees | smackus: restart asterisk please |
| 20:49.25 | Qwell[] | Zodiacal: add more line => lines |
| 20:49.25 | smackus | ok |
| 20:49.32 | mafkees | Qwell ehm, not in my opinion |
| 20:49.41 | Qwell[] | mafkees: opinion or not...it sucks |
| 20:49.52 | smackus | hmmm. seems to have made a difference. |
| 20:49.53 | Qwell[] | it's factual :) |
| 20:49.55 | Zodiacal | qwell and other people can share those lines? |
| 20:49.55 | smackus | let me make a call |
| 20:49.57 | file | Qwell[]: easy boy! |
| 20:49.59 | Qwell[] | Zodiacal: no |
| 20:50.06 | Zodiacal | qwell so two phones can be configured this way? |
| 20:50.06 | Hmmhesays | blargh! |
| 20:50.07 | smackus | i hope it was just that easy and that I am a dumb ass |
| 20:50.12 | Hmmhesays | so close to getting this to work |
| 20:50.14 | Hmmhesays | SO CLOSE |
| 20:50.24 | Qwell[] | Zodiacal: none of the asterisk skinny drivers currently supoprt shared lines |
| 20:50.25 | mafkees | smackus: a reload will not be enough for zaptel |
| 20:50.35 | Qwell[] | they *could*, but... |
| 20:50.49 | Zodiacal | what about sip? |
| 20:51.00 | mafkees | Qwell: I love chan_sccp.so |
| 20:51.10 | Qwell[] | mafkees: doesn't mean it doesn't suck |
| 20:51.31 | Zodiacal | qwell what sccp driver do you prefer? |
| 20:51.40 | Qwell[] | Zodiacal: the one I'm fixing :P |
| 20:51.51 | mafkees | Qwell: well, it may suxor, but it gives me way more functions on my phone then the sip thing |
| 20:51.59 | smackus | seems to have fixed it... |
| 20:52.02 | mafkees | Qwell: you working on chan_skinny.so ? |
| 20:52.07 | smackus | i did reload and everything but restarting asterisk |
| 20:52.08 | Qwell[] | I am |
| 20:52.10 | mafkees | smackus: it's working now ? |
| 20:52.10 | smackus | thanks for the help |
| 20:52.12 | smackus | yeah |
| 20:52.13 | smackus | go figure |
| 20:52.17 | mafkees | ;) |
| 20:52.21 | Zodiacal | qwell i have noticed some strange things with the chan_sccp . i.e. if you press a speed dial 3 times it dials it but the screen loses the current call info.. very strange |
| 20:52.22 | smackus | do you normally have to restart asterisk? |
| 20:52.24 | *** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 20:52.33 | Qwell[] | Zodiacal: like I said...it sucks |
| 20:52.41 | mafkees | Qwell ok, nice. but I wont comment on it here |
| 20:52.46 | Qwell[] | chan_skinny sucks too, but that's okay |
| 20:52.53 | mafkees | <--- remembers a nasty thread on the mailinglist |
| 20:52.57 | Zodiacal | qwell do you know if sip supports line sharing? |
| 20:53.02 | Qwell[] | Zodiacal: no, it doesn't |
| 20:53.08 | Zodiacal | does anything? |
| 20:53.09 | Zodiacal | :P |
| 20:53.18 | Qwell[] | currently? no :p |
| 20:53.19 | Qwell[] | ccm |
| 20:53.19 | mafkees | smackus: no, but for zaptel shit most of the time a restart is needed |
| 20:53.19 | file | there's a spec for it... bridged line appearances |
| 20:53.24 | Zodiacal | ccm? |
| 20:53.34 | mafkees | ccm == cisco call manager |
| 20:54.17 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 20:54.27 | Zodiacal | if * could do that, i would be a happy camper |
| 20:55.05 | file | Zodiacal: it's really a key system-like thing... where you have lines that go to buttons on a phone, people can pick up and use lines and see the status of them on other phones |
| 20:55.30 | Zodiacal | file is there an open source key system? |
| 20:56.48 | file | no |
| 20:57.03 | file | it took me awhile to think of how to answer that, so I'll just say no... |
| 20:58.01 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 20:58.02 | mafkees | file ;) |
| 20:58.07 | justinu|laptop | hey file... you know much about SIP-B? |
| 20:58.41 | file | ugh |
| 20:59.16 | mafkees | is chan_xmmp in trunk ? |
| 20:59.25 | file | if I say yes you'll ask me things, and we can't have that |
| 20:59.29 | justinu|laptop | lol |
| 20:59.50 | justinu|laptop | is it a big secret or something? |
| 20:59.56 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 21:00.00 | file | no, it's just evil |
| 21:00.00 | *** join/#asterisk heison (n=heison@209.167.5.1) |
| 21:00.00 | *** join/#asterisk angler_ (n=angler@gateway.digium.com) |
| 21:00.12 | file | angler_ knows ALL about it though |
| 21:00.15 | file | angler_: don't you? |
| 21:00.29 | angler_ | sure do! |
| 21:00.29 | justinu|laptop | i'd really like to see some kinda technical docs on it |
| 21:01.15 | mafkees | ah, it is |
| 21:01.21 | mafkees | chan_jingle.so |
| 21:01.22 | mafkees | :) |
| 21:01.23 | mafkees | nice |
| 21:01.31 | file | justinu|laptop: let's see here... |
| 21:01.32 | mafkees | so 1.4 will have jingle support ? |
| 21:02.13 | file | justinu|laptop: now, are you talking about SIP BLA? |
| 21:02.38 | dlynes_office | FuriousGeorge: you there? |
| 21:02.51 | mafkees | file: I have a question about app_queue.so |
| 21:03.08 | dlynes_office | ~seen furiousgeorge |
| 21:03.25 | jbot | furiousgeorge is currently on #asterisk (13h 9m 10s). Has said a total of 153 messages. Is idling for 8h 39m 24s, last said: 'maybe when it goes to beta, should be any day nopw from what ive heard'. |
| 21:03.25 | dlynes_office | ~seen FuriousGeorge |
| 21:03.29 | jbot | furiousgeorge is currently on #asterisk (13h 9m 14s). Has said a total of 153 messages. Is idling for 8h 39m 28s, last said: 'maybe when it goes to beta, should be any day nopw from what ive heard'. |
| 21:03.41 | mitcheloc | ~seen myass |
| 21:03.43 | jbot | myass <~myass@203.131.110.67> was last seen on IRC in channel #asterisk, 845d 5h 9m 16s ago, saying: 'or BWHAHAHA if u r evil'. |
| 21:03.45 | dlynes_office | ah...guess he's sleeping or something |
| 21:03.45 | justinu|laptop | file: amonst other things, yeah |
| 21:03.52 | file | justinu|laptop: http://www.ietf.org/internet-drafts/draft-anil-sipping-bla-03.txt that's the draft for BLA |
| 21:03.58 | mafkees | file: is it on purpose that the r option to Queue kills the announcements ? |
| 21:04.12 | justinu|laptop | it's the broadcom switch that implemented the proprietary SIP-B stuff, i think |
| 21:04.33 | dlynes_office | justinu|laptop: yeah |
| 21:04.50 | dlynes_office | justinu|laptop: they called theirs Shared Line Appearance |
| 21:05.06 | file | shared line appearance, bridged line appearance... it's all good |
| 21:05.09 | justinu|laptop | yeah... but the polycom phones support that stuff, but they give no details on it |
| 21:05.13 | justinu|laptop | thanks for the link file |
| 21:05.21 | dlynes_office | justinu|laptop: yeah...same for the Aastras |
| 21:05.40 | dlynes_office | justinu|laptop: the Aastras also implement Nortel's proprietary NAT navigation, too |
| 21:05.54 | file | yay proprietary! |
| 21:06.02 | dlynes_office | file: yeah, no doubt |
| 21:06.03 | mafkees | file: if not, I need to file a bug report on mantis |
| 21:06.13 | file | mafkees: I'm not an app_queue guy |
| 21:06.19 | mafkees | ok |
| 21:06.19 | file | (sadly enough) |
| 21:06.29 | dlynes_office | file: like you want to be :) |
| 21:06.32 | justinu|laptop | lol |
| 21:06.45 | mafkees | I'll check tomorrow |
| 21:06.57 | mafkees | time to join the wife in bed ;) |
| 21:07.02 | dlynes_office | tmi |
| 21:07.12 | justinu|laptop | you're just jealous :P |
| 21:07.23 | dlynes_office | well maybe |
| 21:07.25 | mafkees | lol |
| 21:07.26 | dlynes_office | is she chinese? |
| 21:07.32 | mafkees | no |
| 21:07.33 | mafkees | blonde |
| 21:07.37 | dlynes_office | then i'm not jealous :p |
| 21:07.44 | mafkees | whehehehehe |
| 21:07.46 | mafkees | latero all ! |
| 21:08.15 | justinu|laptop | dlynes been to china? |
| 21:08.36 | dlynes_office | justinu|laptop: yep |
| 21:08.36 | sevard | dlynes _is_ china. |
| 21:08.39 | sevard | fatty |
| 21:08.40 | justinu|laptop | where at? |
| 21:08.47 | dlynes_office | Beijing-shi |
| 21:08.51 | justinu|laptop | i was disapointed with the chicks in HK |
| 21:09.00 | justinu|laptop | but the ones in guangzhou were hawt |
| 21:09.01 | dlynes_office | justinu|laptop: too materialistic? |
| 21:09.08 | sevard | amsterdam girls |
| 21:09.11 | sevard | where's at |
| 21:09.11 | dlynes_office | justinu|laptop: too much makeup? |
| 21:09.27 | justinu|laptop | i dunno, i just dind't see many good lookers |
| 21:09.41 | sevard | dlynes_office: did you do this? I got an email from sherman today |
| 21:09.47 | dlynes_office | I'm not terribly impressed with HK'ers or girls from Guangdong, either |
| 21:09.51 | sevard | he told me to email Yu Chan for authorization |
| 21:09.57 | dlynes_office | sevard: ah...cool |
| 21:10.04 | sevard | that's your doing? |
| 21:10.07 | dlynes_office | sevard: must've been my email yesterday that started it :p |
| 21:10.12 | dlynes_office | sevard: he never emailed me |
| 21:10.13 | podzap | ding da da da ding ding ding ding dong |
| 21:10.13 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 21:10.14 | sevard | son a bitch |
| 21:10.14 | justinu|laptop | i wanna go to shanghai |
| 21:10.18 | sevard | why will they listen to you |
| 21:10.22 | podzap | oriental classic by la choy |
| 21:10.24 | sevard | because you're clearly canadian |
| 21:10.27 | dlynes_office | justinu|laptop: Beijing's the bomb |
| 21:10.43 | dlynes_office | justinu|laptop: most girls there are pretty common looking though |
| 21:10.44 | justinu|laptop | i lost a lot of respect for beijing when they put a starbucks in the forbidden city |
| 21:10.56 | dlynes_office | justinu|laptop: the good looking girls are all in Hangzhou and Suzhou |
| 21:11.18 | dlynes_office | justinu|laptop: they did? |
| 21:11.21 | justinu|laptop | yeah |
| 21:11.27 | dlynes_office | justinu|laptop: must've been after I was there |
| 21:11.31 | sevard | alright, so you looked at girls |
| 21:11.33 | dlynes_office | justinu|laptop: i don't remember seeing one there |
| 21:11.33 | sevard | pick any up? |
| 21:11.43 | sevard | failures. |
| 21:11.59 | dlynes_office | sevard: i could've had one easily...but i would have had to wait for her to graduate first :p |
| 21:12.09 | justinu|laptop | http://archives.cnn.com/2000/FOOD/news/12/11/china.starbucks.reut/ |
| 21:12.13 | dlynes_office | sevard: she just started at Tsinghua University |
| 21:12.15 | sevard | dlynes_office: next time try one loder than 12 |
| 21:12.18 | sevard | older* |
| 21:12.37 | justinu|laptop | sevard: nothing wrong with 12 year old girls in amsterdam |
| 21:12.49 | dlynes_office | sevard: ok, fine...I won't try picking up your sister anymore :(( |
| 21:12.59 | sevard | my sister is 25 :) |
| 21:13.07 | justinu|laptop | bring it on |
| 21:13.11 | sevard | justinu|laptop: that's wrong all over the world man |
| 21:13.25 | dlynes_office | sevard: and here I thought you were like 16... |
| 21:13.27 | sevard | justinu|laptop: i draw the line there |
| 21:13.39 | sevard | dlynes_office: what? you're the one saying "like" |
| 21:13.57 | justinu|laptop | totally |
| 21:14.00 | sevard | OMFG I THOUGH U WER LIK 16!~1~ |
| 21:14.16 | justinu|laptop | a/s/l? |
| 21:14.17 | dlynes_office | hahahahaha |
| 21:14.29 | sevard | 16/yesprz/n e wer u want bebe |
| 21:14.29 | dlynes_office | aids/sex/life? |
| 21:15.00 | file | back on topic! |
| 21:15.06 | sevard | this DJ jsut said "we just got the word gonads, who likes grimy? if you like the word grimey put it up in the chatroom" |
| 21:15.10 | justinu|laptop | ok, what's the topic? |
| 21:15.15 | sevard | i think DJs have run out of shit to say |
| 21:15.18 | file | How to become a VoIP provider. |
| 21:15.28 | sevard | file: Lesson 1, begin. |
| 21:15.54 | justinu|laptop | buy SS7 f-link |
| 21:16.04 | sevard | $$7 you mean |
| 21:16.06 | justinu|laptop | buy SS7 capable voip gateway |
| 21:16.33 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:16.35 | eKo1 | What is f-link? |
| 21:16.39 | dlynes_office | wtf? |
| 21:16.41 | *** part/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 21:16.44 | file | sort of like a g-link |
| 21:16.46 | justinu|laptop | it's like a ghetto SS7 link |
| 21:16.50 | *** mode/#asterisk [+o file] by ChanServ |
| 21:16.54 | dlynes_office | they had a mcdonald's at tiananmen, and a kfc at beihai, too? |
| 21:16.58 | justinu|laptop | sorta like having a NATd IP network |
| 21:17.06 | eKo1 | What SIP message does the caller get back if they surpass their call-limit? |
| 21:17.06 | dlynes_office | none of that crap was there when i was there |
| 21:17.23 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:17.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:17.36 | file | russellb: hug hug! |
| 21:17.44 | russellb | file: better? |
| 21:17.51 | file | russellb: meh |
| 21:18.24 | eKo1 | Anyone? |
| 21:18.35 | justinu|laptop | declined? |
| 21:19.12 | justinu|laptop | i'd have to go browse chan_sip.c to know the answer, i'm sure you're capable of that |
| 21:19.27 | robin_sz | mmmm ... beer |
| 21:20.59 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 21:21.24 | eBody | what is a HUD?? |
| 21:21.44 | justinu|laptop | heads up display |
| 21:23.23 | *** join/#asterisk mtaht4 (n=m@209.213.206.38) |
| 21:23.29 | eBody | how's that differ from a SIP softphone/client |
| 21:23.40 | lunk | is there any way to limit the usage of a trunk to a certain number of minutes? |
| 21:23.58 | lunk | like, i want to use the free 100 minutes i get with broadvoice before dipping into voipjet minutes |
| 21:24.27 | dlynes_office | lunk: Check the documentation for the dial application |
| 21:24.41 | dlynes_office | lunk: check the D() parameter, specifically |
| 21:24.49 | russellb | well, that's only going to work for the first call :) |
| 21:24.59 | justinu|laptop | yeah, he wants to track usage, then change his dialplan or something |
| 21:25.10 | lunk | yea |
| 21:25.21 | dlynes_office | well, obviously he would have to use it conjunction with a database |
| 21:25.31 | lunk | ooooo |
| 21:25.36 | russellb | you could do that with the time limit option to Dial() along with something in astdb |
| 21:25.39 | justinu|laptop | or a script to parse the CDR file, and modify the dialplan |
| 21:25.39 | lunk | dlynes_office: brilliant! |
| 21:25.41 | russellb | using the billsec variable from the cdr |
| 21:25.56 | sevard | hmm |
| 21:25.58 | sevard | file |
| 21:26.00 | dlynes_office | lunk: but afaik, there's nothing precanned for it |
| 21:26.24 | lunk | right, but i could issue a direct mysql query |
| 21:26.30 | sevard | free skype out to usa and canada -- does anyone know if in the EULA they say you can't resell it? |
| 21:26.33 | lunk | get the sum of minutes used, and then gotoif accordingly |
| 21:26.43 | dlynes_office | lunk: yeah...use an agi script or something...it'll probably be easier |
| 21:27.07 | eKo1 | justinu|laptop: found it. thanks |
| 21:27.08 | dlynes_office | lunk: and then based on your return code, it'll jump to the appropriate priority, and then make the call |
| 21:27.14 | lunk | ahhh |
| 21:27.17 | russellb | no priority jumping! |
| 21:27.18 | justinu|laptop | eKo1: so what's the response? |
| 21:27.26 | dlynes_office | russellb: priorityjumping=yes |
| 21:27.30 | lunk | i love it when a plan comes together |
| 21:27.33 | eKo1 | 503 service unavailable |
| 21:27.35 | justinu|laptop | ah |
| 21:27.36 | russellb | dlynes_office: no!! |
| 21:27.48 | dlynes_office | russellb: lol...so what's the 1.2 alternative, then? |
| 21:28.05 | justinu|laptop | hey, speaking of priority jumping... the P/p options to app_dial don't set DIALSTATUS right |
| 21:28.05 | russellb | i don't know ... there should be STATUS variables for every app that has priority jumping |
| 21:28.07 | justinu|laptop | it's broke |
| 21:28.19 | sevard | priority jumping is the shit |
| 21:28.33 | russellb | you guys are making me sad |
| 21:28.33 | dlynes_office | lunk: ah...yeah...set a status variable...and then gotoif based on that status variable |
| 21:28.55 | dlynes_office | lunk: otherwise, if you make the call inside the agi, it won't get properly logged to the cdr |
| 21:29.06 | dlynes_office | heh |
| 21:29.10 | dlynes_office | he left in disgust :p |
| 21:29.15 | justinu|laptop | lol |
| 21:29.34 | justinu|laptop | we just rubbed salt in the wound |
| 21:29.40 | dlynes_office | Yeah...maybe Corydon doesn't love him anymore |
| 21:29.43 | justinu|laptop | lol |
| 21:29.45 | justinu|laptop | dude |
| 21:29.54 | dlynes_office | ? |
| 21:29.57 | justinu|laptop | frightening |
| 21:30.01 | dlynes_office | heh |
| 21:30.03 | lunk | select sum(duration) from cdr where lastdata like 'SIP/BV%'; |
| 21:30.04 | lunk | \o/ |
| 21:30.19 | lunk | (need some monthly group by and junk, but details schmetails) |
| 21:30.30 | justinu|laptop | you gnna do that for every call? |
| 21:30.43 | dlynes_office | lunk: nah...i would have a variable that keeps track of how many minutes you've got left on your broadvoice account |
| 21:30.51 | jbalcomb | The US, Japan, and Iran are equally terrible soccer teams. |
| 21:30.54 | dlynes_office | lunk: decrement it every time you make a call with broadvoice |
| 21:31.06 | justinu|laptop | dlynes idea is certainly gonna scale better |
| 21:31.07 | dlynes_office | jbalcomb: eh? when did Iran become bad? |
| 21:31.12 | lunk | it's just me, so, every call already hits the database multiple times, one more won't cause any trouble |
| 21:31.44 | dlynes_office | lunk: when that variable is down to 0s, or 5s, or whatever you deem to be your minimum, start using the voipjet account |
| 21:32.05 | dlynes_office | lunk: then at the start of the next month, reset the variable to 6000 seconds again |
| 21:32.16 | lunk | how do you reset that? |
| 21:32.18 | *** join/#asterisk adorah (n=Asterjet@87.69.72.228) |
| 21:32.25 | dlynes_office | lunk: using your dialplan and/or agi |
| 21:32.30 | sevard | dlynes_office: I just got 9 pages of forms emailed to me |
| 21:32.35 | sevard | dlynes_office: <3 |
| 21:32.36 | dlynes_office | sevard: heh |
| 21:33.08 | justinu|laptop | did you fill out your 27B stroke 6? |
| 21:33.22 | dlynes_office | 9 pages of forms for a lousy login id and password |
| 21:33.26 | dlynes_office | htat's pretty bad :) |
| 21:33.33 | justinu|laptop | from a movie |
| 21:33.35 | justinu|laptop | brazil |
| 21:33.37 | dlynes_office | sevard: it was an irs inference, I believe |
| 21:33.41 | dlynes_office | or whatever :0 |
| 21:33.47 | justinu|laptop | central services... you do the work, we do the pleasure! |
| 21:34.01 | mountainm2k | To anybody who cares -- I had asked about a second dialtone after "9"... The Polycom will do it, if you program the dialplan correctly... I can provide info if anybody cares... |
| 21:34.10 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:34.12 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:34.16 | dlynes_office | mountainm2k: it's called Disa() |
| 21:34.20 | mountainm2k | (the dialplan on the PHONE, not *) |
| 21:34.24 | dlynes_office | mountainm2k: oh |
| 21:34.26 | Dr-Linux | hi all |
| 21:34.27 | mountainm2k | no it's not... |
| 21:34.29 | sevard | yeah, these bastards are almost worse than the vogans |
| 21:34.33 | mountainm2k | I'm talking about a user's IP phone |
| 21:34.42 | mountainm2k | pick up, dial 9, expect another dial tone, don't get one... |
| 21:34.52 | mountainm2k | the polycom will do it, if you provision its internal dialplan... |
| 21:35.04 | justinu|laptop | mountainm2k: how can I make my polycoms play a reorder (fast busy) after far end hangup? |
| 21:35.07 | dlynes_office | why would you want to, though? |
| 21:35.11 | sevard | at least they won't demolish our earth for a hyperspace bypass |
| 21:35.13 | mountainm2k | pretty cool actually -- tell it the basics, and it will dial instantly without the <SEND> key, too |
| 21:35.15 | justinu|laptop | because his lusers complained |
| 21:35.21 | justinu|laptop | just like mine |
| 21:35.38 | justinu|laptop | people fear change |
| 21:35.42 | CunningPike | Isn't there a way to have a blind transfer show the CID of the original caller instead of the transferror? |
| 21:35.51 | mountainm2k | Dunno, I havn't made it that far yet... |
| 21:35.55 | justinu|laptop | CunningPike: i did some work on that |
| 21:35.56 | sevard | transferrorerorerinator? |
| 21:35.58 | dlynes_office | justinu|laptop: seems kinda stupid...the only use i've found for disa is for being able to dial into a phone number on the pri, and then make calls out from there |
| 21:36.00 | justinu|laptop | wasn't easy |
| 21:36.12 | justinu|laptop | asterisk core doesn't support it |
| 21:36.15 | CunningPike | justinu|laptop: What did you end up doing? |
| 21:36.15 | Dr-Linux | someone told me that normaly a DID costs $11/month , so what will be estimated cost if i buy it with 12 channels? |
| 21:36.16 | mountainm2k | I'd like to make it complete the supervised transfer on hangup, too, but it doesn't seem to do that, it leaves the call on hold... |
| 21:36.17 | dlynes_office | CunningPike: that's the way it normally works |
| 21:36.37 | dlynes_office | CunningPike: asterisk should transfer the caller id normally |
| 21:36.45 | dlynes_office | CunningPike: unless you're doing something strange |
| 21:36.49 | justinu|laptop | CunningPike: i patched the code to implement Remote-party-ID tags in the 180/183 sip responses |
| 21:36.56 | sevard | dlynes_office: you're the one rewriting voicemail, right? |
| 21:36.59 | CunningPike | justinu|laptop: Eww |
| 21:37.01 | dlynes_office | sevard: yeah |
| 21:37.02 | justinu|laptop | yep |
| 21:37.11 | CunningPike | OK - thanks |
| 21:37.21 | dlynes_office | CunningPike: are you using agents and queues? |
| 21:37.31 | Spy000007 | Dr-Linux: we told you how to research prices, it's time to contact the vendors you're interested in, you won't get any more useful info here |
| 21:37.34 | CunningPike | dlynes_office: Just queues and members |
| 21:37.45 | dlynes_office | CunningPike: ah...that's why you're having problems iwth that then |
| 21:37.51 | CunningPike | dlynes_office: We don't need agents |
| 21:37.53 | sevard | dlynes_office: a feature request hit me from like 10 people at the same time who would be donverting |
| 21:37.55 | dlynes_office | CunningPike: i don't use such things, and transfers work just fine |
| 21:37.56 | CunningPike | dlynes_office: Huh? |
| 21:38.15 | sevard | dlynes_office: dialing into your voicemail and pressing a button to get an outside line |
| 21:38.24 | dlynes_office | CunningPike: i don't use queues, members, or agents, and transfers relay the original caller id just fine |
| 21:38.27 | *** part/#asterisk mtaht4 (n=m@209.213.206.38) |
| 21:38.28 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 21:38.30 | sevard | so that you can get calls billed to your buisness line |
| 21:38.39 | Dr-Linux | Spy000007: many peoples are using and doing SIP, so i thought maybe someone help me. |
| 21:38.42 | CunningPike | dlynes_office: How are you transfering? |
| 21:38.54 | dlynes_office | sevard: it already has that feature |
| 21:39.06 | dlynes_office | CunningPike: hitting the transfer button on the Aastra phone |
| 21:39.06 | sevard | dlynes_office: really |
| 21:39.11 | dlynes_office | sevard: yeah |
| 21:39.18 | sevard | dlynes_office: bull |
| 21:39.25 | dlynes_office | sevard: read your sample config file |
| 21:39.26 | mountainm2k | OK, here's a question: incoming call to ZAP, ends up at a SIP phone, and the incoming call detail says "asterisk"... There is no caller-ID -- can I make it say "No Caller ID" or something??? |
| 21:39.30 | dlynes_office | sevard: it's well documented |
| 21:39.32 | CunningPike | dlynes_office: Interesting - these are just normal calls |
| 21:39.38 | sevard | i did not find said feature |
| 21:39.42 | CunningPike | mountainm2k: Yes - in your dialplan |
| 21:39.43 | justinu|laptop | mountainm2k: yeah... there's a way to change that |
| 21:39.44 | sevard | forward me to the documentation :/ |
| 21:40.01 | dlynes_office | CunningPike: or at least I believe that's the case...none of my customers have complained about it, and I would think that's a pretty basic feature :0 |
| 21:40.02 | jbalcomb | dlynes_office: Iran is W0-L2-D1, same as Japan and US.. unless I jumped lines while reading. |
| 21:40.08 | CunningPike | dlynes_office: Must be the way that the Polycoms do transfers |
| 21:40.31 | CunningPike | dlynes_office: If my boss gets a call and blind transfers to me, I get her CID, not the original caller. |
| 21:40.40 | CunningPike | dlynes_office: Nothing to do with queues |
| 21:40.42 | dlynes_office | CunningPike: yeah, i understand the issue |
| 21:40.55 | jbalcomb | CunningPike there is a option in the Dial() function to change that behavior |
| 21:41.08 | sevard | dlynes_office: is that in advanced options? because I don't hear it. maybe if you told me what it's called |
| 21:41.14 | justinu|laptop | oh, that's kinda different from what I was doing |
| 21:41.20 | justinu|laptop | I was implementing connected party ID |
| 21:41.38 | ph|ber | anyone got a few to help out on connecting 2 * boxes via iax? |
| 21:41.45 | dlynes_office | woah |
| 21:41.49 | justinu|laptop | if I call user B, and his phone is forwarded to C, I want to see that I'm ringing/talking with C |
| 21:41.57 | dlynes_office | there's a volume control for music on hold that never gets installed by default |
| 21:42.06 | jbalcomb | ph|ber have you read the 'connecting to asterisk servers' document on the wiki? |
| 21:42.19 | sevard | dlynes_office: oh? neat. |
| 21:42.19 | jbalcomb | s/to/two/ |
| 21:42.19 | ph|ber | yea |
| 21:42.25 | dlynes_office | sevard: muted.conf.sample in your asterisk source root directory |
| 21:42.26 | ph|ber | i have em connected.. but im getting this |
| 21:42.30 | jbalcomb | ph|ber what seems to be the trouble? |
| 21:42.33 | CunningPike | justinu|laptop: Ah - we'd just be happy to get the CID of person B when we called them |
| 21:42.41 | ph|ber | Type: IAX Subclass: REJECT |
| 21:42.45 | ph|ber | <PROTECTED> |
| 21:42.56 | ph|ber | and i have tried with a register script and without |
| 21:42.59 | justinu|laptop | respect my authoriTAY! |
| 21:43.19 | sevard | dlynes_office: wtf does it do it by the manager interface? |
| 21:43.55 | dlynes_office | sevard: dialout= option is for the dialing out from the advanced menu |
| 21:44.16 | sevard | sevard: can you include that for individual clients? |
| 21:44.26 | sevard | s/clients/mailboxes |
| 21:44.46 | tzafrir_laptop | Off-topic: anybody here with a @aol.com email? I need to test that our mail server can send messages there. Please /msg me if you can help me test. Thanks |
| 21:44.52 | dlynes_office | sevard: talking to yourself again, I see? |
| 21:44.57 | sevard | hahaha |
| 21:45.09 | sevard | dlynes_office: can you include that for individual voice mail boxs? |
| 21:45.10 | sevard | :) |
| 21:45.13 | jbalcomb | ph|ber http://www.google.com/search?hl=en&q=Type%3A+IAX+Subclass%3A+REJECT+CAUSE+%3A+No+authority+found&btnG=Google+Search |
| 21:45.25 | sevard | or only each context |
| 21:45.33 | dlynes_office | sevard: yes...individual mailbox basis |
| 21:46.01 | sevard | i thought you said option=dialout |
| 21:46.12 | sevard | that would only work for a whole voicemail context |
| 21:46.12 | dlynes_office | sevard: yeah...muted.conf is for the manager interface |
| 21:46.25 | *** join/#asterisk MoutaPT (n=MoutaPT@a83-132-239-109.cpe.netcabo.pt) |
| 21:46.32 | ph|ber | i have looked at like 10 of those |
| 21:46.46 | dlynes_office | ...|dialout=blahblah|option=blahblah|option=blahblah|option=blahblah|... |
| 21:47.01 | sevard | what! |
| 21:47.03 | CunningPike | jbalcomb: Aha! Option 'o' for Dial - thanks! |
| 21:47.19 | jbalcomb | CunningPike: np. glad it worked out. |
| 21:47.27 | dlynes_office | CunningPike: oh heh...maybe i'm already using that :) |
| 21:47.28 | jbalcomb | ph|ber and? |
| 21:47.35 | ph|ber | i have not found the problem. |
| 21:48.00 | jbalcomb | ph|ber any reason to imagine that your circumstances are special? |
| 21:48.01 | sevard | dlynes_office: the format I'm familiar with is 1338 = 1338,User,1338@localhost |
| 21:48.36 | mountainm2k | OK polycom junkies: Where's a good place to order? Looking at ~30 IP301's, ~10 501's, and a soundstation |
| 21:48.40 | CunningPike | dlynes_office: ;) |
| 21:48.53 | mountainm2k | the standard websites don't seem appropriate for that QTY |
| 21:49.36 | CunningPike | mountainm2k: Where are you? |
| 21:49.42 | mountainm2k | Colorado |
| 21:49.45 | mountainm2k | Denver |
| 21:49.50 | dlynes_office | sevard: boxnumber => password,user name,email@domain.tld,pager@domain.tld,saycid=yes|attach=yes|serveremail=voicemail@domain.tld|delete=1|dialout=vm-outbound|... |
| 21:50.39 | sevard | you're really going to have to show me that documentation, i have no idea what delete=1 is for |
| 21:51.01 | dlynes_office | sevard: it means delete the voicemail after it's been emailed as an attachment to email@domain.tld |
| 21:51.03 | jbalcomb | ph|ber method 1, 2, or 3? |
| 21:51.33 | ph|ber | jbalcomb: huh? |
| 21:51.36 | dlynes_office | sevard: do you not have the sample config files for asterisk? |
| 21:51.39 | sevard | dlynes_office: and I'm assuming vm-outbound is a context, hmm |
| 21:51.45 | jbalcomb | ph|ber http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
| 21:51.47 | sevard | dlynes_office: sort of :) |
| 21:51.50 | sevard | I play by ear brotha |
| 21:51.51 | dlynes_office | sevard: correct...in your extensions.conf file |
| 21:52.01 | ph|ber | method 1 |
| 21:52.18 | sevard | dlynes_office: I have a very non comformist setup |
| 21:52.32 | sevard | a new sort of way of organizing data within the configurations |
| 21:52.38 | ph|ber | i have also tried peer peer |
| 21:52.40 | dlynes_office | sevard: so download the source code then |
| 21:52.45 | jbalcomb | ph|ber pastebin your iax.conf and the relevant section of your extensions.conf |
| 21:52.46 | dlynes_office | sevard: so you've got a copy of the sample config files |
| 21:52.47 | sevard | the logic isn't finalized but i'll show you sometime :) |
| 21:53.40 | sevard | dlynes_office: izzight |
| 21:54.03 | *** mode/#asterisk [-o file] by ChanServ |
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| 21:54.37 | sevard | hahaha |
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| 21:54.44 | Dr-Linux | dlynes_office: i got a customer, they wants me to configure their new call center small setup. |
| 21:55.06 | dlynes_office | Dr-Linux: ? |
| 21:55.22 | Dr-Linux | dlynes_office: they will use outbound/inbound |
| 21:55.23 | sevard | hey, here's a concept, say you have a PRI and have groups of people seperated contexualy.. say you want to give a context of 10 extensions a max of 2 outbound/inbound (the sum) concurrent calls on your PRI |
| 21:55.30 | sevard | dlynes now that's a brain teaser :) |
| 21:55.45 | dlynes_office | sevard: not especially |
| 21:55.54 | sevard | it sounds like a brain teaser :/ |
| 21:56.01 | dlynes_office | sevard: even Dr-Linux could figure that out |
| 21:56.02 | Dr-Linux | dlynes_office: i'll do all for them, but i have no idea about DID's providers etc |
| 21:56.08 | sevard | hahaha |
| 21:56.18 | ph|ber | http://pastebin.com/726328 |
| 21:56.22 | CunningPike | sevard: Outbound is easy - inbound requires the cooperation of your telco |
| 21:56.39 | sevard | so, how would you do it ? |
| 21:56.49 | dlynes_office | CunningPike: well, i'd just send anything over their limit directly into voicemail |
| 21:57.08 | CunningPike | dlynes_office: It would still tie up the channel though |
| 21:57.18 | dlynes_office | CunningPike: ok, so hangup on them then |
| 21:57.24 | CunningPike | sevard: Make groups in your zapata.conf |
| 21:57.39 | dlynes_office | CunningPike: much better solution :) |
| 21:57.39 | CunningPike | dlynes_office: Now _that's_ customer service ;) |
| 21:57.49 | jbalcomb | I need to collect all the MAC Addresses of my phones (10.0.X.X) and populate a MySQL DB. What is good and proper way to do this? |
| 21:57.59 | CunningPike | jbalcomb: arpwatch |
| 21:58.07 | dlynes_office | CunningPike: i figure a ten second voicemail is better than passing the call to them |
| 21:58.21 | sevard | dlynes_office: hrm, that's not working |
| 21:58.33 | vader-- | can you playback multiple waves on one line? |
| 21:58.34 | dlynes_office | sevard: what isnt'? |
| 21:58.36 | CunningPike | jbalcomb: Or netdisco |
| 21:58.38 | sevard | your weiner |
| 21:58.42 | sevard | tee htee hee |
| 21:59.03 | dlynes_office | sevard: ummmm....how old are you again? |
| 21:59.12 | justinu | FYI: don't do any business with a company called FGP.com in south carolina |
| 21:59.14 | sevard | CunningPike: what if one doesn't want to dedicate 2 slots on the span? |
| 21:59.20 | jbalcomb | CunningPike Is netdisco Linux based? |
| 21:59.20 | justinu | they stiffed me on 1200 bucks of consulting services |
| 21:59.44 | CunningPike | jbalcomb: Yes - it's a little fiddly to set up, but it's an invaluable tool for all sorts of network inventory |
| 21:59.47 | *** join/#asterisk JordanN (n=jmnash@pool-71-246-132-102.aubnin.fios.verizon.net) |
| 21:59.53 | sevard | CunningPike: For the sake of over selling that is, if you dedicate 2 slots ont he span you can't over sell because those slots will -always- be in use |
| 22:00.33 | jbalcomb | CunningPike sounds good. i'll check it out tomorrow. |
| 22:00.38 | CunningPike | sevard: I have no idea what you're talking about - you want a group of phones to use only 2 spans? |
| 22:00.47 | CunningPike | s/spans/channels/ |
| 22:00.58 | sevard | CunningPike: right, but not dedicated channels |
| 22:01.26 | dlynes_office | sevard: astdb is your friend |
| 22:01.30 | sevard | ? |
| 22:01.50 | CunningPike | sevard: Ya - it'll need some kind of db tracking mechanism |
| 22:01.53 | dlynes_office | increment/decrement call counters |
| 22:02.02 | CunningPike | sevard: Easy to get out of sync |
| 22:02.03 | dlynes_office | and check those call counters to see if you're able to make a call |
| 22:02.42 | sevard | that sounds like something Dr-Linux can't do :( |
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| 22:03.43 | dlynes_office | Dr-Linux: subscribe to asterisk-biz, and ask around for who will provide you dids |
| 22:03.55 | *** join/#asterisk Samoied (n=Samoied@201.22.205.152.adsl.gvt.net.br) |
| 22:04.28 | Dr-Linux | dlynes_office: actually there are many, but i need to know something |
| 22:05.51 | sevard | dlynes_office: 1337 = 1337,User,1337@localhost|dialout=local |
| 22:05.59 | sevard | dlynes_office: per what you were saying that ought to work |
| 22:12.38 | dlynes_office | sevard: doulbe check what I said |
| 22:12.41 | dlynes_office | sevard: that's not what I had |
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| 22:14.54 | mountainm2k | ~jbot |
| 22:14.55 | jbot | from memory, jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot |
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| 22:16.20 | dlynes_office | jbot hard drive failure |
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| 22:19.39 | Un1x | hello, anyone around was needing to ask a question whats the difference between |
| 22:19.46 | Un1x | the green and red fxo modules |
| 22:19.57 | Un1x | err FXS and FXO |
| 22:20.17 | dlynes_office | fxs = where analog phones and/or phone systems plug into; fxo = where phone lines plug into |
| 22:20.33 | mountainm2k | yeah, FXS is a "station", ie accepts dial tone, and FXO is "office", ie provides dial tone |
| 22:20.36 | Un1x | so wich one do i need, if i want to hook up a regular home phone to it? |
| 22:20.45 | dlynes_office | Un1x: one of each |
| 22:20.48 | mountainm2k | phone lines plug into FXS, phones plug into FXO |
| 22:21.01 | dlynes_office | Un1x: one fxo for the phone line, one fxs for your phone |
| 22:21.01 | mountainm2k | you need an FXO to plug a regular phone into it |
| 22:21.17 | mountainm2k | Whoops, am I backasswords? |
| 22:21.22 | Un1x | but i only need FXS if im providing services for voip to pstn right? |
| 22:21.25 | dlynes_office | mountainm2k: phone lines plug into fxo and use fxs signalling |
| 22:21.27 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 22:21.35 | dlynes_office | Un1x: no...mountainm2k's confusing you |
| 22:21.36 | mountainm2k | I'm back-asswords.. |
| 22:21.38 | dlynes_office | Un1x: you need fxo |
| 22:21.51 | mountainm2k | I'm wrong, listen to dlynes_office |
| 22:21.55 | Un1x | so i need all FXO right if i'm using someonee else as my voip provider? |
| 22:21.59 | mountainm2k | <slinks away slowly> |
| 22:22.10 | dlynes_office | Un1x: correct, but you still need one fxs port for your analog phones |
| 22:22.21 | dlynes_office | Un1x: unless you're going to use a voip phone |
| 22:22.44 | Un1x | so FXO is for voip phones, and FXS is for phones like the ones in regular homes? |
| 22:22.50 | dlynes_office | no |
| 22:22.58 | Un1x | sorry about that but i'm kinda confused |
| 22:23.01 | dlynes_office | fxo is for analog phone lines (o = office) |
| 22:23.08 | dlynes_office | fxs is for extensions (s = station) |
| 22:23.38 | Un1x | ahh i see, like if i wanna be able to dail a extension i.e speak to someone else with a phone in there room kinda thing right? |
| 22:23.43 | Un1x | like those interoffice phones |
| 22:24.00 | dlynes_office | but if all your phones will be voip phones (you won't be using your old analog cordless or analog wall phones), you only need the fxo ports for your phone lines; you don't need fxs ports |
| 22:24.37 | Un1x | noo see youre geting me wrong or im explaining wrong, i will not be using those voip phones, i will be using a phone like one of those from home... |
| 22:24.41 | dlynes_office | Un1x: that's all dependent on how you set up asterisk |
| 22:24.43 | Un1x | like one of those cordless panasonic ones |
| 22:24.45 | dlynes_office | Un1x: yeah |
| 22:24.59 | Un1x | so i need all FXO or FXS? |
| 22:25.03 | dlynes_office | Un1x: so you need at least one fxs port and one fxo port for every analog phone line you want to have coming in |
| 22:25.23 | dlynes_office | Un1x: so if you have 2 lines and 1 cordless phone, you'd want 2 fxo ports, and 1 fxs port |
| 22:25.37 | Un1x | i see so the FXS port is for the phone |
| 22:25.41 | dlynes_office | exactly |
| 22:25.43 | dlynes_office | s = station |
| 22:25.47 | dlynes_office | o = central office |
| 22:26.11 | Un1x | i see but, if im not going to have a standard PSTN line into my asterisk box, then i dont need a FXO module correct? |
| 22:26.19 | Un1x | as if im using a VOIP service from one of those companys |
| 22:26.20 | dlynes_office | Un1x: if that's the case then, yes |
| 22:26.28 | dlynes_office | Un1x: you'd only need an fxs port then |
| 22:26.49 | Un1x | thanks Dlynes_office |
| 22:26.50 | Un1x | :p |
| 22:26.55 | dlynes_office | Un1x: so i would just go with a pap2-na, or a sipura 2002, or something |
| 22:27.06 | dlynes_office | Un1x: cheaper than buying a telephony card with an fxs port on it |
| 22:27.42 | Un1x | dylnes, i wanna be able to plug my cordless phone in |
| 22:27.46 | Un1x | dont wanna be stuck at my windows pc |
| 22:27.52 | dlynes_office | yeah, and? |
| 22:28.14 | Un1x | so isn't SIP phones... O/S based? |
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| 22:28.14 | dlynes_office | no |
| 22:28.14 | dlynes_office | those are skype phones and soft phones |
| 22:28.18 | Un1x | ahh i see |
| 22:28.22 | dlynes_office | SIP is a protocol that many of the softphones use |
| 22:28.31 | dlynes_office | but most hard ip phones also use SIP |
| 22:28.41 | dlynes_office | as do most voip long distance companies |
| 22:28.55 | Un1x | i see |
| 22:28.59 | dlynes_office | Session Initiation Protocol |
| 22:29.08 | dlynes_office | it was developed by Cisco many moons ago |
| 22:29.21 | dlynes_office | and later standardized by the IETF |
| 22:29.26 | Un1x | ahh |
| 22:29.42 | dlynes_office | and cisco has the worst implementation, for whatever reason :) |
| 22:30.18 | Un1x | dlunes_office: can i get one of these with, all Green modules Aka. FXS http://www.digium.com/en/products/hardware/tdm400p.php |
| 22:30.49 | dlynes_office | Un1x: yes |
| 22:31.02 | dlynes_office | Un1x: but why would you want to, if you're only using one cordless phone? |
| 22:31.59 | Un1x | well i'm going to use 2 cordless phones the voip provider i'm getting allows 2 numbers, one us and one canada or both us or both canada |
| 22:31.59 | Un1x | so i'm going to take one us and one canad,a and thus 2 different phones ;P |
| 22:31.59 | dlynes_office | ok, and? |
| 22:31.59 | Un1x | so i should just leave the 2 green and 2 red? |
| 22:32.04 | Un1x | no need for asking for all green |
| 22:32.04 | dlynes_office | why do you need two separate cordless phones? |
| 22:32.14 | Un1x | can i use one phone with 2 phone numbers? |
| 22:32.20 | dlynes_office | Un1x: of course...why not? |
| 22:32.57 | dlynes_office | sipura 2002/pap2-na comes with 2 fxs ports, anyways |
| 22:33.15 | CunningPike | Way easier than setting up a TDM400 |
| 22:33.17 | Un1x | you got a link? |
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| 22:33.33 | Un1x | CunningPike what's so difficult in setting up a TDM400 |
| 22:34.16 | justinu|laptop | somone has to remember that when he comes back asking for help |
| 22:34.27 | CunningPike | Un1x: It can be tricky getting the interrupts etc. sorted out |
| 22:34.46 | Un1x | i see |
| 22:35.18 | Strom_C | most of the time, the TDM400 installation is a no-brainer |
| 22:35.24 | Strom_C | install, compile, run |
| 22:37.35 | Un1x | i see |
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| 22:37.54 | Un1x | wel i like th tdm400 kit i'll get it see how everything goes ;P |
| 22:37.57 | Un1x | this is going to be fun |
| 22:38.15 | justinu|laptop | true dat |
| 22:38.17 | Un1x | now i gotta find voip providers in Norway and stuff |
| 22:38.27 | Un1x | to pump the data into PSTN lines i want a norway number as well :P |
| 22:38.41 | Un1x | http://www.canadianvoipstore.com/product_info.php?products_id=1193 |
| 22:38.46 | Un1x | i like this phone tho very nice :P |
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| 22:41.08 | dlynes_office | Un1x: norway? i thought you were in Ontario? |
| 22:41.23 | Un1x | yes i am, but i have family in norway... |
| 22:41.28 | dlynes_office | ah |
| 22:41.43 | Un1x | if you get a voip provider in countrys u intend to call |
| 22:41.48 | dlynes_office | Un1x: http://www.sipura.com/ |
| 22:41.50 | Un1x | and can find free incoming or free outgoing |
| 22:41.56 | Un1x | u basicly got free long distance :P |
| 22:42.23 | Un1x | thanks you dlunes |
| 22:42.25 | Un1x | err |
| 22:42.27 | Un1x | dlynes |
| 22:42.47 | dlynes_office | Un1x: whereabouts are you in Ontario? |
| 22:43.50 | Un1x | Toronto |
| 22:43.54 | dlynes_office | ah |
| 22:44.02 | dlynes_office | was thinking you might be in Thunder Bay |
| 22:44.11 | dlynes_office | There's a lot of Scandinavians there |
| 22:44.18 | dlynes_office | Especially Finlanders |
| 22:44.43 | Un1x | heh |
| 22:44.57 | Idle | holy crap... LUG's are the best place to ask things... I asked for some free/cheap machine (old), and I had 3 responses within, oh, an hour |
| 22:45.05 | Qwell[] | Idle: nice |
| 22:45.05 | Un1x | im not scandanavian tho, my cousins moved there after |
| 22:45.06 | dlynes_office | a lot of those Finnish girls are pretty cute :0 |
| 22:45.10 | dlynes_office | ah |
| 22:45.10 | Un1x | heh |
| 22:45.16 | Qwell[] | Idle: What're they offering? |
| 22:45.41 | dlynes_office | Idle: in edmonchuck? |
| 22:45.48 | *** part/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 22:45.53 | Un1x | does having a better server to run ur PBX on help at all |
| 22:45.55 | Un1x | or even slightly |
| 22:46.08 | dlynes_office | Un1x: for the amount of lines you're planning on running? |
| 22:46.11 | dlynes_office | Un1x: not really |
| 22:46.13 | Un1x | oh |
| 22:46.20 | Un1x | how much bandwidth does one line use |
| 22:46.26 | Un1x | coz my upload one on line is max 100KB |
| 22:46.27 | Idle | Qwell[]: 1 has a 233 K6, not PCI 2.2 tho :( |
| 22:46.34 | Qwell[] | lame |
| 22:46.37 | Idle | one with a k6 450 or something |
| 22:46.40 | dlynes_office | Un1x: ulaw about 100K; g729 about 43K(?) |
| 22:46.46 | Un1x | hmm, do all new boards come with PCI 2.2? |
| 22:46.53 | Qwell[] | 450...k6 2? |
| 22:46.53 | Idle | Pentium 2 IBM Aptiva |
| 22:46.57 | Qwell[] | Un1x: yes |
| 22:47.04 | Qwell[] | Un1x: anything within the last like...5 years :p |
| 22:47.06 | Idle | Un1x: no |
| 22:47.11 | Qwell[] | no? |
| 22:47.13 | Idle | Un1x: some only have PCI-Express |
| 22:47.15 | Idle | :( |
| 22:47.15 | Un1x | comon someone give me a answer |
| 22:47.17 | Qwell[] | oh..duh |
| 22:47.18 | Un1x | IDle |
| 22:47.22 | Un1x | some have PCI with PCI express |
| 22:47.34 | Un1x | PCI e is mainly for SLI configurations |
| 22:47.34 | Idle | Un1x: yes, but some ONLY have PCI-express |
| 22:47.35 | Qwell[] | If it has pci, it'll be pci 2.2 |
| 22:47.35 | Idle | sadly |
| 22:47.41 | Un1x | ok |
| 22:47.42 | Idle | Un1x: not at all |
| 22:47.43 | Qwell[] | Un1x: nah, it's for far more |
| 22:47.52 | Qwell[] | they're selling NICs and such now, that are pcie |
| 22:47.57 | Un1x | oh nice |
| 22:48.01 | Qwell[] | raid cards, etc |
| 22:48.01 | Idle | we have a PCI-Express server with an onboard graphics card... its scsi interface is pci-e |
| 22:48.10 | Un1x | ahh |
| 22:48.17 | Qwell[] | the *REALLY* killer ones use the 16x slot... |
| 22:48.24 | CunningPike | jbalcomb: Option 'o' works perfectly - thanks again |
| 22:49.24 | Idle | Qwell[]: the one guy with the k6 wants beer in exchange... ;) he has a 850/900 laying around hes gonna find |
| 22:49.29 | Qwell[] | heh |
| 22:49.45 | Qwell[] | is it a k6 2 though? |
| 22:49.46 | Idle | I had to maps.google.com for beer store |
| 22:49.47 | Idle | :( |
| 22:49.53 | Idle | no idea |
| 22:50.03 | Idle | the 850 was a celeron |
| 22:50.08 | Qwell[] | ugh... |
| 22:50.11 | Qwell[] | but, better than a p75 |
| 22:50.20 | Idle | yea |
| 22:50.23 | Idle | a FUCKload better |
| 22:50.24 | Idle | :D |
| 22:50.29 | Qwell[] | not really |
| 22:50.36 | Idle | yea really |
| 22:50.39 | Qwell[] | well... |
| 22:50.43 | Qwell[] | I guess p75 didn't even have mmx |
| 22:51.01 | Idle | 2 FXS and 2 FXO, 2 or 3 softphones, and a SIP phone... no problems there |
| 22:51.08 | Qwell[] | yeah... |
| 22:51.23 | Idle | where did this mysterious p75 come from? |
| 22:51.35 | Idle | I had a p120, that woulda been OK, but not great |
| 22:51.37 | Qwell[] | weren't you saying 75? |
| 22:51.41 | Qwell[] | I swore you did |
| 22:51.47 | Idle | no... :P |
| 22:51.50 | Qwell[] | somebody did |
| 22:51.58 | Idle | 32mb ram... that had A LOT of ram for a p120 |
| 22:52.09 | Idle | came with 4 iirc |
| 22:52.20 | Idle | doubled to 8, then to 16, then 32 |
| 22:52.31 | Qwell[] | 4 x 8? |
| 22:52.39 | Idle | then replaced it with a machine that had 128 |
| 22:52.39 | Qwell[] | or, rather.. |
| 22:52.45 | Qwell[] | 2 x 16 |
| 22:52.48 | Idle | 128mb on a 800mhz |
| 22:53.08 | Idle | 1x 16, 1x 8, 2x 4 |
| 22:53.16 | Qwell[] | 1x? |
| 22:53.25 | Qwell[] | wow... |
| 22:53.26 | Idle | ... |
| 22:53.34 | Idle | 1 16mb, 1 8mb, 2 4mb |
| 22:53.41 | nettie | Hi guys, I'm test asterisk-dev, what sync device should I use please? It doesnt need ohci usb anymore right? |
| 22:53.43 | Qwell[] | unpaired? on a p120? |
| 22:53.45 | *** join/#asterisk Gnuspice (n=planet@222-153-145-37.jetstream.xtra.co.nz) |
| 22:53.49 | Idle | ? |
| 22:53.53 | dlynes_office | Qwell[]: p75 mmx did |
| 22:53.55 | Qwell[] | newb |
| 22:53.57 | Idle | pairing wasn't even availible then |
| 22:54.04 | Idle | at least, not afaik |
| 22:54.10 | Qwell[] | Idle: ram needed to be paired on 386s :D |
| 22:54.12 | Idle | SIMMS man |
| 22:54.18 | dlynes_office | Qwell[]: i think you're thinking about me...I set up a P75 with asterisk yesterday |
| 22:54.20 | Qwell[] | good old 30 pin suckers |
| 22:54.23 | Qwell[] | dlynes_office: probably so |
| 22:54.23 | Idle | these were nice sim's |
| 22:54.34 | Qwell[] | Idle: simms needed to be paired.. |
| 22:54.42 | Idle | uh? no |
| 22:54.45 | Qwell[] | 72 pin.. |
| 22:54.52 | dlynes_office | Qwell[]: but yeah, there was the Pentium, then the Pentium MMX, then the Pentium Pro, Pentium II, ... |
| 22:54.56 | Idle | ever use a simm saver? good luck pairing one of those |
| 22:55.02 | Qwell[] | simm saver? |
| 22:55.11 | Idle | yea, 2 sims in 1 sim spot |
| 22:55.19 | Qwell[] | umm |
| 22:55.30 | Idle | was this little fucked up riser thing that was an oversized stick of ram with 2 simm slots in it |
| 22:55.49 | dlynes_office | Qwell[]: simms only needed to be paired, and that was only 30-pin |
| 22:56.01 | Qwell[] | 72 pin simms also needed to be paired |
| 22:56.05 | dlynes_office | Qwell[]: only certain motherboard manufacturers required 72-pin simms to be paired |
| 22:56.16 | Idle | :) |
| 22:56.17 | dlynes_office | Qwell[]: not all |
| 22:56.18 | Qwell[] | ahh, that must be it then |
| 22:56.28 | Qwell[] | s/certain/most/ ;) |
| 22:56.39 | Idle | none that I've ever worked on |
| 22:56.42 | Idle | literally, none |
| 22:56.45 | ptinsley | oh god, this dicussion is causing me to have flashbacks make it stop |
| 22:56.47 | Qwell[] | 100% of the ones I've worked on |
| 22:56.52 | Qwell[] | maybe it was a canadia thing? :P |
| 22:56.52 | Idle | I just threw ram in them, and they worked |
| 22:56.55 | dlynes_office | Idle: I'd say about 60% of them required paired simms |
| 22:57.10 | Idle | dlynes_office: I must have just been lucky, as I only worked on about 4 |
| 22:57.17 | dlynes_office | Idle: it was only the higher end boards that didn't require paired simms |
| 22:57.19 | ptinsley | dlynes_office, yep, i used to spend so much time going through boxes of ram to find ones that liked eachother |
| 22:57.32 | Idle | dlynes_office: yea, mine was top of the line when it was new |
| 22:57.35 | Un1x | so anyone know any VOIP providers in lika Norway or Asia |
| 22:57.38 | Un1x | like japan or something |
| 22:57.45 | Un1x | coz i heard it's cheap to call from japan to norway as well |
| 22:57.47 | dlynes_office | Un1x: voip's legal in japan? |
| 22:58.02 | Un1x | i do not know |
| 22:58.03 | Un1x | lol |
| 22:58.32 | dlynes_office | Un1x: i'm pretty sure it's one of those asian countries that doesn't allow it |
| 22:58.34 | Idle | if your calling home to, say, india alot, get a voip providor in india and just use that instead of calling long distance |
| 22:58.46 | CunningPike | nettie: What kernel are you running? |
| 22:58.53 | Un1x | ile |
| 22:58.54 | Un1x | idle |
| 22:58.54 | dlynes_office | Idle: again, that'd be impossible...india doesn't allow voip for sure |
| 22:58.59 | Un1x | im trying to find one in norway |
| 22:59.09 | wasim | Un1x: RoyK |
| 22:59.14 | Un1x | Royk? |
| 22:59.14 | dlynes_office | Idle: where there's voip in india, it's only because the telco hasn't found out about it, and ratted the company out |
| 22:59.16 | Idle | dlynes_office: was nothing more then an example, hypotheitcal |
| 22:59.22 | dlynes_office | Idle: ah |
| 22:59.26 | wasim | Un1x: briiz.no i think |
| 22:59.34 | Un1x | okay |
| 22:59.55 | Idle | anyhow, home time, time to see if my WIP300 came in today |
| 22:59.56 | dlynes_office | Idle: heh...if it was legal in India |
| 22:59.57 | Idle | ciao |
| 23:00.00 | knarfly | Help...I've lost the link I had for musiconhold without mpg123. Can anyone point me to it again? |
| 23:00.11 | dlynes_office | Idle: we'd be doing probably 100K minutes per month to india right now :) |
| 23:00.15 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 23:00.21 | nettie | CunningPike sorry I meant uhci usb -- I'm runign centos 4 latest kernel so 2.6.9-34.0.1 |
| 23:00.22 | Idle | ha, yea |
| 23:00.29 | Idle | all about outsourcing |
| 23:00.30 | Un1x | hm |
| 23:00.32 | Idle | anyhow, bbl |
| 23:00.36 | CunningPike | knarfly: Google voip-info.org for native MOH |
| 23:00.39 | wasim | knarfly: mode=files |
| 23:00.56 | dlynes_office | Idle: the owner of our company is indian, so he can get a lot of customers in surrey (predominantly indian populated city near Vancouver) |
| 23:01.00 | CunningPike | nettie: On 2.6 kernels, zaptel uses internal kernel timing - has for ages |
| 23:01.13 | knarfly | Thank. My moh is too choppy with mpg123 |
| 23:01.24 | nettie | CunningPike ok sounds great so, all I have to do is try builing it :) |
| 23:01.26 | dlynes_office | knarfly: do you have a timing source? |
| 23:01.44 | Un1x | has anyone herer ordered from digium before? |
| 23:01.45 | CunningPike | nettie: If you don't have a card, use ztdummy |
| 23:01.54 | CunningPike | Un1x: Not directly |
| 23:01.56 | TESTER2 | Someone get problem (or heard of) with festival (craching and misfunctionning) when zaptel is started? |
| 23:01.58 | knarfly | I don't run ztdummy if that's what you mean |
| 23:02.04 | Un1x | Cunning [ike |
| 23:02.06 | dlynes_office | knarfly: start |
| 23:02.08 | Un1x | what do you mean |
| 23:02.09 | nettie | CunningPike I would like to install it because I'm having issues with jitterbuffer I think it's frames timing problem. |
| 23:02.11 | dlynes_office | knarfly: moh needs a timing source |
| 23:02.19 | dlynes_office | knarfly: otherwise it'll be choppy |
| 23:02.24 | dlynes_office | knarfly: even if you're using native |
| 23:02.33 | nettie | CunningPike I actually have a card but I didnt touch it yet.. it's in the server but I'm planning to use it later |
| 23:02.41 | knarfly | I am still new at this. Can you explain more? |
| 23:02.59 | CunningPike | nettie: Disregard - I got two conversations jumbled |
| 23:03.19 | dlynes_office | knarfly: music on hold, conferencing, and iax2 trunking all require a timing source |
| 23:03.27 | dlynes_office | knarfly: so that the audio quality doesn't degrade |
| 23:03.38 | dlynes_office | knarfly: it's so that asterisk sequences the sound properly |
| 23:03.43 | knarfly | How do I install this? |
| 23:03.49 | Waverly360 | Goodnight guys! |
| 23:03.49 | dlynes_office | knarfly: install zaptel |
| 23:03.59 | dlynes_office | knarfly: and then make sure it loads ztdummy at startup |
| 23:04.22 | knarfly | I run FreeBSD. If I install Zaptel from the ports will that do it? |
| 23:04.31 | *** join/#asterisk philth|work (n=ceac2822@d38-179-126.home1.cgocable.net) |
| 23:04.33 | dlynes_office | knarfly: yeah |
| 23:04.51 | knarfly | Let me give the old college try and see how far I get... |
| 23:04.54 | nettie | CunningPike seems zaptel trunk doesnt build :( |
| 23:05.03 | dlynes_office | knarfly: and then you'll need to run /etc/local/rc.d/zaptel.sh |
| 23:05.14 | nettie | CunningPike maybe I have to prepare modules befor |
| 23:05.19 | dlynes_office | knarfly: i think that's where it was, anyways |
| 23:05.26 | dlynes_office | knarfly: it'll be in the same directory as asterisk.sh |
| 23:05.28 | *** join/#asterisk mtaht4 (n=m@209.213.206.38) |
| 23:05.31 | CunningPike | nettie: What error are you getting? spinlock? and yes, there are dependencies |
| 23:05.43 | Un1x | whats the difference between the soo many pbx's like asterisk and zaptel |
| 23:05.43 | dlynes_office | knarfly: after you install zaptel, you'll need to recompile and reinstall asterisk |
| 23:05.45 | knarfly | BTW - The freaking strange calls in my log were an anmoly...and you wouldn't believe what else happened today |
| 23:06.01 | Qwell[] | knarfly: You saw a flying pig? |
| 23:06.04 | Qwell[] | I'd believe it |
| 23:06.22 | dlynes_office | Un1x: zaptel is a telephony driver; asterisk is a pbx that uses that telephony driver |
| 23:06.29 | Un1x | ahh |
| 23:06.53 | dlynes_office | Un1x: erm zaptel is a collection of telephony drivers for predominantly digium hardware |
| 23:07.18 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 23:07.27 | Un1x | ahh |
| 23:07.46 | nettie | sure |
| 23:08.01 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 23:08.01 | dlynes_office | ~centosbug |
| 23:08.03 | jbot | somebody said centosbug was a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 23:08.04 | dlynes_office | ~redhatbug |
| 23:08.05 | jbot | from memory, redhatbug is is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 23:08.05 | bugz | its pouring balls here |
| 23:08.26 | Qwell[] | Strom_C: You should sue bugz for TM violations.. |
| 23:08.38 | Un1x | lol |
| 23:08.53 | CunningPike | nettie: |
| 23:08.58 | *** join/#asterisk sevard (i=kynan@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 23:09.00 | bugz | i should be the one suing |
| 23:09.02 | CunningPike | ~centosbug |
| 23:09.03 | jbot | it has been said that centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 23:09.03 | ptinsley | %{__perl} -pi -e 's/rw_lock_t/rwlock_t/' /usr/src/kernels/*/include/linux/spinlock.h |
| 23:09.05 | bugz | mines registered ;) |
| 23:09.12 | ptinsley | whoops thats rpmspeak |
| 23:09.13 | Qwell[] | bugz: "balls"? :p |
| 23:09.18 | ptinsley | perl -pi -e 's/rw_lock_t/rwlock_t/' /usr/src/kernels/*/include/linux/spinlock.h |
| 23:09.29 | sevard | ballzzzzzz |
| 23:11.38 | znoG | why do people BOTHER responding to such idiots like the guy who posted complaining about CDRTool ? |
| 23:11.48 | znoG | it's *obvious* the guy who posted has some serious issues |
| 23:12.04 | CunningPike | znoG: That's Harry - he's famous |
| 23:12.22 | *** join/#asterisk beyond (n=beyond@201-13-0-83.dsl.telesp.net.br) |
| 23:12.27 | znoG | haha |
| 23:12.32 | znoG | he must smoke a loooooot of weed |
| 23:12.40 | CunningPike | Harry needs a jbot entry |
| 23:12.44 | CunningPike | ~harry |
| 23:14.02 | znoG | harry = if ( ./configure && make && make install != success) { post-to-asterisk-list-complaining-like-a-clueless-moron }; |
| 23:14.21 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 23:14.43 | CunningPike | Thankfully, he hasn't figured out IRC yet |
| 23:14.49 | ptinsley | lol |
| 23:14.55 | znoG | i hope he does |
| 23:15.01 | znoG | so i can abuse him a littl |
| 23:15.02 | znoG | e |
| 23:15.07 | dlynes_office | ~harry |
| 23:15.09 | jbot | hmm... harry is = if ( ./configure && make && make install != success ) { post-to-asterisk-list-complaining-like-a-clueless-moron } ; |
| 23:15.23 | znoG | maybe ask him a few questions to work out what psychological disorder he has |
| 23:15.45 | dlynes_office | znoG: how about rehan walla allah? |
| 23:15.57 | znoG | ....... should I know who he is? |
| 23:16.13 | dlynes_office | znoG: the freak on asterisk-biz that keeps trying to resell everyone else's routes |
| 23:16.18 | CunningPike | That's some bad hat, Harry |
| 23:16.20 | znoG | hahaha |
| 23:16.27 | znoG | this world has it all |
| 23:16.40 | Spy000007 | haha that annoying douche |
| 23:16.42 | dlynes_office | i wonder if he's got an entry |
| 23:16.43 | dlynes_office | ~rehan |
| 23:16.49 | dlynes_office | guess not :0 |
| 23:16.55 | *** part/#asterisk mtaht4 (n=m@209.213.206.38) |
| 23:17.19 | *** join/#asterisk thock (n=thock@63.133.144.2) |
| 23:17.35 | dlynes_office | He's also the guy that runs didx.org |
| 23:17.49 | dlynes_office | ~didx |
| 23:17.55 | dlynes_office | guess that's not in there, either |
| 23:18.23 | knarfly | dlynes_office: I just tried to install zaptel and it says it installed already. But I can't find zaptel.sh. |
| 23:18.46 | thock | hey guys.. what would cause Unknown extension '1' in context 'from-pstn' requested to show up on the CLI? |
| 23:19.00 | dlynes_office | knarfly: do you know how to list what kernel modules are loaded? it's been a while since I've used freebsd |
| 23:19.06 | thock | it happens when i call into the pbx on my T1 |
| 23:19.18 | dlynes_office | thock: yeah, and you're dialing '1' |
| 23:19.30 | dlynes_office | thock: and there's no extension '1' |
| 23:19.31 | knarfly | dlynes_office: No but let me google it and see what I find? |
| 23:19.32 | CunningPike | jbot, harry is also Harry Gaillac, an irascible Frenchman who complains loudly and repeatedly about everything. Some of his posts to -users are quite entertaining. The last words are Harry's own: "Harry is not Harry Potter!" |
| 23:19.34 | jbot | okay, CunningPike |
| 23:19.50 | dlynes_office | heh |
| 23:19.52 | CunningPike | ~harry |
| 23:19.54 | jbot | it has been said that harry is = if ( ./configure && make && make install != success ) { post-to-asterisk-list-complaining-like-a-clueless-moron } ; Harry Gaillac, an irascible Frenchman who complains loudly and repeatedly about everything. Some of his posts to -users are quite entertaining. The last words are Harry's own: "Harry is not Harry Potter!" |
| 23:20.08 | thock | that's the thing though dlynes_home |
| 23:20.19 | thock | the first thing that does is go to my IVR |
| 23:20.58 | dlynes_office | yeah, and it's probably detecting you asking for extension 1, or you have a fallthrough to jump to extension 1, and you have autofallthrough=yes enabled |
| 23:21.31 | CunningPike | Douglas hasn't made an appearance here either...... |
| 23:21.39 | ptinsley | Garstan? |
| 23:21.44 | ptinsley | or stang or whatever it is |
| 23:21.49 | CunningPike | The same |
| 23:21.58 | ptinsley | hehe, he's funny |
| 23:22.14 | ptinsley | in that why isn't he on prozac kind of way |
| 23:22.46 | CunningPike | He's actually pretty smart - just kinda psychotic |
| 23:23.03 | ptinsley | ya, he has good thoughts at times |
| 23:23.15 | ptinsley | prozac should fix him right up |
| 23:23.23 | dan__t | So I want to play around with Asterisk a bit more. Can I simply have it answer a call via a standard PCI modem and do whatever with it? |
| 23:23.40 | ptinsley | dan__t, nopers |
| 23:23.46 | CunningPike | dan__t: 'Fraid not - you'll need some kind of ATA |
| 23:23.56 | dan__t | ATA? |
| 23:23.57 | ptinsley | would be nice if it was that easy |
| 23:24.00 | dan__t | Sorry, I'm still a bit new :) |
| 23:24.03 | CunningPike | ~ata |
| 23:24.04 | jbot | well, ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 23:24.15 | dan__t | Thanks. |
| 23:24.29 | CunningPike | Man, jbot is great |
| 23:24.29 | Bullseye_Network | ~corvette |
| 23:24.32 | ptinsley | or, if you have a pretty decent internet connection you could get a voip provider for pretty cheap for month just to tinker |
| 23:24.38 | Un1x | hmm |
| 23:24.41 | Bullseye_Network | Nothing? |
| 23:24.44 | Bullseye_Network | darn |
| 23:24.46 | ptinsley | for = per |
| 23:24.47 | Un1x | i wanna get one of those cisco colour screen phones |
| 23:24.48 | Un1x | :p |
| 23:24.50 | Un1x | they look nice |
| 23:24.58 | justinu | great for demos |
| 23:25.00 | ptinsley | Un1x, they look great till you price them |
| 23:25.11 | Un1x | hmm well one colourscreen i saw is 500$ |
| 23:25.20 | Un1x | i wanna get it but dont know how i will set it up |
| 23:25.24 | Un1x | direct ethernet to pbx? |
| 23:25.34 | dan__t | Excellent, tahnks. |
| 23:25.46 | dan__t | Dare I ask if there's a free or "sampler" VoIP provider for testing purposes? |
| 23:25.54 | Un1x | lmoa no |
| 23:25.56 | Un1x | *lmao |
| 23:25.57 | dan__t | I remember someone here was doing a very cheap services out of Michigan |
| 23:26.06 | ptinsley | there are some pay as you go ones |
| 23:26.08 | justinu | voipjet? |
| 23:26.09 | dlynes_office | ~fxsfxo |
| 23:26.11 | jbot | extra, extra, read all about it, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 23:26.18 | justinu | there was one that gives you 25c credit |
| 23:26.31 | dan__t | Man, I forgot what it was |
| 23:26.43 | dan__t | THey did VoIP services out of Michigan, I remember that much. |
| 23:26.56 | ptinsley | https://www.teliax.com/newaccount/?r=1&cp=default .02 cents a minute with a $10 setup fee, they are who I use for my house but I have the residential service plan |
| 23:27.15 | justinu | http://voipjet.com/ |
| 23:27.40 | Spy000007 | teliax and voipjet are excellent for testing, wouldn't use them for a live business though |
| 23:27.44 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:27.52 | justinu | agreed |
| 23:27.57 | ptinsley | ya, teliax has some challenges |
| 23:28.05 | ptinsley | things work, then they magically dont |
| 23:28.13 | Un1x | hmm |
| 23:28.22 | Un1x | can somone tell me where i can find a list of voip providers |
| 23:28.25 | Un1x | by area/region |
| 23:28.29 | Un1x | aroiund the world |
| 23:28.29 | justinu | ~wiki |
| 23:28.34 | ptinsley | support "well your account doesn't have the smiggitybop turned on" me: "it worked last week" support :"hmmmmm" |
| 23:28.36 | [hC] | hey, i have some polycom phones here, when i try to add a contact to my personal directory, it sits and sits, then reboots itself |
| 23:28.39 | [hC] | and nothing goes in |
| 23:28.39 | [hC] | any idea? |
| 23:28.45 | dan__t | thanks, ptinsley. |
| 23:29.12 | ptinsley | dan__t, np |
| 23:29.12 | justinu | check the b2b voip page on voip-info.org |
| 23:29.28 | ptinsley | [hC], you might have a corrupted global directory, i have seen that cause the polycoms to freak out |
| 23:29.33 | dan__t | Would it be dumb to inquire some sort of ATA software to convert a regular computer with an analog modem for this purpose? |
| 23:29.48 | knarfly | dlynes_office: Okay, I think I have something here. |
| 23:30.07 | ptinsley | [hC], do you have entries in your 0000....00-directory.xml ? |
| 23:30.16 | [hC] | I think i may have had bad perms in the ftp directory |
| 23:30.22 | knarfly | dlynes_office: I found the file zaptel.sh but I had to assign some variables to get it to run. |
| 23:30.23 | [hC] | for creating the >mac>-directory.xml file |
| 23:30.36 | ptinsley | that could cause problems for sure |
| 23:31.17 | knarfly | dlynes_office: it crashed saying it could not find /usr/local/etc/zaptel.conf so I touched it and then it ran without errors. But it did not say anthing else either. I assume it loaded. |
| 23:31.46 | knarfly | dlynes_office: how can I test the moh now? |
| 23:31.53 | dlynes_office | knarfly: yeah....modify your zaptel.sh so that it loads ztdummy |
| 23:32.02 | dlynes_office | knarfly: you don't need any of the other modules |
| 23:32.11 | ptinsley | has anybody had odd PRI problems since the 1.2.9.1 release? I have one PBX that had been working fine for months and after that update woudl start screwing up channels and have to have asterisk restarted |
| 23:32.41 | knarfly | dlynes_office: how do I do that? |
| 23:32.56 | dlynes_office | knarfly: vi |
| 23:33.05 | nettie | uhmm guys, I'm testing *-dev but when I make a call, locally or not after a couple of seconds I need very strong buzzzzzzzes, anyone know what could be wrong? I'm running the stble branch since a long time and never had such problems.. any idea please? thanx. |
| 23:33.35 | knarfly | dlynes_office: I use ee but what do I type into this file. load ztdummy or kldload ztdummy? |
| 23:34.50 | dlynes_office | kldload |
| 23:35.08 | Un1x | knarfly FreeBSD? |
| 23:35.37 | Dr-Linux | ptinsley: what's your problem? |
| 23:35.45 | knarfly | dlynes_office: Yes FreeBSD-6.1-RELEASE |
| 23:35.59 | Dr-Linux | ptinsley: i'm using 1.2.9.1 with two PRI's |
| 23:36.02 | knarfly | Un1x: Yes FreeBSD-6.1-RELEASE |
| 23:36.14 | Un1x | heh |
| 23:36.27 | Un1x | i dont think you can use, kldload |
| 23:36.30 | Un1x | for ztdummy |
| 23:36.34 | Un1x | not sure but give it a try |
| 23:37.22 | knarfly | dlynes_office: I just was informed that kldstat will list the kernel modules. zaptel.ko is loaded now |
| 23:37.46 | dlynes_office | knarfly: yeah...that's what it was |
| 23:38.00 | dlynes_office | knarfly: i'm not a freebsd guy...but i did run asterisk on freebsd for a little while |
| 23:39.10 | knarfly | dlynes_office: kldload ztdummy from the command line doesn't work. |
| 23:39.55 | knarfly | dlynes_office: if I edit zaptel.conf do I just need one line that says ztdummy? |
| 23:40.06 | ptinsley | Dr-Linux, it sees an incoming line and tries to pick it up on the wrong zap channel and freaks out saying it's in use |
| 23:40.58 | ptinsley | Dr-Linux, i disabled frame buffer and haven't seen the problem since, but it was a very spotty problem to begin with so I am just crossing my fingers |
| 23:41.36 | Dr-Linux | ptinsley: from where you disbale frame buffer? |
| 23:41.54 | knarfly | dlynes_office: oky that doesn't work. It wants a <keyword>=<value> syntax? Any ideas? |
| 23:41.56 | ptinsley | os boot nofb |
| 23:42.03 | Dr-Linux | ptinsley: did you delete your modules when you were upgradign to new version? |
| 23:42.14 | ptinsley | yes |
| 23:43.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:43.16 | Dr-Linux | ptinsley: the same happend with outgoing calls? |
| 23:43.21 | ptinsley | nope, only inbound |
| 23:43.33 | ptinsley | oubound would still function fine, only inbound calls would fail |
| 23:43.42 | ptinsley | and callers would get the typical something is wrong fast busy |
| 23:43.45 | Dr-Linux | anybody hash addict? |
| 23:44.01 | Un1x | <PROTECTED> |
| 23:44.07 | justinu | drugs are bad, mmmkay? |
| 23:45.53 | Dr-Linux | justinu: tribal growt post(durg seed) widely |
| 23:46.18 | Dr-Linux | growth or what, i can't spell |
| 23:46.28 | justinu | just grow |
| 23:46.58 | Dr-Linux | justinu: and what we call the place where it grow? :S |
| 23:47.09 | Dr-Linux | the earth place |
| 23:47.19 | justinu | fields? |
| 23:47.21 | justinu | farm? |
| 23:47.34 | Dr-Linux | oo yeah fields |
| 23:47.57 | *** join/#asterisk ctaloi (n=Chris@cpe-24-58-22-17.twcny.res.rr.com) |
| 23:48.16 | justinu | growing poppies? |
| 23:48.53 | Dr-Linux | justinu: what's poppies? |
| 23:49.17 | justinu | http://en.wikipedia.org/wiki/Opium_poppy |
| 23:50.27 | *** join/#asterisk backblue (n=moo@87-196-14-128.net.novis.pt) |
| 23:50.46 | Dr-Linux | justinu: yes correct poppies |
| 23:50.57 | knarfly | Okay gang...the docs all explain it with Linux so I'm going to install this spare drive. Build Fedora 5.0 and try it that way. Zaptel and FeeBSD don't seem to get along very well. Wish me luck. I'll report back in an hour or so. |
| 23:51.12 | justinu | it's interesting, in the USA Hash comes from marijuana plant |
| 23:51.22 | justinu | but other parts of the world call opium "hash" |
| 23:51.51 | Un1x | yea Marijuana hash is better then opium hash :P |
| 23:51.57 | dlynes_office | justinu: Canada calls hashish the oil from cannabis, too |
| 23:51.59 | Qwell[] | than.. |
| 23:52.02 | Dr-Linux | yesssssssssss |
| 23:52.06 | justinu | less addictive, for sure |
| 23:52.11 | Dr-Linux | it's Marijuana hash |
| 23:52.33 | Dr-Linux | i play cricket in Marijauna hash fields |
| 23:52.44 | justinu | heh |
| 23:52.52 | justinu | don't bogart it, dude |
| 23:53.20 | Dr-Linux | justinu: we don't do that, bcoz we grow all these shit and we know how much it's bad :( |
| 23:53.24 | ptinsley | totally off topic here, does anybody know a decent not too pricey flash resource. I had one flake out on me for a job so I need to find someone to do the work |
| 23:53.53 | Dr-Linux | dlynes_office: but really i smoked hash a few time, just to check the quality. |
| 23:54.02 | Qwell[] | just a few, eh? |
| 23:54.03 | Dr-Linux | but not hiroin? |
| 23:54.04 | Qwell[] | sure, sure |
| 23:54.07 | justinu | thats my excuse too |
| 23:54.11 | justinu | i was testing the purity |
| 23:54.17 | Dr-Linux | justinu: what you guys call Hirion? |
| 23:54.20 | *** join/#asterisk Ixthod (n=Ixthod@198.174.206.41) |
| 23:54.20 | justinu | heroin |
| 23:55.26 | Dr-Linux | justinu: Ayub afridi is very big dealer sells heroin to USA/UK/CA |
| 23:55.47 | justinu | stay away from that stuff |
| 23:55.58 | Qwell[] | heh |
| 23:56.00 | Un1x | heh, man i'm bored, fuck i wish i lived in asia or somethin |
| 23:56.05 | Un1x | have bunch of weed fiels |
| 23:56.07 | Qwell[] | instant massive addiction |
| 23:56.08 | Dr-Linux | he was wanted by USA, but my country had no power to give him |
| 23:56.19 | Dr-Linux | so he went to USA by himself, |
| 23:56.30 | Dr-Linux | but not sure he came back to work |
| 23:56.32 | CunningPike | ptinsley: Remote OK? |
| 23:56.41 | ptinsley | CunningPike, ya |
| 23:57.01 | ptinsley | as long as they are on earth and can speak or write english we are good |
| 23:57.08 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 23:57.13 | Dr-Linux | justinu: HE is the one supply stinger miseal to USA, do you remember the afghaistan russia war ... story |
| 23:57.14 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:57.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:57.17 | CunningPike | ptinsley: www.elliotmedia.com |
| 23:57.36 | CunningPike | ptinsley: Just did this for us: http://www.dnv.org/popular_report/2006/ |
| 23:57.39 | justinu | Dr-Linux: yeah, but I thought that the USA supplied stingers to the afgan |
| 23:57.46 | *** part/#asterisk ctaloi (n=Chris@cpe-24-58-22-17.twcny.res.rr.com) |
| 23:57.52 | *** join/#asterisk ctaloi (n=Chris@cpe-24-58-22-17.twcny.res.rr.com) |
| 23:57.59 | Qwell[] | Dr-Linux: missile |
| 23:58.30 | Dr-Linux | justinu: yes, but US was enough smart, they asked to Afghan after ending war they will give back the stingers and they will be paid |
| 23:58.52 | justinu | oh, ic |
| 23:59.06 | justinu | but obviously they kept some |
| 23:59.12 | Dr-Linux | justinu: so Ayub afridi sent meseal to US more than then gave to Afghan |
| 23:59.40 | Dr-Linux | they* |
