| 00:00.12 | drmessano | Amongst others, yeah |
| 00:00.19 | drmessano | I dont have friends, so dont use much |
| 00:00.36 | mchou | hehe |
| 00:01.26 | hardwire | drmessano: lul |
| 00:01.27 | mchou | drmessano: what's flowroute's sip server addr? wanna do a traceroute |
| 00:02.26 | drmessano | hardwire: If only I was joking.. Dude, customers love me.. Everyone else hates my guts.. |
| 00:02.44 | drmessano | sip.flowroute.com |
| 00:03.27 | hardwire | drmessano: I like you in small doses. |
| 00:03.32 | hardwire | cheers. |
| 00:03.48 | drmessano | hugs hardwire and humps his leg |
| 00:03.59 | hardwire | an excellent example. |
| 00:04.19 | mchou | hmm, traceout aint bad |
| 00:04.28 | drmessano | On IRC it's mildly amusing, in jail it's difficult to explain |
| 00:04.29 | mchou | traceroute* |
| 00:05.16 | drmessano | mchou: I use them for termination.. I wish there was a little bit cheaper DID option so I can route my google voice to it |
| 00:05.33 | drmessano | Still using the Gizmo thing, and dont like the idea much |
| 00:05.48 | drmessano | Google needs to add SIP or GTalk accounts as options |
| 00:05.48 | mchou | gizmo sucks |
| 00:05.57 | mchou | agreed |
| 00:06.23 | drmessano | Problem is that they make no money from SIP or Gtalk |
| 00:06.27 | mchou | you heard about FCC now looking at google voice w/ microscope? |
| 00:06.46 | drmessano | Since when? |
| 00:06.55 | mchou | sine Friday |
| 00:06.56 | shido6 | so they find out that google voice is transcribing your calls to text in a searchable database, so what? |
| 00:06.58 | riddlebox | I think I will return the magic jack, sign up for broadvoice again, and port my teliax number whoa thats a whole lot to do |
| 00:07.03 | mchou | since* |
| 00:07.06 | *** join/#asterisk moy (n=moy@CPE001d7260250c-CM00140454d33e.cpe.net.cable.rogers.com) |
| 00:07.16 | drmessano | Google can have my calls |
| 00:07.31 | mchou | shido6: no, this is about access |
| 00:07.34 | drmessano | Just me discussing my boil with my doctor, and advanced draining methods |
| 00:07.36 | riddlebox | I have google voice whats going on with that? |
| 00:07.37 | drmessano | Search that |
| 00:07.45 | mchou | so called net neytrality |
| 00:07.48 | hardwire | riddlebox: what happened w/ teliax? |
| 00:08.01 | drmessano | brb, time for my 8PM boil draining |
| 00:08.18 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:08.46 | riddlebox | just not happy with them, they have features you can change in your account, you change it and then the account doesnt work anymore |
| 00:09.08 | dlewis | google needs to add SIP support for Google Voice |
| 00:09.22 | mchou | drmessano: http://www.wired.com/epicenter/2009/10/fcc-google-voice/ |
| 00:09.40 | mchou | I dont even know what the hell they referring to |
| 00:09.53 | *** join/#asterisk t (i=tom@freenode/staff/tomaw) |
| 00:10.12 | *** join/#asterisk iarp (n=iarp@d221-84-143.commercial.cgocable.net) |
| 00:10.22 | mchou | I use sipgate, they use rural" telco and carge outrageous settlent fees. I still get calls routed through google voice |
| 00:10.36 | mchou | charge* |
| 00:10.43 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 00:11.20 | hardwire | riddlebox: doh.. sorry.. |
| 00:11.24 | hardwire | I don't have that issue |
| 00:11.34 | KavanS | I have a problem, my followme is leaving voicemails on my cell phones...what can I do about this? |
| 00:11.35 | hardwire | They are usually who I refer people too. |
| 00:11.44 | hardwire | KavanS: use call ack |
| 00:11.49 | hardwire | I have a macro for it |
| 00:12.08 | hardwire | I think followme supports it however |
| 00:12.15 | riddlebox | hardwire, I was happy till this weekend, and not happy that I can make calls and cannot receive them |
| 00:12.40 | hardwire | riddlebox: doh |
| 00:12.50 | mchou | riddlebox: I suspect that's not teliax but your asterisk |
| 00:12.56 | hardwire | KavanS: what is "takecall" set to? |
| 00:13.32 | mchou | riddlebox: what does asterisk console say when you call yourself vial teliax? |
| 00:13.38 | mchou | via* |
| 00:13.42 | riddlebox | mchou, its not I have been fighting it for two days, I even switched the account back to sip and used my sip.conf and it did the same thing |
| 00:13.47 | hardwire | it says you have a call? |
| 00:13.48 | KavanS | hardwire, would you mind pastebin'ing that? |
| 00:13.54 | riddlebox | mchou, it never reaches me |
| 00:13.58 | hardwire | KavanS: are you using followme.conf? |
| 00:14.01 | hardwire | or followme? |
| 00:14.02 | KavanS | hardwire, I am using a "screen" macro for this, and it says "press 1 to accept" |
| 00:14.21 | KavanS | the reason I am doing this is so I can multi-dial (concurrently), SIP extensions + cell |
| 00:14.21 | p3nguin | drmessano: Does flowroute allow you to set your own CID per call, or is it locked-in on the web portal? |
| 00:14.38 | hardwire | KavanS: are you using the FollowMe application? |
| 00:14.39 | mchou | riddlebox: take down your firewall and call youself |
| 00:14.41 | KavanS | hardwire, so on the voicemail I get "press 1 to accept" |
| 00:14.45 | mchou | yourself* |
| 00:14.51 | KavanS | hardwire, no I am not using the followme application |
| 00:15.31 | hardwire | KavanS: ok.. I'll paste what I have. |
| 00:15.33 | hardwire | it's simple |
| 00:15.41 | hardwire | you should check out FollowMe |
| 00:15.54 | Orbixx | hurr |
| 00:15.59 | Orbixx | I have an epic Asterisk server. |
| 00:16.10 | *** part/#asterisk mumtazah (n=mumtazah@203.82.91.101) |
| 00:16.26 | Orbixx | and that was oddly to the wrong channel |
| 00:16.35 | KavanS | yeah I can't concurrent dial with followme it seems |
| 00:17.45 | KavanS | and doesn't followme do the same thing, play an announcement to accept the call? |
| 00:18.14 | riddlebox | mchou, it is just the router, and it worked all the time up until yesterday |
| 00:18.20 | riddlebox | the asterisk box has no firewall |
| 00:19.06 | mchou | riddlebox: your router's firewall is not active? |
| 00:19.32 | riddlebox | mchou, I put the system in the DMZ and same thing |
| 00:19.56 | mchou | nah |
| 00:20.05 | mchou | I find it hard to believe |
| 00:20.27 | mchou | your expernip in sip.conf is probably fubar |
| 00:20.34 | hardwire | KavanS: http://hardwire.pastey.net/126388 |
| 00:20.34 | riddlebox | the only changes made were to my teliax account to use iax instead of sip |
| 00:20.55 | mchou | externip* |
| 00:22.18 | riddlebox | eternhost is correct |
| 00:22.23 | mchou | riddlebox: in fact take down asterisk and just use a softphone |
| 00:22.39 | KavanS | hardwire, what happens if the cell phone's voicemail picks up and starts recording? |
| 00:22.49 | mchou | riddlebox: that should be all you need for a sanity check |
| 00:23.00 | riddlebox | I can try that real quick |
| 00:23.25 | mchou | riddlebox: I recommend twinkle |
| 00:23.38 | riddlebox | twinkle, never used that one, I always use xlite |
| 00:23.49 | mchou | xlite is nasty |
| 00:24.04 | p3nguin | That new one barely runs for me. |
| 00:24.12 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 00:24.20 | p3nguin | I use zoiper on Windows or twinkle on Linux. |
| 00:24.31 | mchou | agreed |
| 00:24.42 | mchou | accept no substitutes |
| 00:25.06 | [TK]D-Fender | eyebeam > both |
| 00:25.35 | [TK]D-Fender | Zoiper is a little better than X-Lite on Windows for having a Transfer button at least... |
| 00:26.01 | *** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 00:26.57 | hardwire | twinkle.. sigh |
| 00:27.30 | mchou | hardwire: what's wrong with it besides the UI? |
| 00:27.52 | p3nguin | It usually works just fine. |
| 00:28.00 | mchou | yup |
| 00:28.17 | mchou | most solid "free" softphone on linux |
| 00:31.06 | KavanS | christ... |
| 00:31.15 | KavanS | if I can get it to work on ltsp, I would be impressed |
| 00:31.18 | KavanS | twinkle that is... |
| 00:31.26 | hardwire | mchou: it's named twinkle. |
| 00:32.07 | drmessano | Is that worse than "DAHDI"? |
| 00:32.18 | mchou | lol |
| 00:32.34 | mchou | Dahdi needs to twi8nkle |
| 00:32.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:33.36 | drmessano | After 1.6.2 I wont have to worry about DAHDI |
| 00:33.38 | drmessano | So yay |
| 00:35.37 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 00:35.39 | drmessano | For those using PRI, at what number of analog lines does the cost of a partial PRI begin to become cost effective? |
| 00:35.44 | [TK]D-Fender | drmessano: DAHDI...... because Digium can't let Fonality get all the gayest names possible ;) |
| 00:35.47 | mchou | drmessano: you dont use paul mark's gv.agi with google voice? |
| 00:36.21 | KavanS | ok, so how would one send DTMF tones in a macro? are they just sound files for playback? |
| 00:36.37 | drmessano | mchou: Hell no? |
| 00:37.01 | drmessano | [TK]D-Fender: Absolutely.. "Check out our sexy box" |
| 00:37.04 | mchou | drmessano: umm, wgat's wrong with it? |
| 00:37.14 | mchou | what* |
| 00:38.05 | mchou | drmessano: it's a bit ghetto but it gets the job done |
| 00:38.10 | drmessano | mchou: Im sick of hacky bullshit.. I dont want to have to run cron jobs to update hacky scripts to make a friggin phone call.. Even on my home hobby box, I am just not that "Bohemian" |
| 00:38.26 | mchou | OK |
| 00:38.48 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 00:39.23 | drmessano | Its bad enough I run asterisk.. want to talk about HACKY? |
| 00:39.29 | drmessano | j/k |
| 00:39.35 | drmessano | DONT KILL ME RUSSELL |
| 00:40.06 | drmessano | Generally if I read about it on the Nerdvittles site, it's too hacky to be reliable |
| 00:40.29 | mchou | no, the nerdvittles stuff is a total hack |
| 00:40.39 | mchou | paul marks is not a hack |
| 00:40.44 | KavanS | pro tips, how to playback DTMF tones to user? |
| 00:41.12 | drmessano | "Why use Skype For Asterisk when you can use the NV-Skype-Orgasmatical-Afterglowical-Fangoriously-Fangorious-NotSkype-Skype.agi" |
| 00:41.13 | mchou | nerdvittles attempted to "improve" on Mark's script |
| 00:41.19 | drmessano | Ok |
| 00:41.34 | mchou | nerdvittles just made a mess |
| 00:42.26 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 00:42.26 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:42.54 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 00:43.12 | p3nguin | How weird. Why did the default moh files that come with 1.4.24.1 change? |
| 00:43.27 | drmessano | "The script calls 7 friends and plays pleasecallme.gsm to each one, asking them to call your GV number. After commiting by entering 3483798493 on the keypad, asterisk hook flashes on their line to a second outgound line through acoustical coupling and attempts to call 7 of their friends, but NONE of the original 6, once the next call is complete, the script logs in to google and requests a GV |
| 00:44.22 | dlewis | lol |
| 00:44.42 | drmessano | invite.. after waiting 10 to 15 days, the invite is received, number is set up, and another call is placed through the new google voice number back to your google voice number using a PHP-Python-Rail-over-COBOL-C++ script and initiates the 4th leg of the call out to the party you originally dialed" |
| 00:45.12 | drmessano | WOW, FREE CALLS |
| 00:45.14 | drmessano | WGET NOW |
| 00:47.27 | drmessano | "John, I told you to call me back in 30 mins.. You waited two weeks" "NUH UH. I MADE A FREE CALL WITH GOOGLE VOICE OVER ASTERISK OVER PHP OVER GOOGLE OVER SIP OVER ASTERISK".. "Oh, I see.. Sorry I yelled at you" |
| 00:48.41 | Jumpie | lol |
| 00:49.21 | Jumpie | anybody know of any sinatra MOH samples |
| 00:49.26 | Jumpie | maybe already pre looped |
| 00:49.27 | Jumpie | :P |
| 00:49.30 | drmessano | "We mock what we don't understand" - Austin Millbarge |
| 00:49.54 | p3nguin | [Oct 11 19:49:40] ERROR[24880]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
| 00:50.02 | p3nguin | What does this mean? |
| 00:50.13 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 00:50.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:50.36 | Jumpie | somebody had butter fingers when passing the bowl p3nguin |
| 00:50.44 | Jumpie | broke his nice pipe 3 weeks ago :( |
| 00:51.00 | drmessano | Puff, Puff, Pass.. Someone broke the rotation |
| 00:51.15 | p3nguin | I can see how that might return an error. |
| 00:51.44 | p3nguin | I'm more curious about why it appears on my asterisk console when I get a call. |
| 00:52.00 | dlewis | anyone have any issues/bad experience with the openvox tdm400p cards? |
| 00:52.03 | drmessano | Also if you have a girlfriend who has never swung from the chandelier in your bedroom before.. can result in "broken pipe" |
| 00:52.17 | drmessano | dlewis: Mine works fine.. 2 FXO/2 FXS |
| 00:52.32 | Jumpie | lol |
| 00:52.38 | Jumpie | p3nguin only on inbound? |
| 00:53.02 | dlewis | drmessano: ok |
| 00:53.12 | [TK]D-Fender | dlewis: You are just a chronic cheapskate, aren't you? |
| 00:53.13 | p3nguin | jumpie: yes |
| 00:53.13 | dlewis | drmessano: looking to get 1 FXO 1 FXS |
| 00:53.38 | dlewis | [TK]D-Fender: nope, just don't want to spend a lot if I don't have to |
| 00:53.43 | dlewis | you forget, we're in a recession. |
| 00:53.49 | drmessano | Is this for home or office? |
| 00:53.49 | [TK]D-Fender | dlewis: First a Grandsuck gatway, then looking at X100P, Old Siupura device, and now working your way up to DELUXE CRAP |
| 00:54.04 | dlewis | [TK]D-Fender: thanks for your opinion |
| 00:54.07 | dlewis | drmessano: home |
| 00:54.51 | p3nguin | jumpie: I just called one of my DIDs on another provider -- the errors did not appear. |
| 00:54.51 | [TK]D-Fender | Everyone is entitled to my own opinion :D |
| 00:54.52 | drmessano | I wouldnt knock one for home use |
| 00:54.52 | drmessano | Not sure about long term business use.. Especially with newer designs out there |
| 00:54.52 | Jumpie | so then it has to do with the way you're route or connection string is or somethin it sounds like |
| 00:54.52 | dlewis | drmessano: it'll be for home use and that's it. |
| 00:56.06 | drmessano | dlewis: Grantid, I have no friends, but the FXS works well here.. So... |
| 00:56.20 | dlewis | drmessano: thanks |
| 00:58.27 | el_critter | Hi, can I use asterisk to call home? |
| 00:59.05 | [TK]D-Fender | el_critter: * can be used to place calls over any hardware or service you configure it for |
| 00:59.41 | drmessano | Not without a record player, saw blade, and some Reeses Pieces |
| 00:59.42 | el_critter | [TK]D-Fender: but I don't remember the number :( |
| 00:59.54 | [TK]D-Fender | el_critter: Oh well |
| 01:01.30 | *** join/#asterisk khratos (n=jespinal@190.80.200.56) |
| 01:01.50 | drmessano | How could I forget the speak N spell |
| 01:03.10 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:05.15 | ChannelZ | those ruled |
| 01:08.36 | KavanS | dlewis, I just bought some of those, I'm hoping they work |
| 01:10.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:12.02 | dlewis | KavanS: cool |
| 01:12.24 | dlewis | KavanS: you in portland? |
| 01:12.35 | KavanS | dlewis, if I somehow get fucked by using them I'm sure I'll be going off about it |
| 01:12.38 | KavanS | yeah I'm in the burbs |
| 01:13.17 | hardwire | I'm interested in how the openvox cards work too. |
| 01:13.39 | hardwire | I can't discount them as an option until I know more. |
| 01:14.02 | dlewis | ok |
| 01:14.14 | KavanS | yeah, I have an official digium card...but christ, for the price of openvox, I got two cards |
| 01:14.38 | KavanS | the quality is probably not the same though |
| 01:14.41 | KavanS | definitely chinese... |
| 01:15.16 | ChannelZ | does it work in english? |
| 01:15.21 | hardwire | pokes ChannelZ |
| 01:15.38 | ChannelZ | or does it only make gong sounds when the phone rings |
| 01:16.10 | hardwire | I wonder how many chinese people are in this channel. |
| 01:16.10 | KavanS | when you say "really" it comes out "weally" |
| 01:16.19 | KavanS | that's the major problem currently with openvox hardware... |
| 01:16.34 | hardwire | ugh |
| 01:16.54 | hardwire | goes back to clusterip mucking |
| 01:16.57 | KavanS | heh sorry, bad joke |
| 01:17.23 | ChannelZ | snickers |
| 01:17.53 | drmessano | Dont forget the Shlimp flied lice |
| 01:20.16 | hardwire | cries |
| 01:20.50 | ChannelZ | Whats the matter? |
| 01:20.55 | ChannelZ | Did someone pee pee in your coke? |
| 01:21.04 | ChannelZ | ba-dum-tish! |
| 01:21.47 | *** part/#asterisk Baelig (n=chatzill@user-0c8h5rq.cable.mindspring.com) |
| 01:21.49 | hardwire | I'm 0.001% chinese american. |
| 01:22.14 | hardwire | My people do not appreciate those jokes. |
| 01:22.21 | drmessano | I slept with a chinese girl once, and then ate indian food.. I felt like such a bastard |
| 01:22.28 | hardwire | heheh |
| 01:22.33 | hardwire | damnit |
| 01:22.37 | ChannelZ | your people will have the last laugh, the way Obama is spending money |
| 01:22.53 | hardwire | really.. politics? |
| 01:23.04 | hardwire | can't we stick to crappy somewhat topic related hardware? |
| 01:23.06 | drmessano | He was born in Kenya, he's not chinese.. TMZ fail |
| 01:23.26 | KavanS | fail, voicepulse registration |
| 01:23.32 | ChannelZ | belches |
| 01:23.38 | hardwire | wants tacos |
| 01:23.47 | hardwire | I SHALL HAVE THEM TOOOOOO! |
| 01:23.57 | ChannelZ | I just had a burrito bowl |
| 01:23.59 | ChannelZ | tastes of chicken |
| 01:24.11 | [8none1] | hardwire: now that's offensive |
| 01:24.18 | hardwire | Why is your name taco? |
| 01:24.27 | KavanS | if you taco the hell out of my taco, we got a problem. |
| 01:24.56 | [8none1] | The chinese guy is making fun of mexicans |
| 01:25.04 | drmessano | Buying a $30 bottle of spices good for only indian food = indian food 3x a week.. Not sure what the hell we ate tonight, but i'm guessing it was a CURRY |
| 01:25.06 | hardwire | I honestly want a taco |
| 01:25.13 | hardwire | or two |
| 01:25.19 | hardwire | or however many I can fit in my car |
| 01:25.28 | drmessano | I want a burger king taco |
| 01:25.29 | ChannelZ | Sexist, racist, scaremongerers! |
| 01:25.36 | drmessano | RIp Burger King tacos |
| 01:25.46 | hardwire | carl jrs has tacos now |
| 01:25.48 | ChannelZ | BK had tacos? hmm |
| 01:25.50 | hardwire | I'd avoid it. |
| 01:26.04 | hardwire | is going to make tacos using the good stuff |
| 01:26.14 | ChannelZ | real mexicans |
| 01:26.19 | drmessano | http://www.pusateri.org/cruft/docs/bktacos.html |
| 01:26.23 | drmessano | YES |
| 01:26.31 | hardwire | http://foodstuffs.bogomip.com/2007/08/praise-allen-cowley-spice-dust.html |
| 01:26.40 | drmessano | Tacos made with migrant workers are VERY crunchy |
| 01:26.44 | ChannelZ | hey those are total rip-offs of Jack In The Box tacos |
| 01:27.15 | ChannelZ | right down to the paper pockets they come in |
| 01:27.18 | *** join/#asterisk [domon] (n=[domon]@99-171-100-59.lightspeed.irvnca.sbcglobal.net) |
| 01:27.35 | hardwire | drmessano: even though I'm in Alaska.. putting up a huge wall between us and Mexico will just mean less and less tacos. |
| 01:27.38 | drmessano | I didnt taste the e. coli |
| 01:27.41 | hardwire | and I'm not prepared to take that step |
| 01:28.07 | drmessano | hardwire: I can see tacos from my porch! |
| 01:28.12 | drmessano | I mean, russia |
| 01:28.24 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:28.29 | hardwire | is so tired of that. |
| 01:28.33 | drmessano | hahah |
| 01:28.52 | drmessano | The good news is.. |
| 01:28.53 | hardwire | I can see tacos from my porch however. |
| 01:29.00 | [domon] | hi guys, i'm using asterisk and a billing system to offer prepaid service. when a customer dials in to a specific number, i want to run a script to determine their remaining balance. how can i setup asterisk to call my particular script? from what i gather I can use agi to do this, but i'm not sure how... |
| 01:29.16 | hardwire | [domon]: Hi you! http://www.a2billing.org/ |
| 01:29.25 | [domon] | hardwire: i already have a billing platform |
| 01:29.32 | [domon] | that isn't a2billing |
| 01:29.33 | hardwire | doesn't sound like it. |
| 01:29.34 | drmessano | But not the one we linked |
| 01:29.45 | [domon] | actually, i'm the maintainer of it ;) |
| 01:29.51 | drmessano | Wait |
| 01:29.58 | hardwire | you want to do something that billing software does. |
| 01:30.00 | [domon] | (of the billing platform i'm talking about) |
| 01:30.08 | drmessano | Its your billing platform, and you cant figure out how to make it work? |
| 01:30.10 | hardwire | so that implied the lack-there-of. |
| 01:30.16 | hardwire | brb.. |
| 01:30.16 | drmessano | Dude, remind me not to download |
| 01:30.20 | hardwire | hands the floor to drmessano |
| 01:30.23 | [domon] | no, i'm trying to implment this functionality in my billing software |
| 01:30.26 | ChannelZ | clicks excitedly |
| 01:30.28 | [TK]D-Fender | [domon]: "core show application agi" |
| 01:30.32 | hardwire | agi ftw |
| 01:30.36 | hardwire | fastagi ftb |
| 01:30.39 | hardwire | (for the butter) |
| 01:30.42 | [domon] | and from what i gather, agi is the way to do it |
| 01:30.42 | drmessano | Maybe its in one of your INI files |
| 01:30.45 | hardwire | brb.. ladies home.. gonna go have tacos. |
| 01:30.46 | hardwire | wait. |
| 01:30.50 | hardwire | runs away |
| 01:30.58 | ChannelZ | fish tacos |
| 01:31.12 | hardwire | go for the obvious joke why doncha |
| 01:31.12 | drmessano | ChannelZ: thats dessert |
| 01:31.15 | [domon] | the only thing i'm trying to figure out is the interface between asterisk and the billing system database |
| 01:31.30 | drmessano | [domon]: AGI, can't say it enough |
| 01:31.31 | p3nguin | I think I may have found the broken pipe problem! |
| 01:31.33 | drmessano | A G I |
| 01:31.34 | [domon] | (i am not an asterisk guy) |
| 01:31.42 | [domon] | right, i know it's agi...(didn't i already say that?) |
| 01:31.46 | drmessano | p3nguin: Loose chandelier? |
| 01:32.10 | Jumpie | anybody know of a resource with good MOH samples? |
| 01:32.10 | p3nguin | I'm supposed to be running a callerid perl script, but I didn't put the script on the server. |
| 01:32.10 | ChannelZ | Damnation! |
| 01:32.10 | drmessano | [domon]: great, thanks |
| 01:32.18 | ChannelZ | Jumpie: Your CD shelf |
| 01:32.27 | drmessano | Jumpie: Napster |
| 01:32.59 | ChannelZ | Or go click on all the samples in itunes and record them |
| 01:33.04 | drmessano | Jumpie: Audiogalaxy Satellite, KaZaa, Gnutella, WinMX, and Napigator |
| 01:33.16 | ChannelZ | like a medley |
| 01:33.36 | [domon] | drmessano: do you happen to have any examples? i'm assuming what i'd want to do is use 'answer' in my script and then do some functional processing on it (and perhaps say digits) |
| 01:33.39 | Jumpie | i just thought maybe they had to be certain lengths |
| 01:33.44 | Jumpie | and it'd be already looped for seamlessness :D |
| 01:33.48 | Jumpie | i know where to get mp3s lol |
| 01:34.15 | ChannelZ | go grab some techno music then. That shit is nothing but loops |
| 01:34.17 | drmessano | [domon]: Nope, I don't actually use Asterisk.. i'm here for help with a Skype problem |
| 01:34.23 | *** join/#asterisk johnakabean (n=john@static-173-50-101-13.nrflva.east.verizon.net) |
| 01:34.40 | drmessano | So you guys really think it's that one DLL? |
| 01:34.52 | ChannelZ | definately |
| 01:34.56 | drmessano | I'll pastebin my registry, BRB |
| 01:34.58 | ChannelZ | It's always a DLL |
| 01:35.51 | [TK]D-Fender | ~book |
| 01:35.52 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:35.54 | [TK]D-Fender | [domon]: ^^^^ |
| 01:35.56 | ChannelZ | Jumpie: here's a Sinatra sample for you: http://burner.com/sinatra.wav |
| 01:36.20 | *** join/#asterisk OrNix (n=ornix@89.251.107.21) |
| 01:37.29 | p3nguin | That didn't solve it. |
| 01:37.49 | ChannelZ | maybe the script is broken |
| 01:38.08 | p3nguin | I just copied it from my working box to the one where the script wasn't. |
| 01:38.55 | ChannelZ | yeah but it's perl |
| 01:40.39 | Jumpie | aaw ChannelZ you got my hopes up |
| 01:40.42 | Jumpie | thanks for blaring me out of my chair |
| 01:40.54 | Jumpie | you sir, receive one fail token |
| 01:41.15 | ChannelZ | Sorry. I must have mixed up Sinatra with the taco conversation |
| 01:41.19 | Jumpie | :) |
| 01:41.39 | *** join/#asterisk torrancew (n=chatzill@70.177.34.83) |
| 01:42.05 | torrancew | can anyone recommend a good softphone for os x? i'm not too fond of x-lite... |
| 01:43.15 | ChannelZ | zoiper |
| 01:46.08 | p3nguin | This doesn't make any sense. I restarted asterisk fully, (i.e. stopped it, waited a while, started it again), now moh does not work and the console says this: [Oct 11 20:44:54] WARNING[25063]: res_musiconhold.c:665 get_mohbyname: Music on Hold class 'default' not found |
| 01:46.24 | p3nguin | It was just working, and the default class certainly does exist. |
| 01:47.13 | p3nguin | module reload res_musiconhold.so makes it work again. |
| 01:47.18 | *** join/#asterisk misty (n=misty@96.248.147.137) |
| 01:47.35 | p3nguin | What's up with that, and how to I ensure it always works after a restart or fresh startup? |
| 01:47.49 | ChannelZ | never seen that |
| 01:50.20 | misty | I'm trying to trace down a problem with echo I'm having with TE122B card. I realized that I had software echo cancel turned on as well as the hardware module built into the card. Could having them both actually be making the echo problem worse? |
| 01:50.23 | p3nguin | I even tried loading it from modules.conf instead of automatically, but it still does not work after a restart. |
| 01:50.45 | p3nguin | That's going to be a problem, since I might not always be able to babysit moh. |
| 01:51.01 | ChannelZ | your * is hosed |
| 01:52.43 | p3nguin | I don't know what to do about it. |
| 01:53.06 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 01:53.13 | p3nguin | I did learn that the highest level of debug allowed is 2147483647. |
| 01:53.27 | torrancew | also, i've noticed that if someone i call on the pstn from asterisk terminates the call, it doesn't get terminated on ekiga, is there a setting for this in asterisk, or is ekiga not translating properly? |
| 01:54.28 | ChannelZ | they should be tearing down the call |
| 01:54.47 | torrancew | who is the "they"? |
| 01:56.03 | ChannelZ | ekiga? (the PSTN?) |
| 01:56.24 | ChannelZ | p3nguin: what version of * |
| 01:56.39 | p3nguin | I'm pretty sure the broken pipe is related to that script, too. |
| 01:56.49 | p3nguin | Asterisk 1.4.24.1 built by root @ asterisk.local.domain on a i686 running Linux on 2009-10-11 17:12:06 UTC |
| 01:56.59 | ChannelZ | WHOA asterisk.org just changed |
| 01:58.39 | p3nguin | to what? |
| 01:58.48 | drmessano | hey awesome |
| 01:59.12 | ChannelZ | it just looks better now |
| 01:59.13 | drmessano | Now that is cool |
| 01:59.17 | ChannelZ | that shitty purple is gone |
| 01:59.37 | ChannelZ | it's actually navigatable |
| 02:00.20 | drmessano | Ohh |
| 02:00.21 | drmessano | http://www.asterisk.org/docs |
| 02:00.24 | drmessano | ^^^ U go girl |
| 02:00.42 | ChannelZ | anyways I see lots of random bug fixes to MOH in Changelog, perhaps something is just wonky in your version p3nguin |
| 02:00.57 | ChannelZ | wow, finally |
| 02:02.06 | ChannelZ | hehe whoops now they're playing with it |
| 02:04.28 | *** join/#asterisk aurax (n=aurax@212.179.76.199) |
| 02:05.01 | drmessano | I BET ITS DRUPAL BASED |
| 02:05.05 | drmessano | Just sayin' |
| 02:05.07 | aurax | i bet |
| 02:10.26 | ChannelZ | do you have to authenticate at ekiga.net to call their test numbers? |
| 02:10.39 | johnakabean | is the argument passed from asterisk cli the ${ARG1} variable in the dialplan |
| 02:10.46 | johnakabean | the first argument |
| 02:13.38 | [TK]D-Fender | johnakabean: huh? |
| 02:13.45 | ChannelZ | my thoughts exactly |
| 02:13.51 | [TK]D-Fender | johnakabean: ARGX is for use with macros |
| 02:14.02 | [TK]D-Fender | johnakabean: Nothing to do with * CLI |
| 02:14.42 | johnakabean | if i do originate Local/243 extension s@somecontext john |
| 02:14.50 | johnakabean | does john get set as ARG1 |
| 02:15.45 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:16.08 | [TK]D-Fender | johnakabean: "help originate" <- what do the instructions say? |
| 02:16.29 | [TK]D-Fender | johnakabean: Does it tell you you can pass variables like that? |
| 02:17.05 | p3nguin | "The existing menuselect.makeopts file did not specify that 'app_dahdibarge' should not be included. However, either some dependencies for this module were not found or a conflict exists." How do I know what could conflict or depend? |
| 02:18.07 | johnakabean | i was mixed up fender to not use application but it says "This will originate a call between the specified channel tech/data and the |
| 02:18.07 | johnakabean | given application. Arguments to the application are optional. " |
| 02:18.28 | johnakabean | i just need explaination on the arguments |
| 02:18.40 | johnakabean | or is application confined to the applications listed |
| 02:18.53 | johnakabean | as part of asterisk |
| 02:19.22 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 02:19.57 | [TK]D-Fender | johnakabean: You aren't calling an application, you are directing it towards an extension. Completely different |
| 02:20.38 | ChannelZ | hmm the new forums skin needs work |
| 02:20.57 | johnakabean | is the application the context of the dialplan such as [customcontext] |
| 02:21.14 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:22.56 | johnakabean | asterisk programming is really short; they need a couple new dialplan applications such as "SetIf" to define variables based on whether another variable is set to something or based on boolean. |
| 02:22.58 | riddlebox | what do you guys think of the new look on asterisk.org now? |
| 02:22.59 | [TK]D-Fender | johnakabean: No. |
| 02:23.10 | johnakabean | then what is it? |
| 02:23.19 | [TK]D-Fender | johnakabean: You can choose execute a SINGLE APPLICATION *or* point to a dialplan context |
| 02:23.24 | [TK]D-Fender | johnakabean: A or B |
| 02:23.34 | [TK]D-Fender | johnakabean: jsut like in a call-file |
| 02:24.51 | johnakabean | you mean a dialplan context using "extension", not application based on what you just said |
| 02:25.54 | p3nguin | Usage1: originate <tech/data> application <appname> [appdata] |
| 02:25.59 | p3nguin | or |
| 02:26.01 | p3nguin | Usage2: originate <tech/data> extension [exten@][context] |
| 02:26.18 | johnakabean | I see that penguin |
| 02:26.27 | johnakabean | I'm very familiar with using extension |
| 02:26.30 | p3nguin | What was the problem? |
| 02:26.44 | johnakabean | application.........where do the applications come from? |
| 02:26.55 | Katty | peeks in |
| 02:27.00 | [TK]D-Fender | JOHNDialplan apps <- |
| 02:27.48 | johnakabean | my first question was "is application confined to the applications listed |
| 02:27.48 | johnakabean | <johnakabean> as part of asterisk" |
| 02:28.02 | p3nguin | Are you wanting to know an example of a valid app? |
| 02:28.04 | [TK]D-Fender | johnakabean: Yes, * Dialplan Apps |
| 02:28.20 | johnakabean | any application in the dialplan is in the list of "core show applications" |
| 02:28.29 | [TK]D-Fender | johnakabean: johnakabean Yes |
| 02:28.43 | johnakabean | so why did you say "no" when I asked that question |
| 02:29.06 | johnakabean | by saying no you was telling me we was not limited to the applications loaded in asterisk |
| 02:29.20 | [TK]D-Fender | johnakabean: No, you wording was very broken. |
| 02:29.23 | [TK]D-Fender | your* |
| 02:29.33 | [TK]D-Fender | [22:20]<johnakabean>is the application the context of the dialplan such as [customcontext] |
| 02:29.48 | johnakabean | that was AFTEr i asked that questiona nd you said no |
| 02:30.03 | [TK]D-Fender | johnakabean: You said 2 things and assumed which one I was answering. |
| 02:30.21 | [TK]D-Fender | [22:22]<[TK]D-Fender>johnakabean: No. |
| 02:30.28 | [TK]D-Fender | I said no 2 minutes later |
| 02:30.29 | johnakabean | I waited for you to respond in betwen |
| 02:30.43 | *** join/#asterisk moy (n=moy@CPE001d7260250c-CM00140454d33e.cpe.net.cable.rogers.com) |
| 02:30.54 | [TK]D-Fender | johnakabean: So I took longer than you liked and I responded to the one I did. |
| 02:31.11 | [TK]D-Fender | johnakabean: You didn't ask which until after a followup question. |
| 02:31.15 | [TK]D-Fender | johnakabean: Oh well |
| 02:31.32 | [TK]D-Fender | johnakabean: Either way you've got your answer now. Hopefully you're clear on this |
| 02:31.44 | johnakabean | that makes no sense, with me waiting for you to respond, as you would line up in the conversation correctly |
| 02:31.55 | johnakabean | ok thank you |
| 02:33.10 | [TK]D-Fender | johnakabean: I'm multi-tasking sorry if you you feel deprived of my full attention |
| 02:36.04 | johnakabean | so, asterisk needs another new features such as the ability to set arguments in dialplans on the extension command. I.E. "originate Local/200 extension s@context arg1 arg2" where arg1 and arg2 would be either {ARG1} {ARG2}, if set explicitly in the dialplan, or the ordered variables in the dialplan in the order they are set by the "Set" application. |
| 02:36.51 | johnakabean | :) |
| 02:37.54 | johnakabean | I will post the feature request URL on asterisk.org if you are interested; this could grow to be more than just on the command line. |
| 02:41.07 | [TK]D-Fender | johnakabean: AMI Originate and call-files already have this functionality |
| 02:43.53 | johnakabean | yes but call files aren't useful by an operator; I know you can telnet and write call files to be executed as I've done this but a regular switchboard operator would have no idea of the different contexts without the programmers writing them down. They would have to learn asterisk programming themselves. |
| 02:44.27 | [TK]D-Fender | johnakabean: You have to point to your dialplan context in your CLI originate. Same damn thing |
| 02:44.40 | [TK]D-Fender | johnakabean: Only you don't get to set variables |
| 02:44.55 | [TK]D-Fender | johnakabean: Since when is a switchboard operator going to be sitting in * CLI? |
| 02:45.00 | johnakabean | imagine them having to set everything in the call file versus less on the command line |
| 02:45.23 | [TK]D-Fender | johnakabean: Imagine you making a simple shell script to do this <- |
| 02:45.28 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 02:45.38 | [TK]D-Fender | johnakabean: Or a web page to issue it. oooohhh complicated.... |
| 02:45.44 | [TK]D-Fender | only not. |
| 02:46.07 | johnakabean | a script to write a call file to asterisk/outgoing |
| 02:46.12 | file | you don't give a switchboard operator access to such low level stuff, you build something on top of it that you control and can administrate and they use |
| 02:46.31 | johnakabean | I use one and I really like the AMD dialplan |
| 02:46.34 | [TK]D-Fender | johnakabean: Yeah sounds tragically difficult in a "hello world" kinda way |
| 02:46.48 | johnakabean | we still need setif |
| 02:47.07 | [TK]D-Fender | johnakabean: It isn't needed.... it'd just be a nice addition. |
| 02:47.19 | riddlebox | does anyone know of any sort of website where people can submit agi scripts, and you can search and download from it? |
| 02:47.32 | johnakabean | hotscripts.net |
| 02:47.40 | johnakabean | not specifically for asterisk |
| 02:48.04 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:48.12 | [TK]D-Fender | ~wikis |
| 02:48.13 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 02:48.15 | [TK]D-Fender | ^^^^^^^^^ |
| 02:48.26 | [TK]D-Fender | riddlebox: Merry Christmas, go make a page |
| 02:48.28 | drmessano | ~bosch |
| 02:48.44 | riddlebox | [TK]D-Fender, I just may |
| 02:48.56 | drmessano | ~boschlawnmowers |
| 02:48.57 | infobot | http://www.voip-info.org/wiki/view/Bosch+lawnmowers .. "Same to you, buddy!" |
| 02:49.05 | drmessano | My fav |
| 02:56.30 | *** join/#asterisk [8none1]_ (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net) |
| 03:01.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 03:37.06 | *** part/#asterisk khratos (n=jespinal@190.80.200.56) |
| 03:37.08 | navap | iarp: Hey |
| 03:38.16 | *** join/#asterisk raden (n=chatzill@66-168-4-200.dhcp.stpt.wi.charter.com) |
| 03:40.20 | *** part/#asterisk navap (i=navap@gateway/shell/blinkenshell.org/x-uopkiesxpmimonoo) |
| 03:42.12 | Katty | has swiss cheese, triscuits, and banana! |
| 03:44.02 | johnakabean | [TK]D-Fender: how do you make a call in the AMI using this: Channel: Local/240 |
| 03:44.02 | johnakabean | MaxRetries: 2 |
| 03:44.02 | johnakabean | RetryTime: 60 |
| 03:44.02 | johnakabean | WaitTime: 30 |
| 03:44.02 | johnakabean | Callerid:"Test Callerid"<123456789> |
| 03:44.03 | johnakabean | Context: ext-meetme |
| 03:44.05 | johnakabean | Extension: 901 |
| 03:46.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:49.08 | *** join/#asterisk mpls-eric (n=ejo1@ip68-98-123-122.ph.ph.cox.net) |
| 03:49.19 | Katty | so quiet. |
| 03:49.48 | mpls-eric | We're all on our way to Phoenix! |
| 03:50.19 | Maliuta | YAY! KATTY! |
| 03:50.24 | Maliuta | Katty: loud enough ;) |
| 03:50.29 | Katty | Maliuta: better. |
| 03:50.42 | Maliuta | some of us don't set foot on .us soil |
| 03:51.11 | drmessano | Some of .us have ankle bracelets |
| 03:51.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:51.42 | Maliuta | sets up the Asterisk way of life ... similar to the Ancient Egyptians we revere Katty's |
| 03:51.59 | drmessano | lol |
| 03:52.08 | Maliuta | drmessano: no, but I have a ball and chain ;P |
| 03:52.18 | Katty | ^_- |
| 03:52.30 | mpls-eric | So who's here in AZ already? |
| 03:52.41 | Maliuta | mpls-eric: it |
| 03:52.55 | Maliuta | 's [A-Z][a-z] |
| 03:53.15 | Maliuta | sends mpls-eric on a regex course |
| 03:53.28 | Katty | naptime, i do believe |
| 03:53.56 | Maliuta | Katty: but it's only 14:00 |
| 03:54.25 | *** join/#asterisk maagic (i=maagic@83.143.217.200) |
| 04:01.58 | [TK]D-Fender | johnakabean: Book shows the formatting rather clearly... so does the WIKI |
| 04:16.44 | *** join/#asterisk voxter (n=voxter@32.176.27.136) |
| 04:40.53 | p3nguin | channelz: I installed a bunch of perl crap, and now that script is working. |
| 04:46.55 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:48.38 | p3nguin | Looks like perl-libwww, perl-html-parser, perl-html-tagset, and perl-uri are the requisite packages for the script to succeed. |
| 04:49.24 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 04:51.44 | ManxPower-work | ~answers |
| 04:51.45 | infobot | answers is, like, Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 05:01.44 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
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| 05:11.46 | ChannelZ | p3nguin: it's what i hate about perl |
| 05:12.23 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-tqcrkhlusbtindko) |
| 05:12.36 | p3nguin | A little trial and error was all I needed. |
| 05:17.50 | ChannelZ | wanders off to watch Medium |
| 05:18.27 | [TK]D-Fender | rahail: /me waits for the upgrade to Full |
| 05:18.32 | [TK]D-Fender | waits for the upgrade to Full |
| 05:18.34 | [TK]D-Fender | darnit |
| 05:37.48 | *** join/#asterisk _pepo_ (n=kvirc@190.12.5.106) |
| 05:37.56 | _pepo_ | hi friends |
| 05:38.36 | _pepo_ | please, How much cost Cepstral if I try to integrate Asterisk |
| 05:38.36 | _pepo_ | ? |
| 05:40.32 | [TK]D-Fender | _pepo_: http://cepstral.com/ |
| 05:40.51 | _pepo_ | there is not the cost (now) |
| 05:40.53 | mpls-eric | Cepstral store, with prices: http://www.google.com/url?q=http://cepstral.com/store |
| 05:41.45 | [TK]D-Fender | _pepo_: You aren't even trying |
| 05:42.15 | mpls-eric | $29.99 for one channel, (for the lazy folks) (most voices) |
| 05:42.40 | _pepo_ | a lot of thanks |
| 05:43.57 | ChannelZ | I like going to the demo page and putting in dirty things for them to say |
| 05:45.17 | _pepo_ | :) |
| 05:45.49 | _pepo_ | do you work with TTS in Asterisk? which one? |
| 05:53.45 | ChannelZ | I don't |
| 05:54.07 | ChannelZ | although I have an old software demo of DECtalk that I use to make voicemail messages as Stephen Hawkings at home |
| 05:54.41 | *** join/#asterisk jtodd (i=flq7nvp4@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 05:54.41 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 06:30.04 | cjk | hi, i doing some stress tests and my calls from asterisk1 to asterisk2 get autocongested after 83 simultanioius calls. both servers have a load average below 1. any idea? |
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| 06:55.11 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
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| 07:02.49 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:10.10 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 07:17.34 | ChannelZ | cjk: what codecs are the calls using? |
| 07:20.12 | cjk | ChannelZ, i found the problem |
| 07:20.17 | cjk | too manby open files |
| 07:24.14 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 07:24.30 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 07:25.15 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 07:28.11 | *** join/#asterisk jozza (n=chatzill@89-212-75-91.static.t-2.net) |
| 07:28.30 | jozza | is there anyone here that knows anything about ooh323? |
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| 07:37.10 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ac6a:ac08:a061:4489) |
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| 07:56.13 | Brooklyn | I want my agents (members of queues) to be able to place customers back into the queue. |
| 07:56.19 | Brooklyn | Anyone a good idea how to di this? |
| 07:56.24 | Brooklyn | *do |
| 07:59.09 | Brooklyn | Give the queues an extension or something? |
| 08:00.12 | Congzing | Hi everyone |
| 08:00.42 | Congzing | I want to make video conference on Asterisk |
| 08:01.00 | Congzing | I was try asterisk 1.4 and successful |
| 08:01.18 | Congzing | Now I want to upgrade to Asterisk 1.6 |
| 08:01.51 | Congzing | But Asterisk 1.6.0.15 and 1.6.1.6 doesn't support video |
| 08:02.29 | Congzing | I try Asterisk 1.6.2.0 - rc 3 and maked video call |
| 08:02.49 | Congzing | but can't make video conferecen with appKonference |
| 08:03.53 | Congzing | anyone a good idea how to do? |
| 08:16.44 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:18.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:18.51 | Congzing | anyone, pls help me |
| 08:19.52 | *** join/#asterisk defswork (n=andy@195.58.91.66) |
| 08:22.34 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 08:23.23 | Congzing | Asterisk Video Conference on version 1.6.xxxx who know about it? |
| 08:33.48 | *** join/#asterisk ltd (n=z@202.55.149.246) |
| 08:44.36 | *** join/#asterisk stefan1234 (i=x@188.25.196.135) |
| 08:45.08 | stefan1234 | anyone online ? i need a little help :) |
| 08:45.40 | frek818 | What's up? |
| 08:46.30 | stefan1234 | frek818, tell me becvause i'm kinda dumb :). how can i open a new port and forward it to a new ip address? |
| 08:46.49 | Maliuta | go read the manual on your "router |
| 08:46.51 | Maliuta | " |
| 08:47.23 | stefan1234 | Maliuta thanks for the hint :) |
| 08:53.01 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:53.59 | *** part/#asterisk stefan1234 (i=x@188.25.196.135) |
| 09:02.21 | *** join/#asterisk hensema (n=erik@scrat.hensema.net) |
| 09:02.27 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 09:02.57 | hensema | good morning |
| 09:03.02 | *** join/#asterisk Congzing (n=Congzing@125.235.3.158) |
| 09:03.37 | hensema | has trouble getting my linksys spa941 to send a correct outgoing caller-id on internal (941 -> asterisk -> 941) calls |
| 09:04.10 | hensema | asterisk doesn't seem to know the correct internal number of the caller, and the callee can't call back since he only sees the display name which is a simple string |
| 09:04.32 | *** join/#asterisk qu1ckkkk (i=c419fffa@gateway/web/freenode/x-yjexdplmcoluswxc) |
| 09:05.03 | frek818 | You can set the caller ID in sip.conf |
| 09:05.21 | frek818 | callerid=Blah <9123> |
| 09:05.39 | qu1ckkkk | callerid= |
| 09:05.50 | qu1ckkkk | like frek818 said |
| 09:07.03 | qu1ckkkk | or in dialplan : exten => _X.,1,Set(CALLERID(num)=XXXXXXXXXX) |
| 09:08.27 | *** join/#asterisk Dibbler (n=Dibbler@87.194.103.72) |
| 09:09.29 | hensema | my userid seems to be Erik Hensema <asterisk>, even though I set it to $myname <123> in sip.conf :s |
| 09:14.25 | TSM | just about to start programming my users into *, im cut between a 4 or 5 digit number, this will have to span multiple offices in the future, each office will have their own * box and thus having their own park numbers etc.. |
| 09:18.55 | *** join/#asterisk Congzing (n=Congzing@125.235.3.158) |
| 09:22.16 | *** join/#asterisk qu1ckkkk (i=c636caea@gateway/web/freenode/x-hxwkevvsrpcnxeuc) |
| 09:23.03 | qu1ckkkk | when i reload for e.g. extentions or sip its fast, but reload iax2 takes mush longer, any ideas whay this could be? |
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| 09:33.02 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 09:35.06 | qu1ckkkk | what does this warning mean ? "chan_iax2.c:6968 socket_process: Received trunked frame before first full voice frame" |
| 09:35.40 | *** join/#asterisk kmate (n=mkoch@195.70.53.252) |
| 09:35.44 | kmate | hello all |
| 09:36.24 | qu1ckkkk | hey |
| 09:37.44 | kmate | is it possible to create adhoc conference rooms with MeetMe? For example someone calls 98765 and a new room comes into being, so thereafter anyone who calls the same number will join the conference? |
| 09:41.40 | kmate | and of course I want to have it for example with any 5-digit number, without adding 99999 lines to meetme.conf |
| 09:43.06 | *** join/#asterisk Da-Geek (n=Da-Geek@ip-217-204-101-67.easynet.co.uk) |
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| 09:57.24 | *** join/#asterisk soman (n=somnath@115.119.41.110) |
| 09:57.42 | soman | ChannelZ: hi |
| 10:00.31 | soman | i am using asterisk 1.2, and from the agi script i would like to get the number of participants in the conference using MeetMeCount, any ideas on how to get the meet me count |
| 10:04.50 | *** join/#asterisk af_ (n=getsmart@88.149.240.149) |
| 10:08.56 | soman | I tried $AGI->exec('MeetMeCount','$conf_num|count'); |
| 10:09.19 | soman | and $AGI->get_variable(count); |
| 10:09.21 | soman | no luck |
| 10:14.04 | *** part/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 10:20.56 | *** join/#asterisk Yuda-israel1984 (n=Yuda-isr@94.159.129.218) |
| 10:22.02 | Yuda-israel1984 | hi guys i am in need for help with getting festival running |
| 10:22.41 | Yuda-israel1984 | i did everything written on different sites that explain how to compile it and add sscript that would let asterisk use it as well although im stuck because it just wont work |
| 10:23.01 | Yuda-israel1984 | anyone here that can help me??? |
| 10:23.38 | tzafrir_laptop | Saying "___ doesn't work" doesn't really help. Please describe what does happen. |
| 10:23.47 | tzafrir_laptop | What did you do? What happens? |
| 10:24.08 | *** join/#asterisk mikkel (n=mikkel@eye.exoro.dk) |
| 10:24.12 | Yuda-israel1984 | hey tzafrir |
| 10:24.14 | Yuda-israel1984 | ma kore |
| 10:24.19 | Yuda-israel1984 | choref bari |
| 10:24.45 | Yuda-israel1984 | whats the israeli channel of asterisk |
| 10:26.10 | Yuda-israel1984 | so i compiled festival and i added script i think it was like tts_textasterisk and the rest |
| 10:26.24 | Yuda-israel1984 | i typed in festival --server and it says connected |
| 10:26.57 | Yuda-israel1984 | then when i dial the extension running it it says connect from local host and then dissconnect but doesnt play anything to the softphone |
| 10:28.36 | *** join/#asterisk Congzing (n=Congzing@125.235.3.158) |
| 10:29.55 | Yuda-israel1984 | http://pastebin.com/m16ba27a6 |
| 10:30.09 | Yuda-israel1984 | thats to the CLI and festival server at the same time |
| 10:30.21 | Yuda-israel1984 | in the softphone i get absolutly nothing |
| 10:30.36 | tzafrir_laptop | Yuda-israel1984, you mean: #asterisk-il ? |
| 10:43.48 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 11:06.28 | *** join/#asterisk Winkie (n=urmom@87.194.109.4) |
| 11:08.11 | *** join/#asterisk Soeron (n=soeron@mail.nordmetals.com) |
| 11:09.42 | Soeron | Good day people. Is there any order for asking questions ? |
| 11:11.47 | tzafrir_laptop | ~ask |
| 11:11.47 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:12.11 | tzafrir_laptop | Soeron, yes, there is: |
| 11:12.23 | tzafrir_laptop | If you never ask, you'll never get an answer |
| 11:12.47 | Soeron | yea thanks :) |
| 11:12.56 | Soeron | got some kind of "problem" |
| 11:13.12 | Soeron | have a setuped asterisk 1.6.1.1 |
| 11:13.26 | Soeron | and it should receive faxes |
| 11:13.33 | Soeron | got few extenstions that do the work |
| 11:13.48 | Soeron | but on one of the port of the pbx is attached gsm gateway |
| 11:13.59 | Soeron | and this port should act as fax machine |
| 11:14.17 | Soeron | but for some reason it 'should' send its fax number |
| 11:14.26 | Soeron | atleast i was told so by the support of the gsm gateway |
| 11:14.41 | Soeron | and i think i am lost |
| 11:14.57 | Soeron | triesd to set localstationid in the receivefax function |
| 11:15.23 | Soeron | but always get error Disconnected after permit tries |
| 11:15.35 | Soeron | seems both machines can't understand eachother |
| 11:15.50 | Soeron | the other machine is regular fax machine |
| 11:15.54 | Soeron | with gsm gateway too |
| 11:15.55 | kmate | is it possible to create adhoc conference rooms with MeetMe? For example someone calls 98765 and a new room comes into being, so thereafter anyone who calls the same number will join the conference? |
| 11:18.47 | Congzing | No, I think so |
| 11:20.04 | Congzing | That |
| 11:20.25 | Congzing | that's the thing I want to ask everyone |
| 11:20.40 | Congzing | When I call same number room |
| 11:21.09 | Congzing | the new people will kick old people in room |
| 11:21.27 | Congzing | and I don't know how make conference with meetme |
| 11:22.37 | Congzing | I try with asterisk 1.6.2.0 - rc3 and appkonference then I can't make video conference call |
| 11:23.40 | Congzing | someone pls help I want to make video conference call with asterisk version 1.6 |
| 11:28.15 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 11:35.42 | *** join/#asterisk d00gster (n=doughant@94.97.126.216) |
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| 11:44.00 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-kriltenaptrtyypk) |
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| 11:52.02 | *** join/#asterisk Da-Geek (n=Da-Geek@ip-217-204-101-67.easynet.co.uk) |
| 11:54.41 | war9407 | after an upgrade from 1.4 to 1.6 in Debian I get this |
| 11:54.43 | war9407 | load_module: Unable to register channel class 'Console' |
| 11:54.54 | war9407 | [Oct 12 07:51:16] WARNING[24402]: channel.c:564 ast_channel_register: Already have a handler for type 'Console' && [Oct 12 07:51:16] ERROR[24402]: chan_alsa.c:907 load_module: Unable to register channel class 'Console' |
| 11:55.55 | *** join/#asterisk pyite_mac (n=dschreib@unaffiliated/pyite) |
| 11:56.30 | tzafrir_laptop | war9407, Console is provided by one of the following: chan_oss, chan_alsa, chan_console |
| 11:56.42 | tzafrir_laptop | (chan_console uses pulseaudio) |
| 11:57.18 | tzafrir_laptop | I guess chan_alsa came in second after chan_console or chan_oss already loaded |
| 11:57.44 | war9407 | yea, put them as noload, seems to work now, thx |
| 11:57.54 | tzafrir_laptop | this is basically a harmless message. but you can disable the module(s) you want to avoid in modules.conf |
| 11:58.18 | war9407 | it was stopping asterisk from starting actually (going to call and see if it works now)-- |
| 12:13.25 | *** join/#asterisk garymc (n=garymc@host86-159-106-40.range86-159.btcentralplus.com) |
| 12:16.42 | Naikrovek | oh i love my Indian coworkers. "did you got the chance to workings on?" |
| 12:16.53 | Naikrovek | what the FFFF are you asking me |
| 12:17.22 | *** join/#asterisk asif (n=chatzill@122.166.40.72) |
| 12:19.13 | asif | hello all. |
| 12:19.21 | Naikrovek | oh awesome. i ask him for clarification: "did you got the chances to the issue workings on?" much easier to understand when they call me somehow. |
| 12:20.43 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:21.12 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 12:23.57 | qu1ckkkk | lol |
| 12:24.24 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:24.43 | ariel_ | Morning |
| 12:25.19 | asif | I'm using call files to place calls. I'm not getting duration and billsecs for unanswered / failed calls. any idea why? |
| 12:32.11 | *** join/#asterisk Lantizia (n=stevem@212.57.232.254) |
| 12:33.11 | Lantizia | Lo, can someone tell me if this would fail miserably... Linksys SPA3102 as a PSTN to SIP Trunk gateway... Fax comes in... goes over network as SIP... some app on the asterisk box turns into email? |
| 12:33.35 | Lantizia | I hear it probably will as the RTP stream won't have the right quality to maintain the fax... this true? |
| 12:36.03 | ariel_ | Lantizia: it should work. RTP stream does travel over your network. It's always good to have it on a good switch. Biggest issue is more like normal issues with fax machines as they are not all supported. |
| 12:37.00 | Lantizia | ariel_, so is there some preferred codec to maintain the quality of the original fax? |
| 12:37.05 | Lantizia | would g711u do? |
| 12:38.52 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 12:41.27 | ariel_ | that is about the only one you can use ulaw for us and maybe alaw if your using e1's |
| 12:41.28 | *** join/#asterisk dwayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 12:41.59 | Lantizia | ariel_, e1's!? |
| 12:44.38 | ariel_ | In the US we use PRI/T1's which native works well with ulaw, Over seas they use E1's which use alaw. |
| 12:46.28 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 12:47.24 | kmate | is it possible to create adhoc conference rooms with MeetMe? For example someone calls 98765 and a new room comes into being, so thereafter anyone who calls the same number will join the conference? |
| 12:49.56 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 12:50.33 | shido6 | sure u can |
| 12:50.46 | kmate | how? |
| 12:53.13 | kmate | i mean without adding 99999 lines to meetme.conf to define all 5-digit conference number |
| 12:55.28 | kmate | shido6: ^^ |
| 12:55.31 | Lantizia | ariel_, well this isn't ISDN in anyway... this is a POTS/PSTN converted to SIP using an ATA |
| 12:56.22 | shido6 | you can build a custom script to do that for you |
| 12:56.34 | *** join/#asterisk odenkos (n=odenkos@212.81.19.170) |
| 12:58.25 | *** join/#asterisk zigond (n=max@80-240-220-48.dnat.migtel.ru) |
| 13:00.10 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 13:03.43 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 13:04.52 | *** join/#asterisk fmerges (n=fmerges@178.Red-83-36-57.staticIP.rima-tde.net) |
| 13:04.54 | fmerges | hi all |
| 13:05.39 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 13:05.51 | fmerges | quick question, with the demo/ or just running an extension with the echo, it works, but it stays busy when i hangup and in the var log messages i get this: chan_dahdi.c: MWI: Channel 1 message waiting! |
| 13:06.07 | fmerges | i have callerid deactivated... any idea? |
| 13:06.20 | fmerges | and no mailboxes defined... |
| 13:06.38 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 13:09.48 | *** join/#asterisk sezuan (n=sezuan@drogo.vp.ip6.scheff32.de) |
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| 13:15.17 | shido6 | whats the debug look like |
| 13:15.23 | shido6 | pastebin some debug |
| 13:16.55 | *** join/#asterisk psilikon (n=psilikon@static-173-65-4-24.tampfl.fios.verizon.net) |
| 13:19.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 13:21.22 | *** part/#asterisk torrancew (n=chatzill@70.177.34.83) |
| 13:25.37 | fmerges | shido6: http://pastebin.com/d7a543552 |
| 13:26.01 | manxpower | ~answers |
| 13:26.02 | infobot | from memory, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:27.36 | *** join/#asterisk qu1ckkkk (i=c419fffa@gateway/web/freenode/x-qthqfpkowudccnfx) |
| 13:29.53 | *** join/#asterisk DavidR2008 (n=chatzill@208.34.240.2) |
| 13:30.22 | *** join/#asterisk btsteve (n=tstevens@204.10.20.30) |
| 13:41.27 | *** join/#asterisk af_ (n=getsmart@88-149-240-149.dynamic.ngi.it) |
| 13:43.57 | DavidR2008 | has there been a change in dialplan math between 1.4 & 1.6? For example b=$[${year} / 100] where year = 2009 is 20 on 1.4 and 20.09 on 1.6 |
| 13:44.41 | manxpower | DavidR2008: all major changes are documented in UPGRADE*.txt in the Asterisk source code. |
| 13:46.32 | manxpower | BTW, there is really no "1.6". 1.6.0, 1.6.1, and 1.6.2 are all considered "significant releases" with potential incompatible changes between those releases. |
| 13:47.26 | DavidR2008 | are there separate UPGRADE files for the different 1.6 versions? |
| 13:47.31 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 13:48.58 | mbrevda | does asterisk have a problem offering g729 if it have the codec installed? (i.e. if asterisk is being used as a router) |
| 13:49.10 | mbrevda | *if the codes is not installed |
| 13:49.24 | Naikrovek | mbrevda: passthru calls do not use nor require a license |
| 13:49.40 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:49.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:50.31 | *** part/#asterisk Shinsaku (n=Shinsaku@chello089076140236.chello.pl) |
| 13:51.11 | mbrevda | I know, but is there a reason it shouldnt be offered? |
| 13:51.22 | Naikrovek | offered to what, a phone? |
| 13:51.31 | mbrevda | or a trunk |
| 13:52.04 | Naikrovek | yes, it won't be offered to a trunk I don't think without being licensed. |
| 13:52.32 | mbrevda | here's the thing: its only needed for passthru. Meaning, it wont passthru without a lisense? |
| 13:52.41 | mbrevda | I'm sure that not how its 'meant' to be |
| 13:53.14 | Naikrovek | it should pass through without a license |
| 13:53.55 | Naikrovek | if you compiled support for g729 into asterisk it should at least say that it supports the codec, unless you have "disallow=all, allow=ulaw" or something like that in the trunk config |
| 13:56.50 | mbrevda | is this where the remote unit tell us which codec it supports? |
| 13:56.50 | mbrevda | a=rtpmap:0 pcmu/8000 |
| 13:57.55 | Naikrovek | yup |
| 13:58.03 | Naikrovek | then it will have an order of preference |
| 13:58.09 | Naikrovek | yours will just be 0 |
| 13:58.19 | mbrevda | mine? my remote? |
| 13:58.30 | Naikrovek | whatever your looking at |
| 13:58.30 | manxpower | DavidR2008: there should be. check your source |
| 13:58.41 | *** join/#asterisk Drasha (n=chatzill@195.113.224.147) |
| 13:58.50 | manxpower | generally g729 passthru is much less useful than you might thing. |
| 13:58.55 | Naikrovek | yeah |
| 13:59.17 | manxpower | you don't "compile g729 support into Asterisk", you install the binary codec and pay the license fee. |
| 13:59.39 | Naikrovek | well you can straight up remove support within asterisk if you make the choice in 'make menuselect' |
| 13:59.56 | Naikrovek | whether the codec is present or not |
| 14:00.10 | manxpower | that only sets support for passthru, i.e. the file format, not the actual codec. |
| 14:00.26 | Naikrovek | yeah - i misspoke but that's what i meant |
| 14:01.42 | Naikrovek | there's no reason that passthrough won't work so long as it's not forbidden in the phone and/or trunk configs |
| 14:02.09 | Drasha | Hi, everyone. I have a question you might answer. Suppose I have two phones A and B, Asterisk and Alcatel PBX. |
| 14:02.10 | Drasha | If the call goes like this: [A <-- * <-- Alcatel <-- B], everything is allright. |
| 14:02.12 | Drasha | But when the call goes like this: [A --> * --> Alcatel --> B], B can hear A, but A can't hear B. The audio stream stops at Asterisk. |
| 14:02.13 | Drasha | Do you have any idea what might be causing this? It is a communication via SIP |
| 14:02.35 | Naikrovek | NAT |
| 14:02.58 | manxpower | Naikrovek: you can't use the T/t/w/W options to Dial, you can't year asteirsk prompts (including voicemail) unless you install the native G729 prompts and hold music, you can't use ChanSpy, you can't play IVRs without the ivr sounds already being in G729. |
| 14:03.19 | Naikrovek | manxpower: oh wow |
| 14:03.33 | manxpower | Drasha: set canreinvite=no for the Alcatel peer in sip.conf |
| 14:03.38 | Naikrovek | i thought if you installed the g729 prompts that one could hear that |
| 14:03.56 | Naikrovek | ah again with the misreading on my part |
| 14:03.57 | Naikrovek | wtf |
| 14:04.02 | Naikrovek | this headache is really fucking me up |
| 14:04.05 | manxpower | "unless you install the native G729 prompts and hold music" |
| 14:04.09 | Drasha | @Naikrovek: Both Asterisk and phone A are on the same network |
| 14:04.10 | Naikrovek | yeah sorry |
| 14:04.14 | Naikrovek | misread |
| 14:04.31 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 14:04.55 | Drasha | @manxpower: I have it in [general] section and alcatel peer does not have it set to yes. Is it enough? |
| 14:05.00 | Naikrovek | Drasha: A, Asterisk, and Alcatel all on the same network |
| 14:05.10 | Naikrovek | ? |
| 14:05.13 | manxpower | Drasha: that should be enough. |
| 14:05.36 | manxpower | Drasha: make sure you don't have any "bindport" or "bindip" options in sip.conf |
| 14:05.46 | manxpower | or even port= lines. remove those too |
| 14:06.02 | Drasha | Asterisk machine has two NIC. A and Asterisk are in the same local network, Asterisk and Alcatel communicate via internet |
| 14:06.08 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 14:06.46 | Naikrovek | Drasha: and the phone A and Asterisk have private IP addresses? |
| 14:07.11 | Drasha | They are both in 192.168.32.0/24 net |
| 14:07.30 | Naikrovek | Drasha: so why did you immediately dismiss my "NAT" diagnosis |
| 14:07.55 | shido6 | ~sipnat |
| 14:07.56 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:08.27 | Naikrovek | one-way audio is almost always NAT |
| 14:09.33 | Drasha | Did not mean to sound like immediate dismiss - I just thought that when Asterisk is bridging the audio, it creates two separate streams - one two phone A and one to alcatel and that neither is NATed |
| 14:09.51 | *** join/#asterisk xrmx__ (n=rm@host27-26-dynamic.24-79-r.retail.telecomitalia.it) |
| 14:10.40 | Naikrovek | so asterisk has a public IP aswell |
| 14:10.44 | Naikrovek | as well* |
| 14:10.49 | Drasha | yes |
| 14:10.53 | Naikrovek | far out i've never had a headache like this |
| 14:10.57 | Naikrovek | i can't go any further |
| 14:11.16 | Naikrovek | soon [tk]d-fender will show up, if you don't get help before he comes, he'll help you |
| 14:11.37 | verywiseman | how can i know which codecs asterisk 1.6 can support? in asterisk 1.4 there is sow codecs |
| 14:11.39 | shido6 | ? |
| 14:11.52 | Drasha | ok, thanks Naikrovek anyway |
| 14:11.56 | Naikrovek | np |
| 14:12.05 | Naikrovek | verywiseman: core show translation |
| 14:12.11 | shido6 | does the phone ring, Drasha ? |
| 14:12.35 | Naikrovek | shido6: he gets two-way audio on a call incoming to asterisk, one way audio calling out |
| 14:12.43 | Drasha | @shido6, the call gets connected. Only the audio in one direction stops at Asterisk |
| 14:12.49 | Drasha | right |
| 14:13.04 | shido6 | what does wireshark say about the call when you compare the two types of calls? |
| 14:14.20 | Drasha | that the Asterisk send only the ringing tone to the phone and once the call is connected, no audio is transferred to the phone |
| 14:14.39 | manxpower | sounds to me like a firewall problem |
| 14:14.50 | Naikrovek | yeah |
| 14:15.12 | manxpower | remember one of those audio streams in incoming from the Alcatel, which a firewall or iptables could easily default to rejecting |
| 14:15.36 | Drasha | I thought so, there were some changes on firewall that dates to almost the same time since we've been aware of this problem. But we have checked the fw and it seems ok |
| 14:15.46 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:16.08 | Drasha | if I do a capture with wireshark, I only get what passes the fw, right? |
| 14:16.24 | manxpower | Audio arrives on ports 10,000 - 20,000 UDP (can be changed in /etc/asterisk/rtp.conf) |
| 14:16.54 | *** join/#asterisk titoyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 14:17.06 | *** join/#asterisk brad_mssw (n=brad@216.155.101.90) |
| 14:17.29 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:17.49 | Drasha | in rtp.conf I have ports 20000 - 20500. And fw is set to accept 20000 - 50000 |
| 14:18.00 | Drasha | this is a real headache |
| 14:20.02 | UQlev | Drasha: use IAX2 then only 1 port 4569 |
| 14:20.26 | manxpower | IAX2 never solved any problem |
| 14:20.30 | Drasha | :-) |
| 14:20.34 | manxpower | Drasha: turn off firewall, test. Does it work? |
| 14:20.55 | Drasha | also the Alcatel PBX does not support IAX2 AFAIK |
| 14:21.09 | Drasha | @manxpower - will try it |
| 14:21.13 | UQlev | Drasha: or run your asterisk on the interface with reaql public IP |
| 14:21.14 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:21.14 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:21.25 | Drasha | it has real public ip |
| 14:21.25 | *** join/#asterisk umay (n=chris@174-16-31-61.hlrn.qwest.net) |
| 14:21.28 | Naikrovek | UQlev: he is |
| 14:21.33 | manxpower | Drasha: did you make sure you have no bindip and port lines in sip.conf? |
| 14:21.55 | Drasha | only port= line I have is in [general] section and there is port=5060 |
| 14:22.06 | manxpower | Drasha: that should be OK. |
| 14:23.04 | Drasha | I'm going for fw off test - be back in short time |
| 14:23.39 | Soeron | someone using a Vierling Ecotel GSM Gateway with asterisk? |
| 14:24.28 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:24.55 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 14:27.08 | Drasha | hmm... sometimes you can't trust what you see... although the fw settings looked good at first glance, the fw off test has proven me wrong. So it is firewall issue... thatnk you all for cooperation |
| 14:27.56 | shido6 | :) |
| 14:28.06 | manxpower | <-- smarter than he looks |
| 14:28.08 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 14:28.19 | Drasha | :-D |
| 14:29.46 | Drasha | bows to manxpower for his precise insight |
| 14:32.25 | DavidR2008 | is anyone having problems getting to www.asterisk.org? I keep getting a connection reset error, wanted to know if it's just me |
| 14:33.28 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 14:33.37 | zigond | i have "connectioon timed out" from www.asterisk.org |
| 14:34.11 | DavidR2008 | thanks |
| 14:34.58 | zigond | is there a way for asterisk to process only limited number of .call files at a time? |
| 14:35.23 | zigond | for ex.: I have 10 .call files in /var/spool/asterisk/outgoing/ , but asterisk m |
| 14:35.27 | zigond | ust do not more than 2 files at any given moment. |
| 14:36.18 | Naikrovek | keeps mistyping RedHat as RedHATE in email today. thank you freud. |
| 14:36.43 | DavidR2008 | I setup an outbound schedule on 1.4 and at that time, it was stated that the program that wrote the .call files needed to be responsible for knowing how many .call files already existed and what the limit should be. |
| 14:36.57 | DavidR2008 | that's a long way of saying, not that I know of. ;-) |
| 14:38.09 | manxpower | zigond: no. However, if the creation time of the file is in the future Asterisk will not process that file until that time. |
| 14:38.38 | zigond | thank you |
| 14:41.28 | DavidR2008 | can anyone point me to where the expr language is documented for 1.6? |
| 14:41.47 | manxpower | DavidR2008: channelvariables.txt |
| 14:42.15 | zigond | how can i limit number of simultaneous calls in a way that every .call files will be processed as soon as call-limit allows this? |
| 14:42.23 | TSM | is there any way to send text messages directly to polycom phones, ive been looking in the docs and its not clear |
| 14:42.25 | DavidR2008 | doc/tex/channelvaribles.tex? |
| 14:42.31 | manxpower | zigond: you cannot. |
| 14:43.08 | manxpower | DavidR2008: They turned many of the doc files into .tex format. I guess the Asterisk developers thing .txt format is too easy to use. |
| 14:43.42 | DavidR2008 | ok, thanks |
| 14:44.18 | *** join/#asterisk xrmx__ (n=rm@host27-26-dynamic.24-79-r.retail.telecomitalia.it) |
| 14:44.34 | zigond | thank you again, manxpower |
| 14:46.21 | mbrevda | anyone famiar with audiocodes mp-1xx? I'm trying to set up fxo trunk with registration. However If I use 'by dest number' in hunt groups I cannot make any calls |
| 14:46.23 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 14:46.56 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-68-233.link.net.pk) |
| 14:48.51 | mbrevda | here is some debug: http://pastebin.ca/1615512 |
| 14:48.59 | *** join/#asterisk jmacz (n=jmacz@186.81.61.74) |
| 14:52.23 | *** join/#asterisk bzing2 (n=dr105@194.66.208.236) |
| 14:54.25 | *** part/#asterisk bzing2 (n=dr105@194.66.208.236) |
| 14:54.29 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:55.10 | *** join/#asterisk bzing2 (n=dr105@194.66.208.236) |
| 14:58.50 | SuPrSluG | mbrevda: endpoint settings is where you authenticate |
| 15:00.40 | mbrevda | SuPrSluG: I (think im) authenticated. Now im trying to get the call routed based on the trunk I send it too. I have 5001-3 registered. If I send a call to 5001 and its busy, it goes to 5002 if I send a call to 5002 it gos out fxo 1. I want each fxo to work in completly solidaraly from the others, with only asterisk handeling the failvoer |
| 15:03.21 | SuPrSluG | mbrevda: pastebin your ini file |
| 15:03.33 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 15:08.41 | manxpower | mbrevda: It sounds like you are using a GUI |
| 15:09.51 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 15:10.03 | manxpower | ~amp |
| 15:10.04 | infobot | well, amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 15:10.06 | manxpower | ~freePBX |
| 15:10.07 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:11.02 | manxpower | 8985551212 is not found on the destination device, based on your debug info |
| 15:13.01 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:17.26 | *** join/#asterisk CreatixEA (n=chatzill@creanet.creatixea.com) |
| 15:18.50 | CreatixEA | Hi, got TrixBox 2.4, SCCP with Cisco 7920 configured well except for speeddials ... Don't understand the sccp.conf syntax for speeddialing : there's no documentation :( |
| 15:19.31 | manxpower | ~mailinglist |
| 15:19.32 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 15:20.45 | manxpower | The sccp docs that I know of are the sccp.conf sample config files. Other than that try searching the mailinglists. |
| 15:20.53 | *** join/#asterisk faizan (n=Tili@115.167.82.126) |
| 15:21.05 | manxpower | very, very few people use SCCP/Skinny with Asterisk |
| 15:21.10 | faizan | how can I login ss7 traces to a file from asterisk? |
| 15:21.38 | CreatixEA | Thanks manxpower :) |
| 15:21.43 | manxpower | faizan: all logging is configured in /etc/asterisk/logger.conf |
| 15:21.57 | faizan | manxpower: thanks. I will look |
| 15:23.29 | faizan | manxpower: just checked. I dont think logger.conf can help me in writing ss7 traces to a file from a particular linkset |
| 15:24.55 | mbrevda | manxpower: I say that - the question is why! |
| 15:25.05 | mbrevda | SuPrSluG: http://pastebin.ca/1615600 |
| 15:25.09 | mbrevda | *saw |
| 15:26.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:26.54 | SuPrSluG | mbrevda: also check your dialplan , are you putting a 1 before the 10 digit number? |
| 15:27.15 | mbrevda | SuPrSluG: no. why? |
| 15:27.46 | SuPrSluG | well how can you dial a us number w/out it |
| 15:28.30 | SuPrSluG | # |
| 15:28.32 | SuPrSluG | To: <sip:8985551212@192.168.0.166>;tag=1c1070447965 |
| 15:29.05 | SuPrSluG | shouldn't it be 18985551212 ? |
| 15:29.30 | mbrevda | SuPrSluG: no. besides its an ANALOG trunk, so the truk wouldnt know the difference |
| 15:29.54 | mbrevda | and I dont belive the audocodes is programed anywhere for a specific dialplan |
| 15:30.08 | *** join/#asterisk atis_home (n=atis@193.238.213.215) |
| 15:30.34 | mbrevda | (and im trying to keep ALL routing in asterisk. I want to use the AC as a simple dumb sip-> fxo bride) |
| 15:32.56 | SuPrSluG | mbrevda: how is your hunt group set up? |
| 15:33.56 | mbrevda | i out each fxo in its own group. if I set the groups to Channel Select Mode=ascending the calls go thru. but with zero respect for the trunk they were sent to |
| 15:34.10 | mbrevda | (i.e. 5002 can go thru fxo 1) |
| 15:43.34 | *** join/#asterisk voipmonk (n=voipmonk@74.132.202.71) |
| 15:43.50 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:44.49 | bmoraca | does the Transfer() command in 1.4 work on Zap channels? |
| 15:46.29 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 15:49.42 | manxpower | bmoraca: If it does, it won't do what you expect it to. |
| 15:49.51 | bmoraca | i didn't suspect so |
| 15:50.00 | raden_work | omfg my boss is complaining cause the procurve at my desk has a slight whine in the fan :( |
| 15:50.18 | voipmonk | getting a new fan or a new procurve? |
| 15:50.19 | voipmonk | :) |
| 15:50.35 | bmoraca | i wonder if it's possible to transfer calls out a PRI without using multiple B channels...i don't think it is, is it? |
| 15:51.55 | raden_work | voipmonk, call hp get a new fan at my expense cause it my switch :( |
| 15:51.59 | raden_work | I hate people |
| 15:52.19 | voipmonk | get a louder fan |
| 15:52.22 | voipmonk | :) |
| 15:52.28 | raden_work | says he can hear it in his office 50 ft away with his dorr closed |
| 15:52.30 | voipmonk | that hums, "F-off" |
| 15:52.40 | mbrevda | SuPrSluG: ^^ |
| 15:53.13 | mbrevda | bmoraca: no, its not |
| 15:53.26 | [TK]D-Fender | bmoraca: Certainly is. |
| 15:53.43 | bmoraca | [TK]D-Fender: any clues as to the direction I should go? |
| 15:53.59 | [TK]D-Fender | bmoraca: DAHDI/5 <- there |
| 15:54.09 | SuPrSluG | mbrevda: sorry i had work to do. you f-in douchebag |
| 15:54.12 | [TK]D-Fender | bmoraca: Oh wait... TRANSER... |
| 15:54.13 | [TK]D-Fender | ~2bct |
| 15:54.22 | *** join/#asterisk wcselby (n=wcselby@216-110-88-194.static.twtelecom.net) |
| 15:54.28 | [TK]D-Fender | bmoraca: "transfer=yes" in your channel def |
| 15:54.51 | manxpower | bmoraca: the term is "2BCT", try searching on that. |
| 15:54.51 | [TK]D-Fender | bmoraca: Has to be supported by facilities |
| 15:54.51 | mbrevda | SuPrSluG: no need to be sorry, take your time |
| 15:54.53 | [TK]D-Fender | bmoraca: but Zap/DAHDI support it |
| 15:54.55 | bmoraca | ok, i'll give it a shot |
| 15:55.10 | [TK]D-Fender | bmoraca: It'll do it on native bridge automatically if enabled & supported |
| 15:55.17 | manxpower | in 1.4 the switches the 2BCT support in Asterisk is very limited. I don't know about 1.6 |
| 15:55.28 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:56.33 | *** join/#asterisk DND (n=DND@213.132.40.2) |
| 15:56.44 | bmoraca | i don't remember what kind of switch my telco has |
| 15:56.47 | p3nguin | When SIP looks up our own address (when we use NAT), can we see the result of that lookup with core show something? |
| 15:57.11 | DND | guys my asteriskbox is inside a local network, is it possible to port forward all necessary ports so that i can use it outside our office? |
| 15:57.44 | p3nguin | dnd: of course |
| 15:58.33 | p3nguin | dnd: If you're using NAT, you'll want to configure some settings in Asterisk, but as far as port forwarding, just forward what you need. |
| 15:59.01 | [TK]D-Fender | ~sipnat |
| 15:59.01 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:59.03 | [TK]D-Fender | DND ^^^ |
| 16:01.10 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
| 16:03.43 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:ac6a:ac08:a061:4489) |
| 16:06.30 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 16:09.34 | Naikrovek | raden_work: you gotta find new work buddy |
| 16:09.49 | raden_work | Naikrovek, yes i do |
| 16:10.04 | raden_work | Naikrovek, trying to take more on road service so i not in the office ever ! |
| 16:10.09 | Naikrovek | boss says fan is loud 50 ft away, you say "how do you think i feel? do you think I'm deaf, you knowb?" |
| 16:10.14 | raden_work | and trying to find a new place to live cause i have had it with my GF |
| 16:10.15 | Naikrovek | knob?" |
| 16:10.24 | Naikrovek | zoinks |
| 16:10.29 | raden_work | Naikrovek, all i can hear is a suttle fan hum |
| 16:10.30 | Naikrovek | life is not good for mr raden |
| 16:10.51 | Naikrovek | howcome she don't have to find a new place to live |
| 16:11.05 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:11.07 | Naikrovek | you guys got kids? if not, drop kick her |
| 16:11.16 | p3nguin | Does the resultant value of externhost='s lookup get stored somewhere so I can check it on the console? |
| 16:11.26 | raden_work | Naikrovek, we have a daughter together |
| 16:11.30 | raden_work | I have full custody |
| 16:11.36 | Naikrovek | ooh, ooh |
| 16:11.52 | raden_work | Naikrovek, i want a smaller place we have a big place cause she has 3 kids from previous marriage |
| 16:12.01 | Naikrovek | oh holy cow |
| 16:12.14 | Naikrovek | i'd go apeshit in a house with 4 kids |
| 16:12.23 | Naikrovek | though to a degree they can kinda keep themselves busy |
| 16:12.36 | [TK]D-Fender | p3nguin: "sip show settings" |
| 16:12.43 | Naikrovek | my lone daughter is always "daddy daddy daddy daddy daddy daddy daddy daddy daddy daddy... etc." |
| 16:12.57 | [TK]D-Fender | p3nguin: Should be there |
| 16:13.04 | Naikrovek | i don't spank her so she of course prefers to spend her time with me |
| 16:14.08 | raden_work | Naikrovek, i work 60 hours a week she works 20-30 and im the one that does the dishes cleans everything and takes the kids all weekend when shes at work |
| 16:14.16 | raden_work | i moved out for 2 months and life was bliss |
| 16:14.21 | Naikrovek | we're a bit off topic now but i feel ya bro |
| 16:14.24 | raden_work | my daughter is even happier for god sakes |
| 16:14.29 | raden_work | lol |
| 16:15.08 | p3nguin | [tk]d-fender: Not listed. |
| 16:15.32 | [TK]D-Fender | p3nguin: 1st section on network settings should interpret it IIRC to the exdternip. |
| 16:15.44 | [TK]D-Fender | p3nguin: You do see the rest of the settings there, right? |
| 16:16.16 | p3nguin | The bindaddress and SIP Port are listed, if that's what you mean. |
| 16:16.32 | wcselby | [TK]D-Fender - just checked on my own box and externip isn't listed there |
| 16:16.37 | wcselby | using sip show settings |
| 16:17.23 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:19.53 | *** part/#asterisk Brooklyn (i=miracle@the.city.that.never-sleeps.us) |
| 16:20.54 | [TK]D-Fender | P3 what version? |
| 16:20.56 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 16:24.56 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:25.06 | *** join/#asterisk roe_ (n=roe___@216-164-215-125.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com) |
| 16:29.24 | Katty | puts hands over ears and cries. |
| 16:29.43 | Katty | make the noises stop :< |
| 16:30.11 | [TK]D-Fender | presses "Mute" |
| 16:31.18 | *** join/#asterisk jmacz (n=jmacz@186.81.61.74) |
| 16:33.53 | TSM | anyone used sangoma a200 cards, i dont get a dial tone on mine, but it has picked up the call |
| 16:34.45 | Katty | [TK]D-Fender: i got on the phone and started crying earlier for no particular reason too |
| 16:35.11 | manxpower | TSM: FXO ports of FXS ports? |
| 16:35.29 | Katty | it was kind of embarassing |
| 16:35.41 | Katty | thank goodness it wasn't a client of mine >.< eesha |
| 16:36.02 | Katty | but the whole office keeps wandering in my office to see if i'm okay |
| 16:36.19 | TSM | manxpower: FXS |
| 16:37.28 | TSM | manxpower: wanpipe is setup, looked in * full log and nothing was being seen when line goes off hook |
| 16:38.21 | TSM | manxpower: im thinking mabey chan_dahdi is setup wrong |
| 16:39.20 | TSM | manxpower: i have a A101 card as well, its put my two fxs ports on channel=>25 & channel=>26 |
| 16:40.09 | *** join/#asterisk NewCom (i=Newcom@24.120.182.18) |
| 16:40.21 | Kobaz | TSM: paste dahdi system.conf config, wanpipe config, chan_dahdi.cfg... etc |
| 16:40.58 | Kobaz | TSM: remember fxs devices, get fxo signalling on the dahdi config |
| 16:41.10 | *** join/#asterisk wcselby (n=wcselby@216-110-88-194.static.twtelecom.net) |
| 16:41.22 | *** join/#asterisk xpot-mobile (n=james@173.8.94.1) |
| 16:41.51 | TSM | Kobaz:http://pastebin.com/m628a6188 |
| 16:41.58 | Deeewayne | bear hugs Katty |
| 16:44.03 | Kobaz | what's the output of: wanpipemon -i w2g1 -c astats -m 1 |
| 16:44.23 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 16:44.48 | TSM | Kobaz: TIP: -6v RING: -54 VBAT: -64 |
| 16:44.51 | NewCom | very inexperienced user looking for some help getting MP-114 to work with Trixbox correctly. I have it calling in and out, but I can't figure out how to have the MP-114 inform trixbox which line the call was coming in on...anyone here who can help? |
| 16:44.59 | Kobaz | k, that's good |
| 16:45.08 | Kobaz | NewCom: good luck |
| 16:45.19 | *** join/#asterisk wcselby (n=wcselby@216-110-88-194.static.twtelecom.net) |
| 16:45.31 | Kobaz | NewCom: i'm rmaing two mp-114's since they just don' |
| 16:45.35 | Kobaz | don't work properly |
| 16:46.02 | [TK]D-Fender | TSM so you pick up the handset and get no dialtone? |
| 16:46.05 | Kobaz | NewCom: but, for channel... in the automatic dialing section... set each phone number to match the channel |
| 16:46.23 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 16:46.41 | TSM | [TK]D-Fender: no dialtone, but i get talkback which shows its connected |
| 16:46.49 | Kobaz | TSM: do you get green lights on the back of the card? |
| 16:46.54 | TSM | Kobaz: yup |
| 16:47.02 | [TK]D-Fender | TSM: Did you gonnect the molex to the card? |
| 16:47.09 | [TK]D-Fender | connect* |
| 16:47.25 | TSM | [TK]D-Fender: yup, without it connected the system does not even see its there |
| 16:48.35 | [TK]D-Fender | TSM: PB your indications.conf and system.conf |
| 16:48.46 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 16:49.17 | Kobaz | he pasted some stuff |
| 16:49.37 | Kobaz | yeah but not system.conf |
| 16:50.00 | TSM | http://pastebin.com/m255eb73 |
| 16:50.20 | TSM | i dont have system.conf, its a freepbx gui, not sure if that makes a diffrence |
| 16:50.30 | TSM | i have also trimmed the indications.conf to only the UK stuff |
| 16:50.32 | [TK]D-Fender | TSM: DAHDI requires a system.conf <- |
| 16:50.35 | Kobaz | yes |
| 16:50.35 | Kobaz | haha |
| 16:50.40 | Kobaz | system.conf is required |
| 16:50.55 | TSM | is that for FXS ports as my PRI stuff works |
| 16:50.58 | Kobaz | /etc/dahdi/system.conf |
| 16:51.03 | TSM | ahh ok sec |
| 16:51.03 | [TK]D-Fender | ... |
| 16:51.04 | Kobaz | if you don't have one, dahdi won't work at all |
| 16:51.20 | TSM | http://pastebin.com/m3ef9547a |
| 16:51.30 | TSM | my dhadi has us set as the zones |
| 16:51.36 | TSM | should i change it to uk? |
| 16:51.47 | Kobaz | you have no channels defined in your system.conf |
| 16:53.01 | Kobaz | how does your pri possibly work |
| 16:53.45 | p3nguin | [tk]d-fender: Oh, you were talking to me? 1.4.24.1 |
| 16:54.07 | [TK]D-Fender | TSM: DUH <- :p |
| 16:54.16 | TSM | Kobaz: dont know, it just works |
| 16:54.32 | [TK]D-Fender | PRI doesn't care about audio indications |
| 16:54.44 | Kobaz | pri still needs the channels set up in dahdi/system |
| 16:55.00 | Kobaz | maybe they were set up, and the config got clobbered |
| 16:55.29 | Kobaz | anyways |
| 16:55.46 | tzafrir_laptop | TSM, what's the output of dahdi_hardware #? |
| 16:56.03 | TSM | pci:0000:21:04.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 16:56.03 | TSM | pci:0000:31:04.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 16:56.13 | TSM | sorry should have PB that |
| 16:56.16 | Kobaz | you'll need fxoks=25,26 in system.conf |
| 16:56.30 | Kobaz | and you |
| 16:56.36 | Kobaz | 'll need the t1 span config in there too |
| 16:57.19 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:57.33 | TSM | Kobaz: for pri what do i put, pri=1-23? |
| 16:57.52 | tzafrir_laptop | TSM, I suspect the '-' actually means "no drivers loaded", but I'm not completely sure for Sangoma |
| 16:57.53 | Kobaz | no |
| 16:58.07 | NewCom | Kobaz: thank you for the help, I set the number for the channel in the Caller Display Information and now it seems to work! |
| 16:58.27 | Kobaz | TSM: http://pastebin.com/m568d0965 |
| 16:58.45 | Kobaz | i havent configred an fxs/fxo in a while... it's either fxoks or fxo_ks |
| 17:01.09 | *** part/#asterisk NewCom (i=Newcom@24.120.182.18) |
| 17:01.19 | TSM | Kobaz: i just checked i think the settings are diffrent for UK PRI than US T1, span=1,0,0,ccs,hdb3,crc4 |
| 17:02.33 | Kobaz | okay |
| 17:02.42 | Kobaz | didn't know it was uk |
| 17:02.43 | tzafrir_laptop | TSM, span=1,1,0,ccs,hdb3,crc4 |
| 17:03.03 | Kobaz | tzafrir_laptop: the second and third columns don't mean anything when using sangoma |
| 17:03.20 | tzafrir_laptop | It's the second |
| 17:06.59 | TSM | i keep getting "DAHDI_SPANCONFIG failed on span 1: Invalid argument (22)" is this normal? |
| 17:07.04 | TSM | doubt it |
| 17:07.23 | Kobaz | nope that's not goo |
| 17:07.35 | Kobaz | that's from dahdi_cfg -vv |
| 17:07.36 | Kobaz | ? |
| 17:08.27 | TSM | yup |
| 17:08.37 | TSM | its configed all the cahnnels |
| 17:08.43 | TSM | then thats the last error message |
| 17:08.52 | manxpower | no, if you get that message, then it did not configure all the channels. |
| 17:09.14 | TSM | http://pastebin.com/m40d5a78f |
| 17:09.21 | manxpower | the channel list is what YOU have configured in the config file, not what is actually configured in hardware |
| 17:09.21 | [TK]D-Fender | since when is a UK PRI... T1? |
| 17:09.42 | TSM | its not, its a E1 |
| 17:09.58 | [TK]D-Fender | [12:57]<TSM>Kobaz: for pri what do i put, pri=1-23? <- FAIL |
| 17:10.01 | TSM | ahh the D channel is not 24 |
| 17:10.12 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 17:10.25 | *** join/#asterisk dlewis (i=182c9c15@about/security/staff/dlewis) |
| 17:10.31 | [TK]D-Fender | ~pri |
| 17:10.31 | infobot | somebody said pri was [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 17:11.09 | [TK]D-Fender | TSM: bchan=1-15,17-31 |
| 17:11.14 | [TK]D-Fender | TSM: dchan=16 |
| 17:11.37 | Katty | hugs Deeewayne, a bit belatedly |
| 17:18.37 | jaytee | USA always gets the short end of the stick. I want my additional 7 b channels just like the Europeans get!!!! I want it now!!! And I want it wrapped in a pretty pink bow! |
| 17:18.40 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 17:18.53 | TSM | [TK]D-Fender: getting there, do i need to setup a span line for each card? |
| 17:19.24 | [TK]D-Fender | TSM: "SPAN =" is for Digital interfaces, not analog |
| 17:19.25 | Kobaz | span is just for t1/pri |
| 17:19.45 | [TK]D-Fender | Kobaz: Please stop saying T1... that was the source of his problems ;) |
| 17:19.50 | Kobaz | heh |
| 17:19.54 | Kobaz | oh, right... it's e1 |
| 17:20.05 | TSM | im still getting -DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 17:20.39 | Kobaz | [TK]D-Fender: i've never worked with an e1... heh |
| 17:20.53 | TSM | would it matter if my E1 is still not connected to the NTE |
| 17:21.02 | [TK]D-Fender | TSM: and I don't see your new configs |
| 17:21.19 | Kobaz | TSM: no, you can still provision it without a connection |
| 17:21.22 | traxx | hi. i have an asterisk server with 2 nics using two subnets. phones connecting from both nics can dial and receive without problems, yet calling an extension on the other subnet results in no audio. this server is not configured as a router and there is no firewalling between the nics. anybody have any ideas what could cause the audio loss? |
| 17:21.53 | Kobaz | traxx: sip? |
| 17:22.12 | Kobaz | tamiel: since they are on seperate segments... you need to have canreinvite=no |
| 17:22.22 | Kobaz | er... to traxx, rather |
| 17:22.29 | traxx | Kobaz: ok, i'll try that |
| 17:22.34 | [TK]D-Fender | traxx: the endpoints are probably trying to reinvite and finding no route |
| 17:22.43 | Kobaz | otherwise the endspoints wont be able to communicate. yeah |
| 17:23.00 | [TK]D-Fender | traxx: Or do yourself a favour and setup your server to IP forward |
| 17:23.04 | Kobaz | that too |
| 17:23.54 | Kobaz | wanconfig should have generated dahdi/system though |
| 17:23.54 | traxx | hmm, atm i have 'canreinvite=nonat' and 'nat=yes' |
| 17:24.03 | Kobaz | canreinvite=no |
| 17:24.17 | traxx | Kobaz: there's a difference then, eh? |
| 17:24.24 | Kobaz | there is a difference |
| 17:24.28 | traxx | Kobaz: thanks |
| 17:24.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:26.41 | TSM | [TK]D-Fender: http://pastebin.com/m17563b1f |
| 17:27.14 | [TK]D-Fender | TSM: and the rest? |
| 17:27.20 | *** join/#asterisk buttons840 (n=buttons8@63.230.20.246) |
| 17:27.46 | TSM | thats all in the system.conf |
| 17:27.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:28.13 | TSM | ahh all the stuff in chan_dahdi? |
| 17:29.11 | Kobaz | no, we've been just talking system.conf |
| 17:29.16 | TSM | i think there is a mess here, i have chan_dahdi.conf in the asterisk folder that shows all the channels config |
| 17:29.27 | TSM | it seems to be the same as whats in the system.conf |
| 17:29.30 | Kobaz | there's *3* parts.. with sangoms cards |
| 17:30.00 | Kobaz | wanpipe, which controls the sangoma driver... dahdi/system.conf which is the dahdi specific options... an then chan_dahdi.conf which is specific to asterisk |
| 17:30.08 | buttons840 | for the AMI originate command if the channel arguement is SIP/12345@outroute it will place this call through the outroute route? |
| 17:30.23 | [TK]D-Fender | buttons840: Out that peer in sip.conf, yes |
| 17:31.05 | buttons840 | so a route is "kind of like" a phone, sip.conf see it as a peer? |
| 17:31.18 | buttons840 | if i set it that way of course. |
| 17:32.38 | Kobaz | buttons840: no... a 'route' is a context in the dialplan |
| 17:32.55 | Kobaz | a sip peer... is a sip peer... ie... phone...endpoint... 'trunk'... |
| 17:33.03 | voipmonk | or gateway |
| 17:33.08 | voipmonk | :) |
| 17:33.09 | Kobaz | yeah |
| 17:33.28 | Kobaz | so your outroute... is actually a sip peer |
| 17:33.44 | Kobaz | it's sending the digits 12345 to the sip peer called outroute |
| 17:33.47 | *** join/#asterisk n3hxs (n=HAMming@75-145-173-133-Illinois.hfc.comcastbusiness.net) |
| 17:34.24 | [TK]D-Fender | buttons840: "route", "trunk", "phone" are virtually meaningless terms. |
| 17:34.54 | [TK]D-Fender | buttons840: you have a sip.cofn entry to contact another SIP device and pass the call. * doesn't care what that device does. |
| 17:37.19 | TSM | weird, as soon as span was changed from 1 to 2 it started dahdi correctly, dont know why when in wanpipe1 its set as span 1m any ideas |
| 17:37.31 | *** join/#asterisk jayrod422 (n=jarrod@65.196.126.11) |
| 17:38.03 | TSM | still no dial tone though |
| 17:38.26 | Kobaz | well according to your configs... span 1 is the tdm port... and then channels 25 and 26 are the analog |
| 17:38.52 | Kobaz | so span=2 definitly isn't going to work properly |
| 17:39.22 | Kobaz | er |
| 17:39.23 | Kobaz | no wait |
| 17:39.45 | Kobaz | i'm killing you... with assuming stuff is t1... 32 and 33... that's right |
| 17:40.07 | TSM | Kobaz: up i changed it to make it correct |
| 17:40.55 | Kobaz | FE_MEDIA = E1 |
| 17:40.57 | Kobaz | okay that's good |
| 17:41.12 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 17:41.15 | p3nguin | Still having trouble finding the realized value of externhost= |
| 17:41.18 | Kobaz | TE_CLOCK = MASTER <- are you sure that's right? the network should be giving you clock |
| 17:41.34 | Kobaz | although that's not going to affect channel configs |
| 17:41.47 | voipmonk | chuckles |
| 17:41.53 | TSM | im fairly sure, it worked last time i used the E1 line |
| 17:42.12 | [TK]D-Fender | p3nguin: Place a call that should use it |
| 17:42.16 | Kobaz | every span can be configured completely differently... you need to check with your provider |
| 17:42.28 | [TK]D-Fender | p3nguin: Look at a qualify packet being sent... |
| 17:42.39 | Kobaz | [TK]D-Fender: i'm not sure why chan 20 would be failing... i would start talking to sangoma support |
| 17:42.49 | buttons840 | I'm still having trouble with WaitForSilence not working. We are using qwest ipld, sip protocol, and this has something to do with the problem, because if I call local (take the ipld out of the picture), WFS works, but when placing a call over the sip trunk (qwest ipld) WFS does not work. Any idea what is it about this qwest ipld that could cause such a problem? Voice conversations works, and I can detect properly the DTMF tones. |
| 17:43.34 | [TK]D-Fender | Kobaz: I can picture a few obvious reasons.... |
| 17:43.49 | [TK]D-Fender | Kobaz / TSM : CLUE..... 4 + 16 |
| 17:44.59 | Kobaz | and 31-15 = 16 too |
| 17:45.21 | p3nguin | [tk]d-fender: Other than that, shouldn't the value be shown somewhere, since * is looking it up every N seconds? |
| 17:45.29 | [TK]D-Fender | Kobaz: Try again.... |
| 17:45.41 | [TK]D-Fender | facepalms |
| 17:45.45 | Kobaz | [TK]D-Fender: what's magical about 4 + 16... other than 16 is the dchan |
| 17:46.13 | [TK]D-Fender | Kobaz: the 16 channel on the SPAN, but not the 16th DAHDI CHANNEL |
| 17:46.32 | *** join/#asterisk dysinger (n=tim@cpe-75-85-135-191.hawaii.res.rr.com) |
| 17:47.22 | Kobaz | but wanpipe1 is span1, which is the tdm card |
| 17:47.39 | Kobaz | so it would be the 16th dahdi channel, since it starts from 1, at span 1 |
| 17:47.48 | [TK]D-Fender | Kobaz: Which doesn't necessarly imply how DAHDI orders the cards <- |
| 17:48.09 | Kobaz | i've always had dahdi channels come up in the same order as they do in wanpipe |
| 17:48.18 | [TK]D-Fender | Kobaz: 20 could bitch because it is the D-CHANNEL. Now why would it ne on 20? because the analog card is counted FIRST |
| 17:48.31 | Kobaz | could be |
| 17:48.57 | Kobaz | TSM: okay so... according to fender... swap the analog and tdm... start tdm at channels 1,2 and then do the tdm config |
| 17:48.58 | [TK]D-Fender | TSM: Reverse your channel ordering |
| 17:49.13 | [TK]D-Fender | TSM: And start the PRI at 5 |
| 17:50.07 | TSM | dont you mean start at 3, i only have 2 FXS ports? |
| 17:50.13 | Kobaz | you have a 4 port card |
| 17:53.09 | *** join/#asterisk telnettech (n=telnette@71.74.91.116) |
| 17:53.37 | *** join/#asterisk errotan (n=errotan@5403E64C.catv.pool.telekom.hu) |
| 17:53.54 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
| 17:55.37 | Katty | has lunch! |
| 17:55.44 | carrar | share pls |
| 17:55.53 | TSM | ahh just checked /proc/dahdi i have span1 as dahdi dummy, dont know why |
| 17:56.02 | Kobaz | that'll do it |
| 17:56.12 | Kobaz | it's because dahdi was loaded before wanpipe was |
| 17:56.18 | Katty | it's bad for my health. |
| 17:56.25 | Katty | but the doc said i had to relax. |
| 17:56.33 | Katty | kfc bowl. |
| 17:56.37 | Kobaz | haha |
| 17:56.38 | TSM | i use wanrouter restart |
| 17:56.45 | TSM | it does wanpipe then dahdi |
| 17:56.48 | Kobaz | kfc is very healthy indeeed |
| 17:57.03 | p3nguin | no trans fat! |
| 17:57.07 | Kobaz | do wanrouter stop |
| 17:57.10 | Kobaz | and dahdi stop |
| 17:57.22 | Kobaz | and make sure no dahdi modules are loaded in lsmod |
| 17:57.26 | Kobaz | and then start wanrouter again |
| 17:57.34 | Katty | kfc is awful |
| 17:57.40 | Katty | but their bowls are yummy |
| 17:57.53 | Katty | and will help me nap |
| 17:58.06 | carrar | You need turkey for napping |
| 17:58.30 | Katty | turkey doesn't make me sleepy |
| 17:58.35 | Katty | or lettuce |
| 17:58.38 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:58.45 | Katty | carbs do. |
| 18:00.35 | TSM | Kobaz: yup back to span1 |
| 18:01.20 | Kobaz | TSM: after you figure out everything, you'll need to reboot and check your startup order to make sure the stuff is loaded right and you don't get dahdi dummy |
| 18:02.32 | Katty | jaytee: mew. |
| 18:02.41 | TSM | Kobaz: ile stop for tonight, i still have to work out why there is no dial tone, need to eat |
| 18:03.09 | Kobaz | there won't be any dialtone if dahdi/system isn't set up right |
| 18:06.04 | TSM | Kobaz: its starting now, but i bet there is somthing silly with sangoma where the channel numbers start at the highest number for the card |
| 18:06.41 | Kobaz | from my experience, the wanpipe order is the same as the dahdi order |
| 18:06.46 | TSM | Kobaz: the TDM card goes from 32-55, i only have 2 ports |
| 18:07.01 | TSM | ile drop them an email when i get home |
| 18:07.18 | TSM | im off |
| 18:07.20 | Kobaz | k |
| 18:12.58 | *** join/#asterisk hetii (i=576333ac@gateway/web/freenode/x-thqfwimlpgzzenmu) |
| 18:13.02 | hetii | Hello :) |
| 18:13.11 | hetii | whats heppens with asterisk.org ? |
| 18:13.29 | Naikrovek | what happens? |
| 18:13.47 | Naikrovek | or what happened |
| 18:13.48 | carrar | for me, it loads a w eb page |
| 18:14.05 | Naikrovek | it is a new design though |
| 18:14.15 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 18:14.44 | carrar | looks SEXY++ |
| 18:14.48 | hetii | ahh k now works :) |
| 18:15.07 | hetii | but few hour before it was not accesible:) |
| 18:15.22 | hetii | ok with on asterisk i shuld use for current freePBX ? |
| 18:15.32 | hetii | 1.4 or 1.6 ? |
| 18:15.34 | carrar | Use Asterisk from Source |
| 18:15.36 | carrar | Always |
| 18:16.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.01 | hetii | ok but with version of sources? |
| 18:17.23 | carrar | 1.6.1.6 |
| 18:20.06 | hetii | ok |
| 18:20.10 | hetii | thx |
| 18:23.29 | *** join/#asterisk voxter (n=voxter@32.176.27.136) |
| 18:27.34 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 18:29.14 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:29.43 | Gugge | Is it possible to add sip headers to the bye and/or the bye 200 OK response packets? |
| 18:30.41 | hetii | is it possible to set some king of ecryption / additional security on asterisk ? for sip eg ? |
| 18:31.20 | hetii | eg using openssl certificate on authorization process ? |
| 18:35.19 | hetii | Q: should i install asterisk-addons ? what they provide ? |
| 18:35.48 | hetii | i will just use SIP channel and maybe some capi card |
| 18:38.20 | Gugge | i use the mysql stuff from -addons |
| 18:38.22 | Gugge | nothing else |
| 18:39.07 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 18:39.13 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 18:39.41 | bsilberman | anyone have any experience running MixMonitor in a heavy usage environment? |
| 18:40.14 | bsilberman | I need an example of how much additional CPU Load 60 simultaneous calls will add |
| 18:40.47 | hetii | Gugge: should i patch it like is describe in this site: http://www.powerpbx.org/content/freepbx-debian ? |
| 18:41.08 | Gugge | i dont know |
| 18:41.20 | Gugge | i dont patch anything |
| 18:43.26 | buttons840 | does anyone know about the AMDSTATUS variable set by the AMD() app? AMDSTATUS is always Not Sure, and AMDCAUSE is always TOOLONG-122500 (yes, I have the analysis time set to 120,000 milli-secs). The toolong doesn't make sense, does anyone have any ideas? |
| 18:43.43 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 18:44.28 | *** join/#asterisk cuco (n=elcuco@bzq-79-177-136-164.red.bezeqint.net) |
| 18:44.36 | cuco | tzafrir_laptop: http://www.kdenews.org/2009/10/12/free-n900 |
| 18:45.14 | geneticx | Has anyone implemented a digium TDM404E with asterisk 1.4.26 ? would you recommend this card for 4 FXO lines? |
| 18:46.35 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 18:47.17 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 18:47.57 | hetii | Q: i hope last, did capi support is included on asterisk 1.6 ? or should i install it from http://www.melware.org/ChanCapi/ ? |
| 18:50.21 | tzafrir_laptop | cuco, which makes it safe enough for you? |
| 18:51.17 | hetii | ok i find answer on my last q :) |
| 18:53.56 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 18:54.28 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-120.link.net.pk) |
| 18:55.50 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 18:56.49 | *** part/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 18:57.46 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:58.15 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.226.139) |
| 18:59.56 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 19:00.15 | *** join/#asterisk xuser_ (n=xuser@unaffiliated/xuser) |
| 19:00.22 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:03.47 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:04.50 | guax | hi guys, can someone give me some feedback on this? http://www.guax.net/2009/10/call-flow-diagram/ |
| 19:05.05 | guax | oh, and girls as well |
| 19:05.08 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 19:08.16 | wcselby | ooh, I like the new asterisk.org page |
| 19:09.03 | wcselby | except the community stats need to be updated :) |
| 19:09.11 | guax | wow, it changed =P |
| 19:09.55 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 19:09.55 | guax | i liked more the direct download links in the homepage |
| 19:14.11 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:16.06 | buttons840 | I've placed a call over a qwest IPLD line. I've tried calling 2 different phones on the PSTN (my cell phone). I can have a normal voice conversation over this connection. The funny thing is at the beginning of my dial plan I have a waitforsilence call which doesn't work, the playback starts without waiting for silence. I can press DTFM tone, and the dialplan responds to it, and after the tone, the dialplan starts over and waitforsilence begins |
| 19:16.07 | buttons840 | working as expected. Something about pressing the tone cause waitforsilence to go from not working to working? |
| 19:19.21 | [TK]D-Fender | buttons840: I don't see configs and call debug... |
| 19:19.48 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
| 19:21.41 | seanmh | Katty: how's the 1.6 testing going? |
| 19:22.55 | *** join/#asterisk Nafiux (n=IceChat7@189.226.72.140) |
| 19:22.56 | *** join/#asterisk Mikhey (n=Brian@173-15-138-238-BusName-Philadelphia.hfc.comcastbusiness.net) |
| 19:22.58 | *** part/#asterisk Mikhey (n=Brian@173-15-138-238-BusName-Philadelphia.hfc.comcastbusiness.net) |
| 19:23.15 | Nafiux | Good afternoon, can speak spanish? |
| 19:24.34 | [TK]D-Fender | Nafiux: #asterisk-es |
| 19:24.44 | Nafiux | Thank you! |
| 19:26.14 | Nafiux | Nothing in asterisk-es... |
| 19:26.18 | Nafiux | I have a question... |
| 19:26.58 | Nafiux | I have two points communicated with an AP and a Client Bridge |
| 19:27.22 | Nafiux | Can I use asterisk in the two segments? |
| 19:28.02 | [TK]D-Fender | Nafiux: Networking is networking.... * doesn't care about the medium |
| 19:28.56 | manxpower | The only thing you will care about (Asterisk doesn't care) is Jitter |
| 19:29.16 | Nafiux | ok, the next question :P I need a telephone ip? and what about it? this telephones have a rj45 connectors? and have an local ip? |
| 19:29.25 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:29.25 | hetii | what is astapi ? shoudl i use it instand AMI ? |
| 19:29.37 | manxpower | Nafiux: You should try reading the Asterisk Book |
| 19:29.40 | manxpower | ~book |
| 19:29.41 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:29.53 | Nafiux | Perfect!! |
| 19:29.54 | manxpower | ~phones |
| 19:29.54 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 19:29.55 | hetii | or nornall set in manager.conf account for freepbx ? |
| 19:30.28 | manxpower | hetii: FreePBX uses Manager, so you would need to set that up. More info can be found on #FreePBX |
| 19:30.29 | [TK]D-Fender | hetii: Yes, and you are asking in the wrong channel |
| 19:30.34 | manxpower | ~freepbx |
| 19:30.35 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:30.35 | [TK]D-Fender | hetii: #freepbx <------ |
| 19:30.50 | hetii | ok thx |
| 19:30.51 | hetii | :) |
| 19:30.52 | bmoraca | my experiences with Aastra phones has not been good...i'd recommend Cisco over them and the Linksys phones any way |
| 19:31.35 | *** join/#asterisk afink (n=afink@204.26.87.226) |
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| 19:52.47 | *** join/#asterisk DrZeus (n=bobzilla@201.226.170.106) |
| 19:53.59 | DrZeus | hi all |
| 19:54.32 | *** part/#asterisk DrZeus (n=bobzilla@201.226.170.106) |
| 19:58.50 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
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| 20:01.29 | *** part/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 20:01.37 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 20:02.38 | buttons840 | [TK]D-Fender, what configs are you most interested in? I'm getting a pastebin ready. I've done more testing to confirm the behavior. |
| 20:02.42 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:04.40 | *** join/#asterisk svenx (n=sveniu@pat-tdc.opera.com) |
| 20:05.34 | *** join/#asterisk DrZeus (n=bobzilla@201.226.170.106) |
| 20:06.02 | DrZeus | hi all. Questoin: does anyone knows which AMI command does the flash panel operator uses for connecting 2 peers? |
| 20:06.55 | [TK]D-Fender | DrZeus: Originate |
| 20:07.49 | DrZeus | [TK]D-Fender: I was trying to replicate how it works, but i haven't been able to get the peers connected so far |
| 20:08.02 | DrZeus | i thought maybe there was another command to do so |
| 20:10.21 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 20:10.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 20:16.41 | *** join/#asterisk balaa (n=balaa@adsl-99-49-10-78.dsl.pltn13.sbcglobal.net) |
| 20:17.18 | hetii | Q: when i run asterisk using asterisk -cvvvvvv he create /var/run/asterisk.ctl but i suppose that he should do it on /var/run/asterisk because i set this path on /etc/asterisk/asterisk.conf, when i use amportal he dpnt create any asterisk.ctl or asterisk.pid on any location |
| 20:21.10 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 20:26.09 | tzafrir_laptop | is asterisk..conf readable? what is its contents? What version of asterisk? (asterisk -V ) |
| 20:26.47 | DigitalFlux | guys |
| 20:26.55 | DigitalFlux | i remember some one told me a command here |
| 20:27.02 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:27.02 | DigitalFlux | where the output of lspci i got was |
| 20:27.11 | DigitalFlux | unknown device |
| 20:27.27 | DigitalFlux | executing the command got the lspci result just right |
| 20:27.41 | DigitalFlux | any one remember the name of that command ? |
| 20:29.18 | wcselby | update-pciids ? |
| 20:30.01 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 20:30.33 | DigitalFlux | wcselby: that was the magic command :) |
| 20:30.36 | DigitalFlux | wcselby: Thanks |
| 20:30.45 | wcselby | np |
| 20:34.40 | wcselby | from the 1.6.2 CHANGES file: * Added LOCAL_PEEK, which I have no idea how to use, but Leif Madsen wanted it. |
| 20:34.40 | wcselby | <PROTECTED> |
| 20:34.40 | wcselby | <PROTECTED> |
| 20:35.07 | wcselby | lol |
| 20:37.08 | *** join/#asterisk jmoncayo (n=mcdsmail@mail.mcds.gov.ec) |
| 20:37.40 | jmoncayo | hey guys, what kind fo hardware do i need to create a calling center |
| 20:38.54 | p3nguin | A computer, some phones w/headsets, some cables, some switches... |
| 20:40.25 | jmoncayo | how would the conection from the public telephone to the ip call center be? |
| 20:40.55 | wcselby | what kind of connection do you have? T1? E1? Analog? |
| 20:41.58 | wcselby | you can get cards for asterisk that handle all of these types of connections, or you can go completely IP based if you use an ITSP. |
| 20:42.27 | jmoncayo | it will be analog, pstn |
| 20:42.49 | wcselby | you can get FXO/FXS analog cards then that you plug into your asterisk server. |
| 20:43.34 | jmoncayo | ok i will google on that, thanks |
| 20:44.18 | wcselby | you'll need to look for TDM cards, Digium makes some that would probably work for you |
| 20:44.37 | wcselby | also, you'll probably want to take a look at THE BOOK |
| 20:44.39 | wcselby | ~book |
| 20:44.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:44.50 | DrZeus | question: when you use 'originate' to connect two peers, the 'sender' telephone should always ring? |
| 20:44.52 | wcselby | it has a lot of useful information |
| 20:45.34 | jmoncayo | thank u a lot |
| 20:46.00 | p3nguin | drzeus: If you're originating from SIP/1001 to extension 1002@phones, then 1001 should ring, then when you answer it, 1002 will ring and will be connected to 1001. |
| 20:47.00 | DrZeus | p3nguin: i've been trying to do this (from a script, and now directly from the AMI console), but when the 'sender' rings and I pick up, the call never goes through |
| 20:47.50 | DrZeus | i thought at first that maybe the script was doing something wrong, but i can't connect a call even using the AMI console |
| 20:47.53 | p3nguin | drzeus: On your shell, run asterisk -r. |
| 20:48.34 | p3nguin | drzeus: originate SIP/1001 extension 1002@yourcontext |
| 20:48.38 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp3-139.adsl.forthnet.gr) |
| 20:48.49 | p3nguin | drzeus: Use your real extension numbers and an actual context. |
| 20:49.07 | DrZeus | that is from the asterisk console, or the AMI? |
| 20:50.21 | DrZeus | well...this time it worked! |
| 20:50.35 | p3nguin | I have no idea what AMI even is. Just use the asterisk console for now. |
| 20:50.35 | *** join/#asterisk voxter (n=voxter@166.183.59.39) |
| 20:50.43 | DrZeus | asterisk manager interface? |
| 20:51.06 | DrZeus | the port 5038 |
| 20:51.29 | p3nguin | I'm familiar with the acronym, but have no idea what the thing is. |
| 20:51.54 | DrZeus | is another shell, where you write the full instructions |
| 20:52.18 | DrZeus | action: originate |
| 20:52.20 | DrZeus | channel: sip/3001 |
| 20:52.21 | DrZeus | context: from-internal |
| 20:52.23 | DrZeus | extension: sip/3000 |
| 20:52.24 | DrZeus | things like that |
| 20:54.06 | p3nguin | I don't understand why you need yet another interface when the console works just fine. |
| 20:54.18 | DrZeus | how do you send asterisk commands through a script? |
| 20:54.25 | kaldemar | it's an api listening to a tcp socket. |
| 20:54.35 | DrZeus | which port is the one for asterisk console? |
| 20:54.46 | *** join/#asterisk traxx (n=traxx@91-64-130-76-dynip.superkabel.de) |
| 20:55.09 | p3nguin | I use regular shell scripting and "asterisk -rx" |
| 20:55.12 | DrZeus | kaldemar: i've been trying to use the manager interface; never thought I could use the asterisk console directly |
| 20:56.40 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:58.54 | DrZeus | p3nguin: i tried to run this, but didn't made the call: localhost$ asterisk -rx originate sip/3004 extension 3000@from-internal |
| 20:59.08 | p3nguin | quote the string |
| 20:59.19 | DrZeus | oh...got it |
| 20:59.27 | p3nguin | asterisk -rx "originate ... ... ..." |
| 21:01.01 | DrZeus | i tested, it connected |
| 21:01.13 | DrZeus | thank you, A LOT |
| 21:01.33 | DrZeus | i've been having several issues trying to figure out how to do this |
| 21:01.48 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 21:02.06 | DrZeus | now I think I was over-complicating things using the other method |
| 21:02.13 | IBC_jkenney | Has anyone found a good howto on populating the Directory on the spa962 phones? |
| 21:02.14 | p3nguin | maybe |
| 21:02.46 | IBC_jkenney | I would love to see it |
| 21:04.12 | kaldemar | AMI is more robust than using -rx |
| 21:05.13 | kaldemar | there are 4 ways to originate calls nowadays. |
| 21:05.35 | DrZeus | kaldemar: i've been trying to generate the call from the AMI |
| 21:05.42 | DrZeus | but it never goes through |
| 21:06.05 | kaldemar | they do when done right. |
| 21:06.06 | DrZeus | when the sender picks up to get connected, the call dies |
| 21:06.10 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 21:06.13 | DrZeus | kaldemar: really. |
| 21:06.55 | IBC_jkenney | Has anyone found a good howto on populating the Directory on the spa962 phones? |
| 21:07.12 | DrZeus | IBC_jkenney: never used that; sorry |
| 21:07.14 | kaldemar | really. and there's really no sender, just two sides that get called and connected to each other. |
| 21:07.34 | IBC_jkenney | I'd like to mass update them |
| 21:07.40 | IBC_jkenney | 30 phones |
| 21:07.42 | IBC_jkenney | and all |
| 21:07.44 | DrZeus | kaldemar: yes; sender was for making myself clear i guess |
| 21:08.13 | DrZeus | kaldemar: I wrote all the instructions in directly in the AMI, and the call doesn't go through anyway |
| 21:08.58 | DrZeus | i have checked the order of the commands, interchange 'timeout' with 'waittime', still nothing |
| 21:16.20 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 21:19.26 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:23.41 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 21:24.28 | DrZeus | kaldemar: any recommendations for this? i don't know what else to try with AMI |
| 21:24.55 | [TK]D-Fender | DrZeus: I don't see you showing us anything to debug... |
| 21:24.58 | [TK]D-Fender | ~pb |
| 21:24.59 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 21:25.00 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 21:25.12 | DrZeus | ok...missed that |
| 21:25.26 | DrZeus | hold on |
| 21:26.24 | DrZeus | this is the order of the strings that are sent to AMI: http://pastebin.com/d7abf1c23 |
| 21:28.45 | [TK]D-Fender | DrZeus: No skipping lines like that... |
| 21:28.52 | [TK]D-Fender | DrZeus: And I don't see the CALL ATTEMP |
| 21:29.02 | DrZeus | i don't skip lines, that was pastebin |
| 21:29.57 | DrZeus | CALL ATTEMP? |
| 21:30.06 | [TK]D-Fender | DrZeus: extension: sip/3000 <- NOT VALID |
| 21:30.32 | DrZeus | [TK]D-Fender: i have tried with just 3000, but still nothing |
| 21:30.56 | [TK]D-Fender | DrZeus: And I don't see your attempt nor any proof that even if you changed it that there'd be a match |
| 21:31.40 | DrZeus | [TK]D-Fender: i think i didn't understood that |
| 21:32.26 | [TK]D-Fender | DrZeus: I don't see you dialplan so I'm not even going to assume that extension # is valid in there. I don't see * CLI during a call-out attempt. NOTHING. |
| 21:33.24 | DrZeus | let me paste the messages it gave me then; hold on |
| 21:34.00 | *** join/#asterisk [8none1]_ (n=[8none1]@67.107.93.2) |
| 21:37.53 | *** join/#asterisk voxter (n=voxter@166.128.124.175) |
| 21:38.02 | DrZeus | i can't copy/paste in this putty for unix |
| 21:38.17 | DrZeus | does any of you know how? |
| 21:44.27 | [TK]D-Fender | DrZeus: Use another terminal |
| 21:45.41 | DrZeus | here it is: pastebin.com/d660cc300 |
| 21:45.54 | DrZeus | that is what the AMI gave me when I tried; all the messages |
| 21:46.55 | voipmonk | http://pastebin.com/d660cc300 :) |
| 21:48.19 | DrZeus | voipmonk: hehe...thanks |
| 21:48.22 | DrZeus | :) |
| 21:48.32 | voipmonk | chuckles |
| 21:49.46 | [TK]D-Fender | DrZeus: DIALPLAN.... |
| 21:49.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 21:50.32 | DrZeus | hold on... |
| 21:52.04 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 21:52.45 | doolittlework | hi there has anyone found a way to do cyclic dialing on fxo zap channels? |
| 21:53.14 | DrZeus | [TK]D-Fender: diaplan: http://pastebin.com/d43abba02 |
| 21:53.44 | hardwire | What software creates pretty sip trace graphs? |
| 21:53.51 | hardwire | wireshark kinda sorta does |
| 21:54.24 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:54.50 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 21:54.57 | kaldemar | DrZeus: unless you have exten 3000 in extensions_custom.conf, you're trying to connect the call to an invalid extension. and: |
| 21:54.58 | [TK]D-Fender | DrZeus: And where is the CLI output of the attempt? |
| 21:55.02 | kaldemar | ~freepbx |
| 21:55.03 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 21:55.24 | DrZeus | i am not using the CLI; Im using the AMI |
| 21:55.39 | [TK]D-Fender | DrZeus: And where is the CLI output of the attempt? |
| 21:55.52 | doolittlework | DrZeus msg me and i will help i know freepbx |
| 21:56.34 | DrZeus | doolittlework: that's great |
| 21:56.39 | kaldemar | doolittlework: make a group in zapata.conf and dial with Dial(Zap/r1/...) |
| 21:58.22 | doolittlework | thx kaldemar |
| 21:59.09 | bmoraca | faxing over SIP is way starting to piss me off |
| 21:59.59 | bmoraca | there's no good reason for this not to work...jitter is 0 and my latency is only 40ms...and fax tones will NOT be detected by nvfaxdetect |
| 22:00.29 | DrZeus | but, even if its just freepbx, can't the originate just work through the AMI? i used the CLI and the calls could get connected without problems |
| 22:01.01 | voipmonk | its voip , bmoraca there are many reasons for it not to work :) |
| 22:01.38 | bmoraca | i said no GOOD reason for it not to work. it's just not working to spite me |
| 22:01.40 | voipmonk | what codec are you using . bmoraca ? |
| 22:01.45 | bmoraca | g711 |
| 22:01.54 | [TK]D-Fender | And we've got another person who doesn't follow directions... |
| 22:03.53 | hardwire | bmoraca: you down with g711! |
| 22:03.57 | hardwire | is down with g711. |
| 22:04.11 | hardwire | the protocol for E911 shuold be g911. |
| 22:04.21 | hardwire | err.. codec :) |
| 22:04.30 | bmoraca | hardwire: i would be more down with it if my faxing was working properly |
| 22:04.38 | hardwire | bmoraca: aww |
| 22:04.52 | bmoraca | i'm thinking it's a provider thing |
| 22:04.54 | hardwire | faxing over sip is way not fun. |
| 22:05.06 | bmoraca | i wonder if it's possible for my provider to screw with my packets like that |
| 22:05.07 | hardwire | are you pumping the call to iaxmodem at all? |
| 22:05.14 | bmoraca | hardwire: yes |
| 22:05.17 | hardwire | if so.. you can have it write audio to file |
| 22:05.18 | bmoraca | hardwire: and no. |
| 22:05.22 | hardwire | then listen in and be all wtf. |
| 22:06.07 | hardwire | listening to faxes isn't a horrible idea. |
| 22:06.07 | bmoraca | hardwire: fax tone detection isn't working and although iaxmodem handshakes properly, the fax never completes |
| 22:06.08 | hardwire | doh |
| 22:06.15 | hardwire | pulls up his provider black list |
| 22:06.18 | hardwire | what provider? :) |
| 22:06.24 | bmoraca | happens with a direct sip trunk as well as a sip trunk through a TA900 |
| 22:06.36 | bmoraca | hardwire: Telepacific provides my T1s here. |
| 22:06.55 | hardwire | what about between local asterisk servers using sip? |
| 22:07.34 | bmoraca | hardwire: i don't have any trouble with that between my asterisk servers at my colo...it's only one outside that i'm trying to work with. |
| 22:07.44 | hardwire | doh |
| 22:07.57 | hardwire | is something in the middle doing a T.38 hijack? |
| 22:08.04 | bmoraca | that's what i'm afraid of |
| 22:08.12 | bmoraca | although i'm not using t.38 |
| 22:08.16 | hardwire | explains why only a little of the audio is usable |
| 22:08.28 | bmoraca | but i wouldn't put it past telepacific to monkey with my shit |
| 22:08.34 | hardwire | shit monkeys. |
| 22:08.55 | bmoraca | i wish i had another internet connection handy that i could test with |
| 22:09.22 | voipmonk | thats what they do |
| 22:09.23 | hardwire | you can set up an asterisk server somewhere else to bounce back to your own server. |
| 22:09.31 | voipmonk | keeps their technicians paid |
| 22:09.31 | hardwire | and force it to not allow reinvites |
| 22:10.07 | bmoraca | i don't recall seeing reinvites in my sip debugs, though |
| 22:10.35 | hardwire | I only mention it to make sure audio goes to/from the remote server vs just bridging it locally. |
| 22:10.44 | bmoraca | right |
| 22:11.14 | hardwire | at least for testing with another asterisk box somewhere on teh webz |
| 22:11.31 | bmoraca | the way i want it set up in the end is PRI to Asterisk and then SIP from Asterisk to a TA900 and then PRI from the TA900 to Asterisk |
| 22:11.33 | hardwire | my ravioli in a can wasn't enough food. |
| 22:11.49 | carrar | have 2 |
| 22:11.54 | hardwire | I ran out |
| 22:12.06 | voipmonk | have some spoon cake |
| 22:12.28 | carrar | drink 2 glasses of water |
| 22:12.29 | hardwire | bmoraca: so local to asterisk to ta900 to same asterisk to local? |
| 22:13.00 | bmoraca | hardwire: no, the first Asterisk is in my Colo...the TA900 and the second Asterisk could be anywhere on any provider |
| 22:13.06 | hardwire | ah |
| 22:13.16 | voxter | hardwire: astricon this year? |
| 22:13.34 | hardwire | voxter: I'm a frayed knot. |
| 22:13.43 | voxter | hardwire: doh. im here already |
| 22:13.56 | hardwire | yeh well.. you didn't go drinking with me last time so I'm totally not going. |
| 22:14.17 | *** join/#asterisk alrs (n=lars@66-214-241-226.dhcp.gldl.ca.charter.com) |
| 22:14.28 | doolittlework | sry to bug if one uses the "r" for roundrobin in zap groups , what must i do to play all lines are busy if if all the lines are indeed busy? |
| 22:14.53 | hardwire | doolittlework: if that is the case then it hops to a CONGESTION extension if you allow it |
| 22:15.21 | hardwire | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 22:15.24 | doolittlework | will something like this work , xxx,101,Congestion is this right? |
| 22:15.26 | hardwire | doolittlework: see the first example |
| 22:15.39 | doolittlework | thx hardwire |
| 22:16.18 | hardwire | doolittlework: you have my paypal info eh? |
| 22:16.27 | hardwire | isn't cheap. |
| 22:17.22 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
| 22:17.54 | doolittlework | lol |
| 22:18.28 | doolittlework | i need some money before i can pay a good pal |
| 22:18.36 | hardwire | lol don't keep the kitty litter coming. |
| 22:18.37 | hardwire | hehe |
| 22:18.59 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:21.24 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 22:22.55 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:22.56 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 22:26.27 | mchou | anyone here ever who has ever used an allworx 9112 phone? |
| 22:28.38 | doolittlework | i read somewhere but can't find it now that if a application fails say 123,3,Dial(zapxxxxxxxxxx) the if you do something like 123,103,Playback(all-circuit-are-busy) will this work or am i missing the plot? |
| 22:29.42 | *** join/#asterisk fskrotzki_ (n=fskrotzk@74.74.247.157) |
| 22:30.59 | [TK]D-Fender | doolittlework: You are looking at 1.0 ancient priority jumping crap. |
| 22:31.03 | [TK]D-Fender | doolittlework: and thats +101 |
| 22:31.07 | [TK]D-Fender | doolittlework: no 100 |
| 22:31.09 | [TK]D-Fender | not* |
| 22:31.26 | bmoraca | gross |
| 22:31.37 | hardwire | doolittlework: that handy dandy example I gave explains it. |
| 22:31.47 | hardwire | but n+101 is the old fashion method. |
| 22:31.56 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 22:35.31 | doolittlework | ok i see 123,104,Playback("repeat to yourselve, YOU WILL NEVER GET THIS RIGHT") |
| 22:46.02 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 22:56.45 | *** join/#asterisk [8none1]_ (n=[8none1]@cerberus.franklinamerican.com) |
| 23:01.49 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 23:01.56 | drmessano | ~verbose |
| 23:01.57 | infobot | rumour has it, verbose is try running the verbose option on that command and looking at output for likely problems. |
| 23:02.04 | drmessano | ~verbosity |
| 23:02.48 | drmessano | ~verbosity |
| 23:02.49 | infobot | There once was a man named VVVVVVVVVVVV, who screamed when he hurt his knee. A little lady walked by, with a look in her eye, and said "You're just too verbose for me". |
| 23:03.03 | drmessano | \o/ |
| 23:03.42 | drmessano | Feel free to use that on the next batch of Digium Koozies |
| 23:05.54 | *** join/#asterisk riddlebox (n=james@75.132.225.75) |
| 23:11.11 | *** join/#asterisk darkdrgn2k (n=darkdrgn@70.31.4.196) |
| 23:12.22 | riddlebox | got the problem with teliax figured out |
| 23:12.28 | carrar | whew |
| 23:12.34 | bmoraca | so i can fax to fairbanks alaska over a SIP trunk but i can't fax 30 miles to my office over one...arg! |
| 23:12.37 | darkdrgn2k | hey guys, i just got flooded with calls from new york saying i called them. My asterisk box shows no activity, and provider seems to show no activity. any idea what this was? |
| 23:13.02 | riddlebox | logs? |
| 23:13.13 | darkdrgn2k | riddlebox: logs show nothing of interest.. just the incomming calls |
| 23:13.38 | riddlebox | maybe someone spoofed your caller id? |
| 23:13.51 | darkdrgn2k | riddlebox: thats what im wondering but what would be benefit of that be? |
| 23:14.52 | riddlebox | someone didnt want the other people to know who was calling them? what did the people say the caller said to them? |
| 23:15.13 | darkdrgn2k | that i called them something about a mortgage.. |
| 23:15.26 | darkdrgn2k | i got like 150 calls within an hour |
| 23:15.37 | riddlebox | ouch |
| 23:15.39 | darkdrgn2k | and they are still trickeling in.. but i think thats just answering machine call bakcs |
| 23:15.49 | darkdrgn2k | that kinda volume of outbound calls would be HARD to miss in the logs |
| 23:15.50 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 23:17.07 | *** join/#asterisk geneticx (n=geneticx@adsl-10-114-216.mia.bellsouth.net) |
| 23:18.02 | *** join/#asterisk Trepinon (n=nemus@lbuttars.dsl.xmission.com) |
| 23:18.11 | darkdrgn2k | hmm lsat call was "10/12/09 6:40 PM" |
| 23:18.21 | ChannelZ | riddlebox: good to hear, was going to get you in contact with a friend to hopefully get you in contact with teliax |
| 23:19.52 | darkdrgn2k | AHAHAA definatly was hacked |
| 23:19.56 | darkdrgn2k | i got an extesions called "DAD" |
| 23:20.01 | darkdrgn2k | but dad has been dead for 10 years |
| 23:20.14 | riddlebox | teliax needs to update their doc saying that all you have to do is switch it in the dashboard, you also have to check to make sure your proxy is set (it defaults it to a different one) and your number is also set to go Nowhere - they dont have that on the site |
| 23:23.35 | ChannelZ | eeeediots |
| 23:25.03 | *** join/#asterisk torrancew (n=chatzill@ip70-177-34-83.br.br.cox.net) |
| 23:27.54 | riddlebox | darkdrgn2k, which log were you looking at for that? cdr records? |
| 23:28.41 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 23:29.12 | darkdrgn2k | riddlebox: yes as well as full debug logs |
| 23:29.22 | darkdrgn2k | looks liek they just wanted incomming.. |
| 23:29.29 | darkdrgn2k | they re-routed all calls to "DAD" |
| 23:29.46 | darkdrgn2k | no outgoing calls were actually made |
| 23:30.12 | darkdrgn2k | wonder if they will be sutpid enough to log back in? |
| 23:31.52 | darkdrgn2k | "3004@10.150.8.5:5060" |
| 23:32.19 | darkdrgn2k | dam thats a lan ip :'( |
| 23:34.13 | darkdrgn2k | does this mean they used this ip address |
| 23:34.14 | darkdrgn2k | Via: SIP/2.0/UDP 10.150.8.5:5060;branch=z9hG4bK37933280;received=66.227.69.131;rport=5060^M |
| 23:34.18 | [TK]D-Fender | got hacked by 127.0.0.1 |
| 23:34.22 | [TK]D-Fender | is going to nail that fucker.... |
| 23:34.48 | darkdrgn2k | LOL good luck with that :-P make sure you use a botnet to reall yget him |
| 23:34.53 | darkdrgn2k | hates bot nets |
| 23:36.59 | ChannelZ | what about infobotnets |
| 23:37.15 | darkdrgn2k | haha figures |
| 23:37.19 | darkdrgn2k | the ip address in in the USA |
| 23:37.20 | darkdrgn2k | city |
| 23:37.21 | darkdrgn2k | King Of Prussia |
| 23:38.00 | carrar | Burger King |
| 23:38.24 | ChannelZ | so someone broke into your server and was making calls? |
| 23:38.35 | darkdrgn2k | ChannelZ: thats the thigns they didnt MAKE calls |
| 23:38.53 | darkdrgn2k | ChannelZ: i got flooded with incomming calls.. turns out.. he hacked my server.. created a new extension.. and was Receiving calls |
| 23:39.03 | darkdrgn2k | i got 160+ calls in less than an hour |
| 23:39.24 | ChannelZ | ah.. forwarding them on elsewhere? |
| 23:39.29 | darkdrgn2k | nop |
| 23:39.34 | darkdrgn2k | i think he loged right into the extension |
| 23:39.38 | darkdrgn2k | he created |
| 23:39.42 | darkdrgn2k | and chaged inbound route to that |
| 23:39.45 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:39.54 | darkdrgn2k | (to bed that DID was being forwarded @ the provider level too) |
| 23:39.54 | ChannelZ | but I mean he was remote |
| 23:40.01 | darkdrgn2k | ChannelZ: looks like it.... |
| 23:40.09 | darkdrgn2k | registation shows |
| 23:40.10 | darkdrgn2k | Via: SIP/2.0/UDP 10.150.8.5:5060;branch=z9hG4bK37933280;received=66.227.69.131;rport=5060^M |
| 23:40.29 | darkdrgn2k | who should i go after.. |
| 23:40.31 | darkdrgn2k | Reliance Globalcom Services, Inc RGS-BLK5 66.227.0.0 - 66.227.127.255 |
| 23:40.41 | darkdrgn2k | or HIGH POINT SOLUTIONS 66.227.69.128 - 66.227.69.143 |
| 23:41.13 | *** join/#asterisk e4 (n=e4@76.84.81.72) |
| 23:41.18 | TJNII | darkdrgn2k: How did they get into your box in the first place? |
| 23:41.33 | darkdrgn2k | TJNII: havent figured that part out yet :-S |
| 23:41.40 | darkdrgn2k | im guessing stupid user's week password |
| 23:41.43 | ChannelZ | that was my question |
| 23:41.44 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 23:41.47 | darkdrgn2k | but i cant find any brute force attempts |
| 23:42.06 | [TK]D-Fender | darkdrgn2k: Where is the peer actually located? |
| 23:42.06 | darkdrgn2k | knows now to make sure he has a good firewall script in place |
| 23:42.19 | darkdrgn2k | im guessing "66.227.69.131" |
| 23:42.20 | [TK]D-Fender | darkdrgn2k: the device * is supposed to be talking to that is... |
| 23:42.45 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 23:42.51 | darkdrgn2k | the extesnsion registration http://www.pastebin.ca/index.php |
| 23:42.57 | darkdrgn2k | soh |
| 23:43.04 | darkdrgn2k | http://www.pastebin.ca/1616878 |
| 23:43.40 | darkdrgn2k | i shut down the box the first sign of trouble. he has not tried connecting since |
| 23:44.06 | [TK]D-Fender | darkdrgn2k: No....where is [3004] supposed to be located? |
| 23:44.19 | darkdrgn2k | [TK]D-Fender: 3004 was never there. it was created by the intruder |
| 23:44.31 | darkdrgn2k | [TK]D-Fender: 3004 was flaged as "DAD" my dad dies like 10 years ago :-P |
| 23:44.36 | darkdrgn2k | uhh extension was named |
| 23:44.46 | [TK]D-Fender | darkdrgn2k: Guess that means you left HTTP or SSH open... SMRT |
| 23:45.18 | darkdrgn2k | [TK]D-Fender: yeh i know.. it wasnt a production box... but... i probably should of known better :'( |
| 23:45.50 | darkdrgn2k | just learned that the hard way |
| 23:47.28 | ChannelZ | using an admin interface or they had to have logged in as someone with high enough privs to edit your config? |
| 23:48.00 | darkdrgn2k | im not quite sure how yet... |
| 23:48.09 | darkdrgn2k | but like i said it wasnt a production box so... not many users to choose from |
| 23:48.25 | ChannelZ | inside job :) |
| 23:48.31 | ChannelZ | he was just helping you test |
| 23:48.40 | darkdrgn2k | haha doubtfull |
| 23:48.49 | darkdrgn2k | the only other person on that box paysteh bill to the datacenter :-P |
| 23:48.53 | [TK]D-Fender | is pretty sure its straight HTTP |
| 23:49.08 | darkdrgn2k | is pritty sure tk is right. |
| 23:50.48 | darkdrgn2k | guess thats one reason why asterisk servers should be INSIDE nats |
| 23:52.12 | [TK]D-Fender | darkdrgn2k: No, if you left HTTP open, that is your own fault..... |
| 23:52.36 | ChannelZ | http PUT or something? |
| 23:53.08 | TJNII | has a box with ssh, Apache, ssh, and a public IP, no issues in 2 years.... |
| 23:54.26 | darkdrgn2k | [TK]D-Fender: :-P |
| 23:54.41 | darkdrgn2k | There we go i found him |
| 23:54.42 | darkdrgn2k | 85.120.229.15 |
| 23:54.46 | darkdrgn2k | Romania |
| 23:54.49 | [TK]D-Fender | TJNII: Yes, and do you leave an admin panel for an * box as part of it? |
| 23:55.07 | jblack | which version of * got sip over tcp? |
| 23:55.30 | jblack | was that 1.6.0 ? |
| 23:55.40 | TJNII | [TK]D-Fender: No, of course not. |
| 23:55.53 | *** join/#asterisk e4 (n=e4@76.84.81.72) |
| 23:56.04 | jblack | Don't blame the the twit for being a twit. Heh. :) |
| 23:56.13 | [TK]D-Fender | jblack: Yup |
| 23:56.15 | darkdrgn2k | *sigh* i wish there was a way to block all the overseas networks... |
| 23:56.22 | TJNII | Though that does answer my question of what method you think they used to crack his system... |
| 23:56.25 | jblack | to tcp, or twitdom? |
| 23:56.51 | darkdrgn2k | jblack: i think they are talking to me :-S |
| 23:56.56 | TJNII | darkdrgn2k: I have my firewall configured to only allow access to * from IPs I know. |
| 23:57.11 | darkdrgn2k | TJNII: yeh thats what my production does.. this was just a toy... |
| 23:57.16 | darkdrgn2k | TJNII: learned my lesson.. |
| 23:57.59 | TJNII | My * box used to be naked to the wind, too, until I started getting brute-force SIP attacks. |
| 23:58.15 | jblack | hrmmm. at some point I went from 1.4 to 1.6 without realizing it |
| 23:58.19 | darkdrgn2k | TJNII: yeh well... i got 160 phones calls within 60 minutes :-S |
| 23:58.34 | jblack | no I didn't. :) |
| 23:58.38 | darkdrgn2k | TJNII: so block inomming 5060? |
| 23:59.16 | [TK]D-Fender | darkdrgn2k: So let me guess you basically DMZ'd your server, right? |
| 23:59.28 | TJNII | I block everything and then set exceptions for what I need. For SIP and RDP, I limit further to the ISPs of people I have granted access. |
| 23:59.33 | darkdrgn2k | [TK]D-Fender: hahah nop... worse.. |
| 23:59.51 | darkdrgn2k | [TK]D-Fender: public ip |
| 23:59.58 | TJNII | s/RDP/RTP/ |