| 00:01.10 | Moc___ | Im a BAD navigator !!! |
| 00:01.47 | paveraware | Any bug marshalls here? |
| 00:02.40 | paveraware | strange... |
| 00:02.43 | paveraware | ok bye |
| 00:03.26 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 00:03.33 | hennker | i am using the capiCD from junghanns.net, with a fairly old fritzcard pci and the binary capi drivers from avm. maybe this card doesn't support this? |
| 00:06.32 | hennker | jas_williams: but i can make 2 outgoing calls at the same time using the sip-phone |
| 00:07.51 | *** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net) |
| 00:07.51 | *** part/#asterisk zapa (~zapa@200.92.151.177) |
| 00:12.09 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 00:13.22 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 00:13.37 | hardwire | wow |
| 00:13.40 | hardwire | I love asterisk |
| 00:13.51 | Rienzilla | great... asterisk segfaults... |
| 00:13.57 | hardwire | no.. love it |
| 00:14.11 | hardwire | Rienzilla: what did you do? |
| 00:14.17 | Netgeeks | heh, hardwire |
| 00:14.23 | hardwire | I can get it to segfault if I don't configure zap chans correctly |
| 00:14.37 | hardwire | Netgeeks: totally just plugged in 100 extensions + queues for each tenants operators |
| 00:14.50 | hardwire | with secondary queues that ring even more phones if the first one times out |
| 00:14.52 | hardwire | I loff it. |
| 00:14.53 | Rienzilla | hardwire: I'm trying to get chan_misdn to work |
| 00:15.07 | hardwire | Rienzilla: you have fun with that :) |
| 00:15.22 | Rienzilla | why not? |
| 00:15.32 | Netgeeks | I don't know squatt about isdn either |
| 00:15.37 | hennker | jas_williams: well, thx for your help. i have to go to bed now |
| 00:15.38 | hardwire | one.. I don't have an standard ISDN line anywhere :) |
| 00:15.38 | hennker | gn8 |
| 00:16.26 | Rienzilla | well |
| 00:16.30 | Rienzilla | I don't either |
| 00:16.42 | Rienzilla | but I have ISDN equipment |
| 00:17.09 | Rienzilla | so I'm trying to let a PC talk isdn, and throw the calls to a iax provider :) |
| 00:17.19 | hardwire | Rienzilla: hah |
| 00:17.26 | hardwire | I take it you are very very very bored. |
| 00:17.37 | Rienzilla | uhm... no |
| 00:17.50 | Rienzilla | is there a better way? :) |
| 00:17.54 | hardwire | you have phones on an ISDN network.. and you are wanting to work with things that way right? |
| 00:18.02 | hardwire | like an ISDN interface to a PBX |
| 00:18.07 | Rienzilla | yep |
| 00:18.10 | hardwire | gotcha |
| 00:18.22 | hardwire | and its an 8 channel ISDN? |
| 00:18.27 | Rienzilla | no |
| 00:18.34 | Rienzilla | it's just a 1 channel HFC card |
| 00:18.38 | hardwire | weird |
| 00:18.40 | hardwire | what PBX is it? |
| 00:18.50 | Rienzilla | no no |
| 00:18.58 | hardwire | yes yes |
| 00:19.15 | Rienzilla | it's just asterisk with an isdn card in NT mode |
| 00:19.18 | hardwire | time to pay the piper for some voice prompts |
| 00:19.32 | Rienzilla | and some isdn phones hooked up to it |
| 00:19.39 | hardwire | 4? |
| 00:19.44 | hardwire | or just 1 channel |
| 00:19.44 | Rienzilla | cuirrently, 1 |
| 00:19.55 | hardwire | 4 total right? |
| 00:19.58 | hardwire | if you wanted |
| 00:20.03 | Rienzilla | I guess so |
| 00:20.07 | Rienzilla | not sure |
| 00:20.18 | hardwire | brb.. |
| 00:20.23 | blitzrage | psst |
| 00:20.25 | hardwire | I need to play with my USB HID device |
| 00:20.32 | hardwire | and make it pull the PTT |
| 00:21.15 | blitzrage | pssst |
| 00:21.23 | hardwire | what? |
| 00:21.41 | blitzrage | nothin', just wondering if anyone would react :) |
| 00:21.41 | Rienzilla | what is it |
| 00:21.45 | Rienzilla | lol |
| 00:21.54 | blitzrage | heh |
| 00:21.55 | SkramX | ~slashdot |
| 00:22.06 | hardwire | heh |
| 00:22.45 | blitzrage | just found out the first 50 people to register for Astricon in Anaheim get a free IAXy |
| 00:22.59 | drumkilla | holy crap |
| 00:23.06 | blitzrage | drumkilla: its true! :) |
| 00:23.26 | *** join/#asterisk eSmoke (~johnnysmo@modemcable155.52-130-66.mc.videotron.ca) |
| 00:23.27 | drumkilla | I don't think i qualify :) |
| 00:23.32 | blitzrage | drumkilla: lol, me either |
| 00:23.42 | *** join/#asterisk ManxPower (~eric@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 00:24.26 | Rienzilla | bweuh |
| 00:24.33 | Rienzilla | I hate it when programs segfault |
| 00:24.53 | hardwire | brb |
| 00:28.31 | *** join/#asterisk laserfox (~jimbob@81-179-85-198.dsl.pipex.com) |
| 00:28.48 | laserfox | any quick tips to improve latency generally? |
| 00:29.10 | laserfox | hello :) |
| 00:29.26 | blitzrage | get a better network : |
| 00:29.27 | blitzrage | :) |
| 00:29.35 | shido | dont use a hub? |
| 00:29.41 | laserfox | is x-lite just a really bad softphone for latnecy? |
| 00:29.47 | blitzrage | nope, not really |
| 00:29.49 | shido | dont use 2.6 in vmware when running asterisk in production? |
| 00:29.54 | blitzrage | shido: lol |
| 00:29.58 | laserfox | lol |
| 00:30.00 | mishehu | don't use an 8 year old north korean child laborer to relay your packets |
| 00:30.07 | laserfox | lmao |
| 00:30.32 | laserfox | come on guys.. seriously :) |
| 00:30.51 | blitzrage | don't use RFC 1149 |
| 00:30.54 | shido | stop re-encoding porin in dr divx on your production server? |
| 00:31.03 | shido | thats right, porin - |
| 00:31.05 | shido | dman keyboard |
| 00:31.09 | laserfox | lol |
| 00:31.42 | shido | xlite works if you have a decent mic |
| 00:31.45 | blitzrage | laserfox: generally, your latency is what it is once it hits your Internet connection - after that, you have little -> no control |
| 00:32.03 | laserfox | I am bidding on a nice looking phone on ebay.. gonna be interesting comparing the latency |
| 00:32.08 | blitzrage | laserfox: do a traceroute to google and see what your latency is between hops |
| 00:32.08 | shido | tell your isp to stop randomly unplugging and plugging peoples net connections |
| 00:32.28 | blitzrage | laserfox: check your latency now - I doubt a new phone is going to do anything to reduce latency |
| 00:32.47 | shido | cancel the gremlin cat-5 double dutch contest in the data center |
| 00:32.53 | blitzrage | lol |
| 00:33.11 | blitzrage | wow, I need new speakers so badly |
| 00:33.29 | blitzrage | anyone got any 6 ohm speakers for an AIWA stereo? |
| 00:33.29 | Rienzilla | it is reported that shouting foul words to your softphone reduces latency |
| 00:33.44 | blitzrage | yes... 6 ohm (I know, its crazy) |
| 00:33.58 | Rienzilla | just hook 8 ohms up to them |
| 00:34.20 | laserfox | it might take a few mins to compile traceroute |
| 00:34.29 | blitzrage | Rienzilla: I'd rather put 4 ohm speakers and add a 2 ohm resistor in series |
| 00:34.37 | laserfox | or is it tracert |
| 00:34.39 | Rienzilla | wtf :) |
| 00:34.48 | blitzrage | traceroute in linux, tracert in windows |
| 00:34.51 | Rienzilla | put a resistor in your signal path after the amplifier :)) |
| 00:34.58 | eSmoke | Hi, Anybody has an idean where I can found Citel Button Mapping documentaion? |
| 00:35.06 | Rienzilla | it's never a problem to connect speakers with a higher impedance |
| 00:35.09 | laserfox | i thought so |
| 00:35.09 | Rienzilla | afaik |
| 00:35.25 | Rienzilla | only a lower impedance may break things, as the amplifier might not be able to handle it |
| 00:35.41 | blitzrage | Rienzilla: yep, you're right. Either way, I still need new speakers for cheap :) |
| 00:35.56 | blitzrage | already had one speaker fall off a high shelf and explode :) |
| 00:35.59 | Rienzilla | I bought some very nice speakers for EUR 100 |
| 00:36.05 | Rienzilla | or is that not 'cheap' :) |
| 00:36.22 | blitzrage | that's not cheap :) I was kind of hoping someone would donate some :D |
| 00:36.26 | Rienzilla | lol |
| 00:36.43 | Rienzilla | I've got some sonys here but the shipping costs is probably more than they're worth :) |
| 00:36.50 | *** join/#asterisk SuperMMan (TestMasTer@d198-53-32-174.abhsia.telus.net) |
| 00:37.00 | blitzrage | yah, thats always a bitch eh? |
| 00:37.04 | Rienzilla | yep |
| 00:37.14 | blitzrage | anyways, I'm going to check out the Jays ball game - back latah |
| 00:37.18 | Rienzilla | see ya |
| 00:37.20 | SuperMMan | hello all question, i want to register someone in to sip. but with out a username or password. just their ip address. can i do that? |
| 00:37.21 | blitzrage | remember... free iaxy's! :) |
| 00:38.02 | Rienzilla | *brr* |
| 00:38.10 | laserfox | ok.. heres a trace route.. can anyone help me understand it ? :) http://pastebin.com/328004 |
| 00:38.33 | SuperMMan | anyone? |
| 00:38.44 | Mavvie | SuperMMan: just don't add a password, and add insecure=yes |
| 00:39.06 | SuperMMan | Mavvie ok but doing that you still need a username right? |
| 00:40.43 | Mavvie | the one between []'s? yes |
| 00:40.55 | SuperMMan | shoot... how do I get away from that? |
| 00:41.13 | SuperMMan | or will they need to use.. the username on there end? |
| 00:41.19 | Mavvie | you need it in your sip-configuration, you don't need it on the remote side. |
| 00:41.29 | SuperMMan | ok great thanks |
| 00:42.03 | SuperMMan | one more question i am trying to make a call i keep getting Aug 2 19:20:15 WARNING[13452]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 324991757f33b70a509e155c71cf8283@192.168.1.101 for seqno 102 (Critical Response) |
| 00:42.08 | SuperMMan | and idea why? |
| 00:42.15 | Mavvie | not me |
| 00:42.38 | *** join/#asterisk jaxxan (~Snak@202.70.125.109) |
| 00:42.40 | Rienzilla | something went wrong and it retried it some times and it still went wrong :) |
| 00:43.02 | SuperMMan | that doesn`t help much :-p |
| 00:43.22 | jaxxan | hey ya'll |
| 00:43.42 | Rienzilla | I know but its all I can say from your info :) |
| 00:44.08 | SuperMMan | ya i know... Rienzilla when it hits the other box thats all its sending... it is not even trying to dial a phone number |
| 00:45.30 | BhaalWK | Hey, just wondering, now that the Skype linux SDK is out, do we know if anyone is working on an asterisk <-> skype setup? |
| 00:53.08 | *** join/#asterisk colinm_ (~colinmatt@VDSL-130-13-9-155.PHNX.QWEST.NET) |
| 00:53.56 | laserfox | cya :) |
| 00:54.27 | *** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985234.sympatico.ca) |
| 00:57.19 | SuperMMan | does anyone know how to resolve this error WARNING[17274]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 21a59a311add17af28c98a571586a49c@192.168.1.101 for seqno 102 (Critical Response) |
| 01:01.28 | hardwire | damn |
| 01:01.33 | hardwire | that was a waste of an hour of my life |
| 01:01.54 | hardwire | found all the HID events for controlling a radio shark.. and a dingotel USB PTT audio device |
| 01:01.58 | hardwire | but I can't seem to set anything |
| 01:02.21 | *** join/#asterisk asteriskmonkey (~phil@69.158.154.80) |
| 01:02.28 | hardwire | monkey monkey |
| 01:02.36 | asteriskmonkey | :D hey hardwire |
| 01:03.04 | asteriskmonkey | anyone got this error before : Ouch ... error while writing audio data: : Broken pipe |
| 01:03.58 | *** join/#asterisk outtolunc (outtolunc@adsl-69-110-15-184.dsl.pltn13.pacbell.net) |
| 01:04.12 | hardwire | doh |
| 01:05.55 | asteriskmonkey | ive never got it before.. my box was working fine i updated some things non asterisk related and when i rebooted asterisk would not go anymore :( |
| 01:06.24 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 01:06.29 | hardwire | anybody have experience with "Local" channels |
| 01:06.33 | mishehu | yah broke the bloody ship! |
| 01:06.44 | hardwire | cause I can't seem to use Local/1111@blahcontext from duhcontext |
| 01:09.38 | hardwire | ah |
| 01:09.46 | hardwire | I should but Dial in that extension |
| 01:09.52 | hardwire | :P |
| 01:10.36 | *** join/#asterisk kingtux (~susekid@pool-151-196-126-24.balt.east.verizon.net) |
| 01:10.56 | kingtux | Whats up my fellow asteriskers |
| 01:10.57 | kingtux | lol |
| 01:11.52 | syle2 | asteriskmoney : kernel i bet |
| 01:11.54 | kingtux | Anyone of a credit card system that intergrates with asterisk |
| 01:12.15 | asteriskmonkey | i didnt update the kernel though :P so im lost |
| 01:12.25 | syle2 | what did you update |
| 01:12.42 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 01:12.44 | asteriskmonkey | just php , mysql cups little things |
| 01:12.52 | asteriskmonkey | source code for kernal but no recompile etc.. |
| 01:12.56 | kingtux | I mean I have a credit card system running on asterisk |
| 01:13.21 | kingtux | I was looking for a app that can intergrate to it so customers can refil thier cards |
| 01:13.37 | *** join/#asterisk file[laptop] (~file[lapt@69.158.162.238) |
| 01:13.39 | asteriskmonkey | i can write you a php on kintux |
| 01:14.06 | kingtux | Is there anything already out there |
| 01:14.07 | kingtux | ?? |
| 01:14.16 | syle2 | ummm |
| 01:14.18 | asteriskmonkey | i can write you a web based one |
| 01:14.20 | syle2 | you touched the kernel |
| 01:14.26 | asteriskmonkey | ah |
| 01:14.32 | asteriskmonkey | how do i back it back then |
| 01:14.34 | syle2 | ever think asterisk might be linking against those sources |
| 01:14.35 | asteriskmonkey | i am usign centos |
| 01:14.50 | syle2 | well easiest way... |
| 01:14.57 | file[laptop] | SO |
| 01:15.01 | kingtux | I'm using centos as well running AHH for card system |
| 01:15.02 | file[laptop] | I got stranded in Toronto |
| 01:15.12 | asteriskmonkey | im in toronto |
| 01:15.28 | syle2 | finish off your kernel, and recompile zaptel drivers and install, then remove all current asterisk modules, then reinstall asterisk |
| 01:15.49 | asteriskmonkey | damn thats not easy at all |
| 01:15.55 | kingtux | ?? |
| 01:15.55 | Rienzilla | rats |
| 01:15.58 | Rienzilla | damn misdn |
| 01:15.58 | asteriskmonkey | would be quicker to bomb it lol |
| 01:15.59 | kingtux | who u talking to |
| 01:16.07 | syle2 | wtf that would take like 10 min |
| 01:16.13 | asteriskmonkey | really? |
| 01:16.19 | kingtux | yeah did it last night |
| 01:16.27 | kingtux | took me 5 min |
| 01:16.46 | kingtux | on slackware |
| 01:16.47 | syle2 | well depending how fast your processor is i guess |
| 01:16.50 | asteriskmonkey | mind walking me through it |
| 01:16.58 | asteriskmonkey | mine is a 2800xp chip |
| 01:17.01 | syle2 | if you got a 200mhz machine could take longer |
| 01:17.05 | asteriskmonkey | lol |
| 01:17.12 | kingtux | i have a p4 2.8gz |
| 01:17.40 | syle2 | fedora is the best i think for kernel upgrades |
| 01:17.46 | syle2 | yum upgrade kernel |
| 01:17.47 | syle2 | about it |
| 01:17.53 | syle2 | then reinstall zaptel and asterisk |
| 01:18.16 | syle2 | sorry |
| 01:18.22 | syle2 | yum install kernel-devel to hehe |
| 01:18.32 | asteriskmonkey | ok thanks |
| 01:18.51 | syle2 | after you reboot |
| 01:19.36 | dijungal | can i get a flat rate VOIP into the US..? |
| 01:19.51 | syle2 | 1.3 cents US |
| 01:20.41 | *** part/#asterisk colinm_ (~colinmatt@VDSL-130-13-9-155.PHNX.QWEST.NET) |
| 01:20.42 | syle2 | unless you got alot of cash to put down then you can get it at like .008 cents or .006 cents |
| 01:21.53 | kingtux | WHo is the cheapest VOIP provider in the US |
| 01:22.34 | Cybertoy | I use broadvoice.com ... and I'm happy with them |
| 01:22.46 | Cybertoy | I have the international world plus plan ... flat fee to many countries. |
| 01:22.52 | kingtux | screw BV |
| 01:22.54 | kingtux | they suck |
| 01:23.05 | Cybertoy | their service sux... |
| 01:23.13 | Cybertoy | I mean customer service |
| 01:23.20 | Cybertoy | but other than that I don't have a problem with them. |
| 01:23.20 | kingtux | I had them...They were my 1st voip provider when i started |
| 01:23.31 | kingtux | lines always down |
| 01:23.32 | kingtux | up |
| 01:23.33 | Cybertoy | they had massive problems in May |
| 01:23.33 | kingtux | don |
| 01:23.44 | kingtux | What about now?? |
| 01:23.46 | Cybertoy | but I'm very stable since mid June. |
| 01:23.52 | kingtux | How long u been with them |
| 01:23.57 | Cybertoy | a year. |
| 01:24.11 | syle2 | kingtux its not as black and white as that, you find the cheapest ones, try them out , and pick one that is reliable |
| 01:24.28 | Cybertoy | I would agree with syle.. |
| 01:24.38 | Cybertoy | you get what you pay for... |
| 01:24.45 | kingtux | Man its just nutts how many voip providers are poping up everyday |
| 01:24.52 | Cybertoy | I also have voxee, voipjet, and voipbuster... |
| 01:24.57 | kingtux | i have telasip right now |
| 01:25.09 | Cybertoy | oh .. and stanaphone ... |
| 01:25.13 | syle2 | hmmm haven;t tried voxee and voipbuster how are they? |
| 01:25.17 | Cybertoy | I like them as they're free and give you incoming fax... |
| 01:25.29 | Cybertoy | voipbuster seems to be having problems at the moment... |
| 01:25.33 | kingtux | yup |
| 01:25.36 | kingtux | can't call out |
| 01:25.38 | Cybertoy | but they have many countries for free. |
| 01:25.45 | Cybertoy | so once they work they're fine... |
| 01:25.52 | Cybertoy | voxee I'm not really using that much. |
| 01:25.54 | kingtux | yeah |
| 01:25.58 | syle2 | i haven;t had much problems with voipjet, and my callerid always comes up as i set it with them which is good |
| 01:26.08 | hardwire | heh |
| 01:26.10 | kingtux | what kind o plan |
| 01:26.14 | kingtux | for voipjet |
| 01:26.17 | Cybertoy | yeah .. same here... |
| 01:26.21 | Cybertoy | pay as you go |
| 01:26.22 | kingtux | are they sip |
| 01:26.27 | syle2 | iax |
| 01:26.28 | Cybertoy | iax and sip |
| 01:26.30 | hardwire | call 907-762-8400 |
| 01:26.34 | Cybertoy | actually not sure about sip |
| 01:26.37 | hardwire | thats my new pbx intro |
| 01:27.02 | hardwire | tried to use the AT&T Voice - Mike |
| 01:27.27 | hardwire | 8003779807? |
| 01:27.34 | hardwire | I know your number! |
| 01:27.39 | hardwire | I KNOW ALL YOUR NUMBERS! |
| 01:28.01 | kingtux | so |
| 01:28.23 | kingtux | what kind of biz |
| 01:28.26 | kingtux | u running |
| 01:28.35 | *** join/#asterisk criptos (~criptos@201.145.202.198) |
| 01:28.43 | hardwire | kingtux: its an umbrella corp |
| 01:28.45 | syle2 | how much you guys paying per min on your 800 numbers? |
| 01:28.47 | criptos | hi! :) |
| 01:29.08 | syle2 | norm seems to be 2 cents a minute, wondering if you guys paying less than that anywhere |
| 01:29.30 | criptos | Does anyone know why asterisk cannot be able to read a callerid info from a panasonic switch ? |
| 01:29.31 | kingtux | where 2 cen'ts a min |
| 01:29.52 | Cybertoy | I think BV has 2c per min as well ... |
| 01:29.57 | hardwire | hmm weird |
| 01:30.03 | Cybertoy | and 60 min per month included... |
| 01:30.05 | Cybertoy | if I remember right |
| 01:30.11 | Cybertoy | I don't have a 1800 number... |
| 01:30.15 | hardwire | who called in a little earlier |
| 01:30.18 | hardwire | did you even hear prompts? |
| 01:30.24 | kingtux | Yeah I heard prompts |
| 01:30.32 | hardwire | weird |
| 01:30.34 | kingtux | why |
| 01:30.38 | syle2 | well if your luck enough to have a PRI line, then don;t cost you much at all hehe |
| 01:30.38 | kingtux | what happend |
| 01:30.50 | hardwire | its not playing MoH to my cell |
| 01:30.54 | hardwire | now it is |
| 01:30.59 | hardwire | man asterisk is being weird |
| 01:31.05 | kingtux | hmmmm |
| 01:31.10 | kingtux | what kind of biz u run |
| 01:31.12 | kingtux | ? |
| 01:31.19 | hardwire | I don't run any biz |
| 01:31.28 | hardwire | but those aren't official prompts either |
| 01:31.41 | hardwire | I wouldn't normally tell people to show some damn patience |
| 01:31.41 | kingtux | oh |
| 01:31.49 | kingtux | why |
| 01:31.55 | hardwire | cause its rude?! |
| 01:32.07 | kingtux | hmm i'ml lost |
| 01:32.28 | SuperMMan | i am having a problem i am trying to trying place a call in to another asterisk system, i have tried disabling username/password but nothing is happening |
| 01:32.31 | hardwire | kingtux: hah |
| 01:32.35 | hardwire | the prompts for the company |
| 01:32.37 | SuperMMan | it keeps giving me a 404 error |
| 01:32.42 | hardwire | it says "Please show some damn patience...." |
| 01:33.05 | kingtux | huh |
| 01:33.21 | hardwire | indeed |
| 01:34.01 | hardwire | so I should probably be using Asterisk CVS |
| 01:34.02 | hardwire | damnit |
| 01:34.22 | SuperMMan | any one with anyideas on resolving this issue |
| 01:34.31 | hardwire | the snoms can monitor |
| 01:34.38 | hardwire | and I have them set up to do so.. just asterisk doesn't grok it |
| 01:34.41 | SpaceBass | arrruuuuggg stupid email |
| 01:34.46 | hardwire | err.. record |
| 01:35.01 | *** part/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 01:35.18 | hardwire | SuperMMan: doh |
| 01:35.28 | hardwire | SuperMMan: are they peered together? |
| 01:35.39 | kingtux | So whats the deal...Does anyone know if there is a credit card appp for asterisk |
| 01:35.55 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 01:35.55 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:35.55 | hardwire | kingtux: I would think it is part of a callign card product |
| 01:36.39 | kingtux | One i'm using doesn't come with one |
| 01:38.20 | hardwire | I think I am done for the day |
| 01:39.10 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 01:42.05 | *** join/#asterisk dalabera (~dalabera@adsl-9-162-125.mia.bellsouth.net) |
| 01:42.57 | dijungal | guys |
| 01:43.12 | dijungal | which one of the codecs are under license..? |
| 01:43.26 | dijungal | and cannot be used freely...? |
| 01:43.40 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 01:44.37 | hardwire | grr |
| 01:45.05 | dijungal | ?? |
| 01:45.14 | syle2 | g729 |
| 01:45.16 | *** join/#asterisk mago2-cn (~maxglucks@200.109.166.83) |
| 01:45.20 | dijungal | ok thanx |
| 01:45.25 | *** join/#asterisk Berkey (~berkey@las-cust-208.57.3.251.mpowercom.net) |
| 01:45.40 | dijungal | yuh see... i'm on a low bandiwdth connection so i am tryin to determine which codec will work best for me.. |
| 01:45.46 | dijungal | i am on a 64k up |
| 01:46.02 | mago2-cn | hello, can anyone advise on how to change a 7905 h.323 image to a sip image? |
| 01:46.08 | syle2 | http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html |
| 01:47.18 | dijungal | i was looking at that |
| 01:47.24 | dijungal | but i had no idea what IAX2 trunking is |
| 01:47.42 | dijungal | i know IAX is the native asterisk channel or something so.. |
| 01:47.46 | dijungal | but that is truncking..? |
| 01:47.49 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 01:47.52 | dijungal | trunking |
| 01:47.55 | syle2 | http://www.voip-info.org/tiki-index.php?page=Asterisk%20-%20dual%20servers |
| 01:49.58 | syle2 | in your case i;d use gsm codec and do more reading on iax |
| 01:50.16 | dijungal | iax.. |
| 01:50.20 | Nugget | trunking is when two asterisk machines are able to send multiple voice channels over the same connection. |
| 01:50.24 | dijungal | but sip seems to over more features |
| 01:50.31 | dijungal | on the features chart |
| 01:50.32 | dijungal | :) |
| 01:50.43 | dijungal | that's why i choose to work with sip |
| 01:50.49 | syle2 | sip is good |
| 01:50.53 | Nugget | it reduces the bandwidth requirement a bit if you have multiple channels going the same place |
| 01:51.03 | syle2 | setup ain;t that much different |
| 01:51.07 | Nugget | the only problem with sip is difficulty working with nat. |
| 01:51.17 | syle2 | how so |
| 01:51.19 | Nugget | if you can avoid nat (and who doesn't want to avoid nat?) it's fine |
| 01:51.35 | Nugget | sip embeds the IPs inside the headers, so it causes all kinds of hell with nat |
| 01:51.43 | syle2 | nat=yes and qualify=yes and your fine |
| 01:51.46 | dijungal | oooh |
| 01:51.56 | Nugget | as long as nat is only on one side, sure. |
| 01:52.06 | syle2 | and how is that different from iax |
| 01:52.29 | Nugget | iax doesn't include the IPs in the protocol, as I just explained. |
| 01:52.42 | syle2 | your missing the point |
| 01:52.50 | syle2 | how does it know where to talk to begin with then |
| 01:52.54 | Nugget | no, you're just being all twitchy because you think I pissed on your shoes. |
| 01:53.05 | dijungal | lol |
| 01:53.07 | Nugget | iax is much better able to circumnavigate through nat. |
| 01:53.10 | syle2 | lol |
| 01:53.12 | dijungal | ok ok guys... i'm the one that needs help here.. :P |
| 01:53.31 | syle2 | it doesn;t circumvent nat it cuts out the 3rd party, thats why it is kewl |
| 01:53.34 | syle2 | big difference |
| 01:53.40 | Tuneman | anyone use astrisk@home? |
| 01:53.40 | Nugget | because iax doesn't internalize the ip addresses like sip does, so it doesn't get damaged with the addresses are translated |
| 01:54.08 | Nugget | what do you mean "cuts out the 3rd party"? they're both just protocols. your statement isn't relevant. |
| 01:54.23 | dijungal | ok.. does IAX reinvite the two phones.. so they communicate directly with each other instead of through the server..? |
| 01:55.06 | Nugget | iax and sip both have reinvite mechanisms. |
| 01:55.07 | syle2 | how does that mean anything, they map an open udp connection to transverse the gateway |
| 01:55.15 | syle2 | simple as that |
| 01:55.20 | Nugget | no, it's not that simple, syle. |
| 01:55.42 | dijungal | oooh boi..:| |
| 01:56.02 | Nugget | if both endpoints are behind nat, sip falls apart |
| 01:56.16 | syle2 | so would iax |
| 01:56.22 | Nugget | and the protocol is inherently unfriendly to nat, since it tries to self-manage ip addresses |
| 01:56.40 | Nugget | no, with iax you can reliably do port forwarding through nat if you want to. |
| 01:56.51 | syle2 | again how is peer 1 going to talk to peer 2 if he is behind nat to begin with |
| 01:56.58 | dijungal | :| |
| 01:57.11 | syle2 | you need a public ip |
| 01:57.24 | Nugget | or a port forward. |
| 01:57.30 | syle2 | all the discussions on port forwarding aren;t bs you know |
| 01:57.45 | Nugget | what? |
| 01:58.02 | syle2 | my point is.... |
| 01:58.14 | syle2 | if your port forwarding , sip can take over at that point to |
| 01:58.23 | syle2 | you;ve not given one good reason yet |
| 01:58.30 | Nugget | sure i have. |
| 01:58.43 | Netgeeks | Not given 1 good reason for what? using IAX? |
| 01:58.47 | Nugget | sip embeds the ip addresses within its own protocol, which leads to confusion and disconnects when the ip address is translated. |
| 01:58.51 | Nugget | that is a reason. |
| 01:59.01 | mishehu | bah. |
| 01:59.03 | syle2 | there are good reasons netgeeks, he just doesn;t understand how iax really works |
| 01:59.12 | Netgeeks | TRUNKING |
| 01:59.12 | Nugget | nat also causes problems for sip because both the sip signalling and the media use udp as transport. |
| 01:59.18 | Netgeeks | thats my one good reason |
| 01:59.39 | dijungal | IAX also used udp..:P |
| 01:59.51 | Nugget | I can tell when syle stops talking TO me and starts talking ABOUT me that he's run out of arguments. |
| 01:59.54 | mishehu | Nugget: that's not actually the reason. it's because the IP information is encoded in the SIP part of the packet. |
| 02:00.03 | Nugget | isn't that what I just said? |
| 02:00.05 | mishehu | whether or not it's udp is irrelevant. |
| 02:00.14 | Nugget | the udp aspect is a second issue. |
| 02:00.36 | mishehu | actually there is no problem with it being udp w/regards to nat. if it is, please explain how. |
| 02:00.42 | Netgeeks | And you really have to have disdain for any protocol that breaks the layer barriers |
| 02:00.53 | mishehu | if I am wrong, I'd like to know how. |
| 02:00.54 | Nugget | do you agree with syle that sip is equivalent to iax when traversing nat? |
| 02:00.55 | Netgeeks | even if it's a good protocol :) |
| 02:01.02 | Juggie | udp does not have any problem with nat |
| 02:01.17 | Juggie | any good nat device leaves the source port open for return traffic |
| 02:01.27 | syle2 | exactly!@!!! |
| 02:01.31 | mishehu | which is a conntrack |
| 02:01.38 | syle2 | that is what i am trying to get through to him lol |
| 02:01.41 | Juggie | no its not conntrack its jus good udp |
| 02:01.49 | Nugget | syle2: you are still fundamentally wrong. (lol) |
| 02:01.50 | Juggie | when a nat device sends a udp packet |
| 02:02.03 | Juggie | for x number of seconds the source port of that packet is open for return traffic |
| 02:02.05 | Nugget | iax is better able to traverse nat and sip combined with nat is notoriously problematic. |
| 02:02.15 | Nugget | you've done nothing to disprove that |
| 02:02.22 | dijungal | ok ok.. hear's the deal... i have a low bandwidth connection (64k) to my server (T1), all i wanted to know is WHAT"S THE BEST WAY TO DELIVER calls over this low bandwidth to my server.. :P |
| 02:02.23 | Juggie | that is why qualify=yes helps with nat |
| 02:02.30 | dijungal | lawd.... i din ask for a fight on it |
| 02:02.37 | Juggie | because it keeps nat udp mappings alive |
| 02:02.43 | syle2 | nugget how about i run many sip phones with options juggie just said and works just fine? |
| 02:02.53 | Nugget | I said that iax is superior to sip for nat environments and syle is acting like I called his kid ugly. |
| 02:02.56 | Netgeeks | dijungal: okay, do you have an asterisk server on both ends of the path? |
| 02:03.06 | dijungal | noep |
| 02:03.07 | dijungal | nope |
| 02:03.13 | Juggie | syle2 & nugget, sip has no problems with nat if you understand the protocol and your nat devices are good. |
| 02:03.16 | dijungal | i have a software sipphone.. x-lite |
| 02:03.17 | Netgeeks | okay, on one end? |
| 02:03.21 | Juggie | i run my server with double nat |
| 02:03.23 | Juggie | nat on both sides |
| 02:03.24 | dijungal | and an asterisk server in the US on a t1 connection |
| 02:03.27 | Juggie | and it runs perfectally |
| 02:03.43 | Netgeeks | well, you don't have much of an option, SIP for the protocol |
| 02:03.49 | Juggie | you just have to understand the protocol |
| 02:03.51 | dijungal | but most of my friends here.. have my type of connection.. cable... i want them to be able to use their shitty bandwidht to make calls on my server.. |
| 02:03.55 | Netgeeks | and at 64k gsm speex, or g729 |
| 02:04.14 | dijungal | ok THANK YOU netgeeks... |
| 02:04.22 | Juggie | i have a sip phone here right now, behind nat, and my server is behind nat |
| 02:04.25 | Juggie | and it works just fine |
| 02:04.25 | Nugget | Juggie: do you not have sip clients within the nat as well? I can see how you could make it work if the server only ever had to talk to external addresses. |
| 02:04.26 | dijungal | that's all i needed to know.. so now i know where to direct my study.. :) |
| 02:04.26 | Netgeeks | g729 will use about 20kbps per call , gsk around 32, and speex, I forget |
| 02:04.33 | Netgeeks | gsm, even |
| 02:04.37 | Juggie | Netgeeks, wrong. |
| 02:04.40 | Nugget | or can asterisk now spoof the external ip only when needed? |
| 02:04.51 | syle2 | iax is superior yes(in asterisk), but in the real world it isn;t , if your hooking up a to your local telephone company;s tandem switches you aren;t going to be using iax trust me, so sip and SER etc get the real world attention |
| 02:05.01 | shido | read your private messages dijungal |
| 02:05.03 | Juggie | Nugget, * doesn't need to spoof the external ip for the udp packat |
| 02:05.07 | dijungal | problem is.. it's hard to find a free phone using g729 |
| 02:05.14 | Juggie | the nat device will set the source ip & port to something |
| 02:05.21 | Juggie | it will essentially rewrite the packet |
| 02:05.27 | Nugget | eww :) |
| 02:05.28 | Netgeeks | Juggie: what was wrong about my statments? |
| 02:05.39 | Nugget | isn't the internal ip leaked out within the sip header? |
| 02:05.50 | Juggie | g729 is 8kbps per call, but overhead is much more then 12kbps |
| 02:05.56 | Nugget | or do you mean actual protocol-level rewriting? I used to use that for dns on my cisco -- it was wonky but effective |
| 02:06.16 | Juggie | Nugget, yes asterisk allows you to set an external ip within sip.conf |
| 02:06.25 | Juggie | and then you setup which networks are internal as well |
| 02:06.49 | Juggie | so it uses externip=? only when the dest network != to one of the internal networks |
| 02:06.49 | syle2 | nugget why you even discussing iax if your are at all familiar with things like cisco and lucent TNT configurations? |
| 02:06.52 | Nugget | at one time it used to be that setting the externalip broke connectivity to sip endpoints which were also behind the nat |
| 02:07.03 | *** join/#asterisk SplasPood (~jwb@brooklyn.paravolve.net) |
| 02:07.10 | Netgeeks | Okay, my iftop indicated a single g729 call from a 7960 to asterisk took up right around 20kbps |
| 02:07.12 | Juggie | well i can assure you it works |
| 02:07.17 | Nugget | syle2: why are you so fucking eager to argue with me? do I smell funny or something? |
| 02:07.17 | dijungal | but g729 has licensing issues right... |
| 02:07.22 | dijungal | so i canot freely use it |
| 02:07.23 | Nugget | you act like this is a personal attack |
| 02:07.44 | Netgeeks | dijungal, yes, for using g729 on asterisk, you would need to get some licenses |
| 02:07.45 | Juggie | Netgeeks, dont forget all the network overhead you dont see on that |
| 02:08.22 | dijungal | ok so that leaves speex and gsm |
| 02:08.23 | syle2 | nugget hardly personaly, you should see a councellor about those issues, i;m just stating whats what |
| 02:08.28 | Netgeeks | I don't remember off the top of my head whether or not xten lite supports ilbc or not |
| 02:08.31 | dijungal | or g.723 |
| 02:08.33 | Juggie | it does |
| 02:08.40 | Juggie | it supports ilibc |
| 02:08.46 | dijungal | yes it does.. |
| 02:08.57 | Nugget | syle2: I'm just getting tired of you turning a discussion about network protocols into speculation about my person. It's unneccesarry and inflammatory. |
| 02:08.58 | Netgeeks | ilbc then dig.. it should be around the same bandwidth load as g729 |
| 02:09.05 | Juggie | Netgeeks, see http://www.voip-info.org/tiki-index.php?page=Bandwidth+consumption |
| 02:09.05 | *** join/#asterisk BhaalWK (bhaal@bhaal.staff.freenode) |
| 02:09.39 | *** join/#asterisk ManxPower (~eric@adsl-065-082-224-226.sip.msy.bellsouth.net) |
| 02:10.01 | dijungal | k |
| 02:10.07 | dijungal | <PROTECTED> |
| 02:10.08 | dijungal | <PROTECTED> |
| 02:10.08 | dijungal | <PROTECTED> |
| 02:10.08 | dijungal | <PROTECTED> |
| 02:10.08 | dijungal | <PROTECTED> |
| 02:10.08 | dijungal | <PROTECTED> |
| 02:10.10 | dijungal | <PROTECTED> |
| 02:10.12 | dijungal | <PROTECTED> |
| 02:10.13 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 02:10.14 | dijungal | <PROTECTED> |
| 02:10.14 | Nugget | if you need an outlet for your rage, perhaps you could channel it into correcting the dozens of pages in the wiki which disagree with your position. |
| 02:10.15 | dijungal | Speex * |
| 02:10.18 | dijungal | gsm - 13Kbps * |
| 02:10.18 | *** join/#asterisk Barmal (~info@c-24-30-75-206.hsd1.ga.comcast.net) |
| 02:10.19 | dijungal | i added gsm |
| 02:10.50 | syle2 | nugget again i was attacking your discussion not you personally, there are many good councellors available talk with one |
| 02:10.53 | dijungal | and speex.. but i do not know the kbps for speex |
| 02:10.54 | ManxPower | ~pastbin |
| 02:11.00 | Nugget | if SIP to SIP does indeed work just fine when both endpoints are behind NAT, then there's plenty of places where the documentation is inaccurate. |
| 02:11.07 | ManxPower | dijungal, use patebin to post stuff that would flood the channe |
| 02:11.13 | Barmal | what does debug mean by "caller from user 'xxxx.' is 1 of 0? |
| 02:11.36 | ManxPower | Barmal, sounds like you are running with debugging enabled. |
| 02:11.42 | twisted | Barmal, that you shouldn't be running with debugging enabled |
| 02:11.42 | Barmal | yep |
| 02:12.00 | Netgeeks | The whole problem generalizing talk about NAT devices is that proably somewhere around half of all NAT implementations do goofy stuff |
| 02:12.02 | jontow | speex is 'adjustable' to some extent.. |
| 02:12.07 | dijungal | manxpower: ok |
| 02:12.11 | Barmal | twisted :) |
| 02:12.21 | dijungal | pastebin.com ? |
| 02:12.27 | Netgeeks | you might be able to make a SIP -> NAT <-> NAT <-> SIP work on one nat device and fail on another |
| 02:12.51 | Netgeeks | the same goes for IAX |
| 02:13.14 | Nugget | Can we all agree that NAT blows goats at least? |
| 02:13.24 | Zaw | NAT rocks. |
| 02:13.25 | syle2 | not really |
| 02:13.27 | syle2 | nat rocks |
| 02:13.31 | Nugget | figures |
| 02:13.45 | dijungal | lol |
| 02:13.48 | dijungal | if it works properly.. |
| 02:13.49 | ManxPower | NAT works fine for me, even roaming between subnets |
| 02:13.57 | Juggie | Netgeeks, with the proper configuration it will work fine |
| 02:14.06 | ManxPower | (well roaming between "behind the nat" and "not behind the nat") |
| 02:14.08 | Netgeeks | haha, For tyring to set up a reliable SIP or IAX connection travelling between two unknown nat devices, yes, NAT bites |
| 02:14.12 | Juggie | i have yet to find someone who doesnt work with my * configuration |
| 02:14.20 | Qwell | Anybody know how I can get ahold of oej? |
| 02:14.24 | Juggie | and i need to do not specific port mapping |
| 02:14.24 | ManxPower | Netgeeks, why would I want to do that? |
| 02:14.39 | ManxPower | Netgeeks, unless I was a service provider -- in which case I would use SER. |
| 02:14.55 | Netgeeks | That was what the discussion started out as Manx.... SIP - NAT - NAT - SIP |
| 02:15.05 | Juggie | which works fine |
| 02:15.11 | *** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net) |
| 02:15.19 | ManxPower | I am actually on that EXACT setup right now. |
| 02:15.47 | dijungal | what's the echo channel for FWD again..? |
| 02:15.48 | ManxPower | SIPura at hotel -> Hotel Nat -> Internet -> Cisco NAT -> Asterisk |
| 02:15.51 | Juggie | i have that exact setup in production |
| 02:15.57 | dijungal | 614 ..? |
| 02:16.06 | syle2 | cisco nat fun |
| 02:16.15 | Netgeeks | And if you didn't control either NAT devices? |
| 02:16.19 | dijungal | i wanna hear myself... |
| 02:16.32 | Nugget | I've had SIP fail me way more often than work for me when in hotels. I run asterisk on my powerbook because it's more reliable to sip to a local asterisk and then send iax to my asterisk server. |
| 02:16.42 | ManxPower | Netgeeks, You can't have Asterisk behind a NAT device that you don't control in this setup. |
| 02:16.50 | Netgeeks | right |
| 02:16.51 | Nugget | but hotel ethernet is the polar opposite of sane networking, generally. |
| 02:17.15 | ManxPower | Wait! |
| 02:17.28 | ManxPower | SIPura NAT at hotel -> Hotel Nat -> Internet -> Cisco NAT -> Asterisk |
| 02:17.46 | ManxPower | so I'm actually SIP->NAT->NAT->Internet->NAT->Asterisk |
| 02:17.51 | Nugget | ouch |
| 02:18.01 | ManxPower | but I guess that's doesn't really count since the SIPura is doing the NAT |
| 02:18.13 | syle2 | put a asterisk box as hotel nat, let them pay bw charges |
| 02:18.29 | Netgeeks | isn't that like plugging a UPS into a UPS? Should space-time collapse and fold in upon itself somewhere in there, MAnx? |
| 02:18.32 | ManxPower | so I'm actually Laptop->SPA-2100->NAT->Internet->NAT->Asterisk |
| 02:19.08 | ManxPower | Netgeeks, naw, since most hotels seem to auth on mac after you auth via a web broswer, all packets are coming from the same IP/MAC so the hotel stuff works |
| 02:19.12 | syle2 | maxpower, haven;t tried spa ATA's but are the 2100 and 2000 just as good? |
| 02:19.39 | Netgeeks | but in reality, your Asterisk is only physically behind a NAT device, and the approrpiate ports are being forwarded? |
| 02:19.57 | Rienzilla | lol |
| 02:20.14 | ManxPower | syle2, The SPA-2100 is pretty much a combination of a SPA-2000 (like Cisco ATA-186) and a Linksys BESR41 with only one lan port and 1 wan port and 2 fxs |
| 02:20.49 | ManxPower | Netgeeks, Yes. porwarding 5060 and the RTP ports (16384-16394). |
| 02:21.42 | syle2 | so they are both just as reliable just 2100 has extra ports |
| 02:21.47 | ManxPower | Now, if you need to do SIP reinvites between SIP devices behind NAT then you'll have issues if you don't use something like SER |
| 02:21.58 | Netgeeks | Now what should work is a re-invite where a central public SIP device acts as a middle man for startup of a call between two true natted UA's... That should work just fine |
| 02:22.00 | ManxPower | syle2, reliable as what? |
| 02:22.13 | ManxPower | Netgeeks, That's called SER 8-) |
| 02:22.27 | Netgeeks | Or Asterisk as well |
| 02:22.37 | ManxPower | Netgeeks, no, asterisk is not a SIP proxy. |
| 02:22.52 | syle2 | manxpower i;ve never understood that, at what point can asterisk itself not handle multiple invites, 10 devices on? |
| 02:22.56 | Netgeeks | canreinvite=yes still has asterisk step out of the udp stream |
| 02:23.06 | ManxPower | Asterisk also cannot do the rewriting of packets required to make reinvites work netween different devices behind different nats |
| 02:23.31 | syle2 | i beleive yes SER is installed usually for main reason that it is faster at handling the packets than asterisk |
| 02:23.55 | Rienzilla | Yeah! |
| 02:24.01 | Rienzilla | got misdn to work |
| 02:24.06 | ManxPower | syle2, Asterisk cannot do the packet rewriting needed to make it work with two devices behind different nats that need to reinvite |
| 02:24.32 | Netgeeks | right, the wrong ip addresses get sent in the re-invite... |
| 02:24.36 | Netgeeks | bah |
| 02:24.40 | ManxPower | As far as I can tell, most of SER configuration is done is what is basically a language for rewriting packets |
| 02:24.42 | syle2 | i;ve never seen anyone use reinvites behind nat to begin with |
| 02:25.13 | ManxPower | syle2, there really aren't many situations where you need NAT reinvites between devices between different nats. |
| 02:25.38 | ManxPower | An ITSP would have most of the calls going to or from the PSTN, so reinvites don't gain you anything there. |
| 02:25.46 | syle2 | gimme an example i;m curious |
| 02:25.56 | dijungal | can asterisk take the call in one codec and send it to another server in another codec..? |
| 02:26.11 | Netgeeks | dijungal: yes |
| 02:26.21 | ManxPower | syle2, A company with many SIP devices behind different NAT routers that need to call each other and the company does not have enough bandwidth to have all the audio go via Asterisk |
| 02:26.26 | Qwell | dijungal: its called transcoding, and it can take a bit of CPU to do so |
| 02:26.29 | dijungal | hmm.. maybe i should do that then... |
| 02:26.31 | Qwell | (which quickly adds up) |
| 02:26.39 | dijungal | oooh.. :| |
| 02:27.07 | ManxPower | dijungal, "show translations" or "show translation" I don't recall which |
| 02:27.34 | ManxPower | shows you the ms needed to transcode between different codecs on yuur system. |
| 02:27.37 | Netgeeks | depends on the amount of calls you plan to have running... 24 g729 to g711 is possible on a modest machine |
| 02:27.41 | Qwell | ManxPower: show trans<TAB> |
| 02:28.23 | dijungal | thanx qwell |
| 02:28.51 | dijungal | emm... i'm trying to figure out the table.. :s |
| 02:28.59 | dijungal | ok i got it now |
| 02:29.09 | dijungal | a little reading helps.. ;) |
| 02:32.23 | ManxPower | ~docs |
| 02:32.24 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 02:32.52 | dijungal | ok.. let me try the transcoding thing.. |
| 02:33.04 | dijungal | i hope it's easy to implement into the didalplan |
| 02:33.25 | Qwell | dijungal: its not something you implement in the dialplan |
| 02:33.37 | Qwell | sip.conf and iax.conf |
| 02:33.39 | ManxPower | "We hope our SpeedLinks(tm) system, unmatched in the industry...." It's a f*cking ADSL connection from the local telco with NAT |
| 02:35.05 | ManxPower | syle2, The only significant issue I have with my travel setup is I don't have a small, compact, 2-line phone w/2.5mm headset jack |
| 02:39.12 | Qwell | ManxPower: needs to be battery powered and wireless also |
| 02:40.25 | ManxPower | Qwell, gads why? |
| 02:40.33 | Qwell | dunno |
| 02:40.35 | Qwell | why not? |
| 02:40.53 | ManxPower | why? |
| 02:41.04 | ManxPower | I'm going to have power. 8-) |
| 02:41.12 | Qwell | not when you're...like... |
| 02:41.17 | Qwell | on the go? I don't know. :p |
| 02:41.28 | blitzrage | is it just me, or does the bug tracker seem slow? |
| 02:42.12 | brookshire | blitzrage: mailing list is making it slow |
| 02:42.29 | brookshire | we're trying to move it onto a new server |
| 02:42.37 | brookshire | soon :) |
| 02:42.47 | file[laptop] | blitzrage: I'm sorta in your territory |
| 02:43.09 | Qwell | file[laptop]: Did bkw poke you? |
| 02:43.23 | file[laptop] | I'm stranded in Toronto |
| 02:43.26 | Qwell | hrm |
| 02:43.33 | file[laptop] | see http://www.cnn.com/ for details |
| 02:43.42 | dijungal | any websites on transcoding..? |
| 02:43.48 | dijungal | can't seem to find it on the wiki |
| 02:43.54 | Qwell | file[laptop]: The Air France shit? |
| 02:43.57 | file[laptop] | yeah |
| 02:44.01 | file[laptop] | cancelled my flight to Chicago |
| 02:44.09 | Qwell | wtf |
| 02:44.14 | Qwell | cancelled, or delayed? |
| 02:44.16 | Juggie | plane crashed there |
| 02:44.19 | file[laptop] | all flights tonight were cancelled |
| 02:44.24 | Juggie | file, drive to ottawa |
| 02:44.25 | file[laptop] | I'm rebooked for tomorrow |
| 02:44.28 | Qwell | oh...I see |
| 02:44.36 | Juggie | no one died tho which was good |
| 02:44.37 | dijungal | emm... yeah... ok... transcoding..??? |
| 02:44.39 | file[laptop] | I'm staying at a friend's house |
| 02:44.39 | *** join/#asterisk Inv_arp (junya@adsl-3-237-158.mia.bellsouth.net) |
| 02:44.40 | Juggie | stupid pilot landing in T&L |
| 02:44.41 | Qwell | file[laptop]: Did you at least get to see the wreckage? :( |
| 02:44.45 | file[laptop] | well |
| 02:44.51 | Qwell | let me rephrase |
| 02:44.55 | Qwell | file[laptop]: Did you at least get to see the /flaming/ wreckage? :( |
| 02:44.55 | file[laptop] | i saw the fire trucks go |
| 02:44.56 | file[laptop] | that's it |
| 02:45.03 | Qwell | totally not worth it then |
| 02:45.03 | brookshire | that sucks :( |
| 02:45.05 | Juggie | i'm flying thursday hopefully its all cleared by then no moredelays and such |
| 02:45.22 | blitzrage | file[laptop]: oh yah... holy shit that sucks! |
| 02:45.25 | file[laptop] | blitzrage: yeah |
| 02:45.26 | blitzrage | file[laptop]: do you need a place to stay? |
| 02:45.34 | file[laptop] | nah I'm at a friend's house in Markham |
| 02:45.34 | blitzrage | file[laptop]: my place is super easy to get to |
| 02:45.42 | blitzrage | file[laptop]: wow, that was convenient ;) |
| 02:45.57 | file[laptop] | hehe |
| 02:46.02 | file[laptop] | how far away from Pearson are you? lol |
| 02:46.06 | blitzrage | file[laptop]: that sucks though - when do you fly out next? |
| 02:46.12 | Inv_arp | who has an voicepulse IP i can ping test before i sign up.... |
| 02:46.13 | file[laptop] | because if you swing by Pearson you can meet me |
| 02:46.14 | blitzrage | file[laptop]: ummm, not that far... like a 30 min cab ride |
| 02:46.30 | Juggie | blitzrage, your not going to clue con i take it? |
| 02:46.32 | blitzrage | file[laptop]: oh, yah, its not that easy to get to... will be easier when I move to Mississauga |
| 02:46.37 | blitzrage | Juggie: no sir |
| 02:46.39 | file[laptop] | yeah |
| 02:46.46 | file[laptop] | anyway I fly out tomorrow at 12 |
| 02:46.47 | Juggie | i thought about it but i cant make it either |
| 02:46.53 | Juggie | file, are you late for cluecon? |
| 02:46.57 | file[laptop] | yes, yes I am |
| 02:46.57 | Hmmhesays | damn b0rken planes |
| 02:47.01 | blitzrage | file[laptop]: not too shabby - too bad you'll be a bit late |
| 02:47.06 | file[laptop] | yeah |
| 02:47.08 | Juggie | why didnt you ask could u get a flight out of somewhere else |
| 02:47.09 | Qwell | file[laptop]: You'll have to come to astricon now. |
| 02:47.19 | file[laptop] | I'll fly into my hotel room, shower, go insane, and fly down to watch the presentations |
| 02:47.24 | file[laptop] | fly through the air |
| 02:47.28 | Hmmhesays | my plane broke too file |
| 02:47.29 | Juggie | like ottawa |
| 02:47.35 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 02:47.56 | blitzrage | could have taken the bus up to Ottawa or the Via train |
| 02:48.06 | blitzrage | Via would have probably been the easiest |
| 02:48.07 | Hmmhesays | so here I sit in fargo |
| 02:48.16 | Qwell | Hmmhesays: What happened to yours? |
| 02:48.32 | Hmmhesays | one of the doors malfunctioned |
| 02:48.42 | file[laptop] | blitzrage: meh waiting is fine with me |
| 02:48.51 | file[laptop] | blitzrage: and if you reallly do want to meet me... well, I'll be there ;) |
| 02:50.19 | Hmmhesays | mine was a blessing in disguise though |
| 02:51.05 | Hmmhesays | no I know when I get back my lady friend is going to tell me we shouldn't see each other again |
| 02:51.12 | Hmmhesays | now I |
| 02:52.51 | *** join/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg) |
| 02:52.55 | Hmmhesays | which is better than had it been a suprise |
| 02:59.53 | Inv_arp | who has an voicepulse IP i can ping test before i sign up.... |
| 03:03.04 | SkramX | Im ba-ack! |
| 03:03.48 | kingtux | voipbuster still down |
| 03:03.49 | kingtux | ?? |
| 03:04.19 | Qwell | kram: afternoon |
| 03:06.46 | kram | hi qwell |
| 03:07.34 | Qwell | kram: should find me at astricon...I'll buy you that beer I promised |
| 03:07.45 | Qwell | I was able to convince my boss today, heh |
| 03:08.30 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 03:09.17 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 03:09.31 | kingtux | Anyone know of a billing system for a calling card system?? |
| 03:09.41 | kingtux | I have areskicc installed and working |
| 03:09.41 | file[laptop] | astcc? |
| 03:09.55 | kingtux | looking to take it 1 step futher |
| 03:12.07 | kingtux | nothing |
| 03:12.07 | kingtux | ?? |
| 03:12.30 | hardwire | blah and 1/2 dookie |
| 03:12.32 | hardwire | time to go to the gym |
| 03:12.36 | file[laptop] | kingtux: astcc is a cc system... but all of this is free, can't expect the world for nothing |
| 03:13.36 | kingtux | don't understand what u mean |
| 03:13.36 | kingtux | ?? |
| 03:14.00 | kingtux | I have areskicc install same thing as astcc, just not mysql based |
| 03:15.00 | dalabera | guys is there a chat room for Developments questions, I'm working on pbx.c and pbx_spool.c from the sources |
| 03:15.04 | file[laptop] | you're not going to find something that works exactly for what you want |
| 03:16.44 | Netgeeks | +++++++++++++ |
| 03:17.18 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 03:19.18 | *** join/#asterisk MustDie (~Alik@ool-18b91f29.dyn.optonline.net) |
| 03:23.19 | dijungal | yoyo yo |
| 03:23.28 | dijungal | who changed the topic to dalug..? |
| 03:24.10 | syle2 | dalavera: join asterisk-devel mailing lists and perhaps #asterisk-bugs |
| 03:30.28 | Math` | its funny how timeouts and hangups are being routed thru dundi-test |
| 03:30.28 | Math` | lo |
| 03:30.30 | Math` | lol* |
| 03:31.32 | Barmal | how many simuolteneous calls can * handle at the same time? |
| 03:31.35 | brookshire | dalabera: #asterisk-dev |
| 03:32.00 | jskcr | Barmal: its depends on alot of factors |
| 03:32.01 | brookshire | Barmal: a lot |
| 03:32.02 | brookshire | :) |
| 03:32.08 | Barmal | 1000? |
| 03:32.14 | brookshire | no |
| 03:32.18 | brookshire | but |
| 03:32.18 | Barmal | 100? |
| 03:32.41 | brookshire | it can handle 1000 if it's routing the media not through asterisk |
| 03:32.50 | Math` | dalabera: join asterisk-dev we'll talk there |
| 03:32.52 | Inv_arp | who has an voicepulse IP i can ping test before i sign up.... |
| 03:33.27 | Barmal | brrokshire, what do you mean routing the media not through asterisk? |
| 03:35.14 | brookshire | well.. in sip |
| 03:35.20 | brookshire | you have a control link |
| 03:35.25 | brookshire | and a media link |
| 03:35.32 | brookshire | the control link controls a voip device |
| 03:35.42 | brookshire | and the media link the actual voice |
| 03:35.44 | twisted | sip happens |
| 03:35.48 | brookshire | sip does happen |
| 03:35.52 | brookshire | :( |
| 03:37.19 | file[laptop] | ahhhhhhhh |
| 03:37.36 | *** join/#asterisk JunK-Y (~foobar@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 03:38.19 | file[laptop] | JunK-Y! |
| 03:38.44 | Barmal | brookshire, so would it handle 400 users routing media through *? |
| 03:38.54 | *** join/#asterisk santiago (~santiago@63.245.87.180) |
| 03:39.11 | JunK-Y | file |
| 03:39.22 | JunK-Y | fucker, i told you to transfer in montreal :) |
| 03:39.26 | file[laptop] | :P |
| 03:39.29 | file[laptop] | too bad |
| 03:39.32 | JunK-Y | where are ya now^ |
| 03:39.36 | file[laptop] | Markham |
| 03:39.38 | twisted | JunK-Y, nobody listens to you :P |
| 03:39.38 | file[laptop] | at a friend's house |
| 03:39.47 | file[laptop] | I'll be in tomorrow at lunch time |
| 03:39.54 | JunK-Y | they dont transfer everyone by trudeau? |
| 03:40.00 | JunK-Y | twisted |
| 03:40.01 | JunK-Y | hehehe |
| 03:40.23 | brookshire | baramal: it would probably take a cluster of asterisk boxes to do that |
| 03:40.37 | brookshire | i mean |
| 03:40.50 | brookshire | 400 lines |
| 03:40.51 | Barmal | brookshire, so whats the bottle neck for one box? |
| 03:40.51 | brookshire | not users |
| 03:41.01 | file[laptop] | JunK-Y: how goes it? |
| 03:41.16 | brookshire | Barmal: just depends |
| 03:41.30 | brookshire | are you doing g729, gsm, g711? |
| 03:41.35 | brookshire | is it just the media |
| 03:41.40 | Barmal | brrokshire, gsm |
| 03:41.44 | JunK-Y | its great. |
| 03:41.46 | brookshire | i have no idea |
| 03:41.59 | Barmal | brookshire, :) ok g711 |
| 03:42.12 | file[laptop] | JunK-Y: good good |
| 03:42.24 | brookshire | i know of people doing 12 t1s of ptsn on one box with g729 |
| 03:42.41 | brookshire | 12 * 23 |
| 03:42.48 | file[laptop] | not math!!! |
| 03:42.56 | brookshire | ~jbot 12*23 |
| 03:42.56 | jbot | 276 |
| 03:42.58 | brookshire | haha |
| 03:43.12 | brookshire | wait |
| 03:43.15 | brookshire | maybe it's not g729 |
| 03:43.16 | brookshire | haha |
| 03:43.18 | brookshire | g711 |
| 03:43.29 | brookshire | g729 eats a lot of cpu |
| 03:44.05 | Barmal | so basically * doenst have limitations as a software |
| 03:44.12 | brookshire | 150 lines with g729 |
| 03:44.45 | brookshire | Barmal: any large scale asterisk installation will run into problems, but i don't think there is anything out there that will do all that asterisk does |
| 03:45.04 | Barmal | I am trying to image big companys how many boxes they have lets say broadvoice? |
| 03:46.01 | jskcr | 1 or 2 |
| 03:46.46 | jskcr | you can handle 100,000+ sip connections on ser and then use asterisk for voicemail and or pstn pbx |
| 03:50.17 | SkramX | damn! |
| 03:50.45 | jskcr | 250,000 on a dual xeon |
| 03:50.54 | SkramX | nice |
| 03:50.56 | SkramX | i want one |
| 03:51.00 | brookshire | heh.. ser is kinda stupid though.. it still needs something else |
| 03:51.15 | brookshire | meaning.. not much intelligence in ser |
| 03:51.17 | jskcr | exactly asterisk is not a sip proxy ser is |
| 03:51.20 | Barmal | so billing software will stand on asterisk? |
| 03:51.52 | jskcr | yea its a bit of a learning curve, you have to write all your intelligence into the ser conf. its kind off like sendmail |
| 03:52.46 | Qwell | Where does Record() save to? |
| 03:53.08 | Qwell | ahh, nevermind |
| 03:54.08 | SkramX | hehe |
| 03:54.12 | SkramX | ~voip-info.org |
| 03:54.12 | jbot | from memory, voip-info.org is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 03:54.23 | SkramX | Good job, jbot! |
| 03:54.46 | jskcr | ~ser |
| 03:54.46 | jbot | rumour has it, ser is Sip Express Router - see http://www.iptel.org/ser/ |
| 03:56.10 | brookshire | Barmal: asterisk is very configurable, even integrates in with databases |
| 03:56.55 | jskcr | Barmal theres cdrtool |
| 03:57.16 | jskcr | The odbc cdr reporting from asterisk works quite well |
| 03:57.24 | Barmal | I know I was trying to install this areski today |
| 03:58.40 | Barmal | but I was thinking if you planning to have about 1000 users and all of them will start calling at the same time asterisk will not handle.... I need to read on that SER... |
| 03:59.22 | brookshire | asterisk can handle that many... you'll just have to cluster |
| 03:59.25 | *** join/#asterisk PBXtech (~nik@h460659c4.area1.spcsdns.net) |
| 03:59.35 | drumkilla | brookshire: !!!!!!!!! |
| 03:59.39 | brookshire | russell! |
| 03:59.42 | *** join/#asterisk derobert (~anthony@Maxwell.derobert.net) |
| 03:59.46 | drumkilla | I just watched a *terrible* movie |
| 03:59.52 | brookshire | uh oh |
| 03:59.53 | drumkilla | it was called "Mosquito Man" |
| 03:59.56 | brookshire | HAHA |
| 04:00.02 | brookshire | you should have known from the title |
| 04:00.08 | drumkilla | yeah |
| 04:00.16 | drumkilla | but it was fun making fun of |
| 04:00.41 | brookshire | was it better than hackers? |
| 04:01.20 | brookshire | did they hack the gibson? |
| 04:01.24 | drumkilla | no way |
| 04:01.38 | mog_home | nothing is better than hackers |
| 04:01.39 | mog_home | ever |
| 04:01.41 | drumkilla | but a whole lot of people got killed by an 8-foot Mosquito |
| 04:01.48 | brookshire | haha |
| 04:01.48 | file[laptop] | drumkilla!!! |
| 04:02.03 | brookshire | sucksored! |
| 04:04.06 | drumkilla | mhm! |
| 04:04.47 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 04:06.47 | derobert | Hello. I'm setting up an asterisk server (using the asterisk 1.0.7 from debian sarge) and have run into a problem with meetme. Every once and a while, everyone in the conference goes silent --- the bandwidth usage doesn't go down, so I think asterisk is transmitting silence. The changelog in 1.0.9 does not show anything relevant. Suggestions? |
| 04:07.52 | twisted | drumkilla, |
| 04:07.55 | jskcr | cvs head |
| 04:07.58 | twisted | you left before I could tell you my story |
| 04:07.59 | twisted | bastardo |
| 04:08.11 | *** join/#asterisk nwhit (~chatzilla@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 04:08.11 | derobert | (Also, I noticed that the latency is rather bad. Not sure if that's asterisk's fault, or if its linphone's fault). Load on the asterisk server (dual PIII) is nil. Have tried multiple codecs, including gsm, ulaw, and speex. |
| 04:08.34 | derobert | jskcr: is that directed at me? |
| 04:08.42 | jskcr | yup |
| 04:08.59 | twisted | derobert, what's your timing source? |
| 04:09.09 | `Sauron | hum, did the pthreads implementation change with a 2.6 kernel? |
| 04:09.09 | derobert | twisted: ztdummy, linux 2.6 |
| 04:09.27 | `Sauron | Cuz the * threads no longer show as seperate pseudo-processes |
| 04:09.47 | derobert | `Sauron: yeah, linux 2.6 has a new thread model. Try ps -T |
| 04:09.53 | `Sauron | hum, ah |
| 04:10.05 | twisted | derobert, heh. do you have SMP turned on? |
| 04:10.07 | jskcr | twisted anyone report issues with asterisk and enabling PREEMPT in the kernel? |
| 04:10.17 | *** join/#asterisk Cresl1n (~Cresl1n@user-24-236-124-147.knology.net) |
| 04:10.19 | derobert | twisted: yes, SMP is on |
| 04:10.21 | `Sauron | hum |
| 04:10.24 | Cresl1n | twisted! |
| 04:10.25 | derobert | twisted: its an SMP box |
| 04:10.26 | Cresl1n | hey |
| 04:10.30 | drumkilla | Cresl1n: !!!!! |
| 04:10.31 | twisted | jskcr, considering asterisk doesn't deal with the kernel, probably not ;P |
| 04:10.35 | Cresl1n | drumkilla! |
| 04:10.36 | twisted | Cresl1n, !! |
| 04:10.39 | twisted | drumkilla, you bastardo |
| 04:10.42 | brookshire | heh |
| 04:10.50 | `Sauron | Ah, there. |
| 04:10.50 | drumkilla | wtf |
| 04:10.52 | Cresl1n | brookshire! |
| 04:10.57 | brookshire | matt! |
| 04:10.58 | twisted | derobert, run it in single proc mode. i don't know how well ztdummy deals with smp on a 2.6 box |
| 04:11.06 | `Sauron | Ugh |
| 04:11.06 | dalabera | guys is there a chat room for Developments questions, I'm working on pbx.c and pbx_spool.c from the sources |
| 04:11.18 | dalabera | trying to implement some changes |
| 04:11.20 | twisted | dalabera, #asterisk-dev |
| 04:11.24 | dalabera | cool |
| 04:11.24 | jskcr | ya ive had problems with ztdummy with smp kernel |
| 04:11.25 | `Sauron | derobert: And how do you get strace to attach to a thread? :) |
| 04:11.39 | jskcr | Got weird poping noices and silence |
| 04:11.46 | derobert | `Sauron: not sure on that one... strace -f will attach to all of them, I think |
| 04:12.04 | `Sauron | Hum, apparently not. |
| 04:12.08 | twisted | derobert, try non-SMP, or try zaprtc |
| 04:12.35 | derobert | twisted: well, non-SMP would be a real waste of hardware.... I'll give zaprtc a try. |
| 04:12.55 | jskcr | strace -f |
| 04:13.18 | twisted | ztdummy uses a pseudo timing source from the kernel. zaprtc talks to the RTC in the machine ;) |
| 04:13.42 | jskcr | derobert: got a pci slot open? |
| 04:13.48 | derobert | jskcr: yeah |
| 04:13.52 | `Sauron | Ah |
| 04:13.54 | twisted | or yeah, slap in a x100p/tdm01b |
| 04:14.02 | twisted | and use a real hardware timer |
| 04:14.45 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 04:14.55 | `Sauron | ztdummy should work fine on 2.6 |
| 04:15.12 | `Sauron | Hehn. |
| 04:15.22 | twisted | `Sauron, so should my usb powered pocket pussy with variable speed conrol |
| 04:15.22 | `Sauron | found the problem, just dunno why it's happening |
| 04:15.24 | twisted | *control |
| 04:15.26 | twisted | but it doesn't |
| 04:15.37 | jskcr | not on smp systems or ones with hyperthreading enabled. |
| 04:15.50 | derobert | ok, well, quikest to try is non-smp... |
| 04:16.14 | jskcr | yup |
| 04:16.55 | jskcr | I tried ztdummy on a smp kernel earlier this week and it even messed up the sip audio it was like talking underwater. |
| 04:17.22 | brookshire | showing while talking on the phone is not a good idea |
| 04:17.28 | brookshire | showering |
| 04:17.31 | twisted | LOL |
| 04:17.33 | mog_home | get a real timing device |
| 04:17.50 | twisted | brookshire, that +90V kinda sucks in water, eh/ |
| 04:18.03 | brookshire | soothing really |
| 04:18.07 | twisted | haha |
| 04:18.11 | twisted | cleansed your bowels |
| 04:18.20 | jskcr | lol |
| 04:18.23 | brookshire | gental pulse of relief |
| 04:18.27 | *** join/#asterisk yipdw (~trythil@64.58.0.54) |
| 04:18.35 | derobert | twisted: use a cordless phone, seal it in a bag, maybe? |
| 04:18.35 | twisted | haha |
| 04:18.43 | jskcr | ~x100p |
| 04:18.43 | jbot | [x100p] an obsolete card. you don't want to bother trying to make it (or any of the "digium compatible" clones work. Get a TDM01P, you will save your sanity. |
| 04:18.46 | twisted | i'm not the one who did it, derobert |
| 04:19.11 | CoaxD | haha |
| 04:19.25 | CoaxD | they told us.. "The X100P is a GREAT card!" |
| 04:19.27 | derobert | twisted: no, but you realized the 90V problem. Hence, my offering of a solution :-) |
| 04:19.32 | CoaxD | "ONLY buy the expensive Digium boards!" |
| 04:19.41 | CoaxD | ...within a couple months, its all.. "That old thing? Chuck it." |
| 04:19.53 | *** join/#asterisk nwhit (~chatzilla@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 04:19.54 | CoaxD | some of us JUST BOUGHT our x100p's within the last year |
| 04:20.03 | jskcr | put a condom on the phone. |
| 04:20.43 | derobert | hmmmm..... wonder if the box is going to boot :-( |
| 04:21.04 | jskcr | they should come out with just a cheap clocking device for people who only use sip |
| 04:21.29 | derobert | what is the clocking device actually used for? |
| 04:21.41 | Qwell | derobert: meetme and trunking |
| 04:21.43 | derobert | (and why does it need special hardware) |
| 04:22.04 | Nugget | I'd love a crossplatform timing solution. |
| 04:22.25 | Qwell | a usb timing device? heh |
| 04:22.39 | jskcr | lol |
| 04:22.42 | drumkilla | that's like ... an oxymoron |
| 04:22.45 | derobert | Ugh. nosmp seems to have left the box not booting. And they're no monitor/keyboard on it.... :-( |
| 04:22.47 | jskcr | thats been tried |
| 04:22.47 | jskcr | ztdummy |
| 04:23.14 | brookshire | haha.. |
| 04:23.16 | derobert | what kind of timing is required for a meetme conference? |
| 04:23.28 | twisted | teh good kind |
| 04:23.32 | Nugget | heh |
| 04:23.36 | `Sauron | hum |
| 04:23.39 | Qwell | derobert: It needs to know when (accurately) to send packets |
| 04:23.41 | Qwell | or something |
| 04:23.59 | CoaxD | derobert; A device capable of interrupting 1000 times per second, exactly, and very precisely |
| 04:24.11 | *** join/#asterisk cgcorea (~cgcorea@63.245.14.194) |
| 04:24.19 | Qwell | CoaxD: very precisely...or thereabouts ;] |
| 04:24.41 | CoaxD | qwell: ...as is the case with the usb timing devices, or ztdummy |
| 04:24.43 | *** join/#asterisk Lathos42 (~Lathos42@h46091809.area4.spcsdns.net) |
| 04:25.03 | blitzrage | god I love Visio and PowerPoin! |
| 04:25.04 | derobert | CoaxD: ummm, but doesn't /dev/rtc do that? |
| 04:25.06 | blitzrage | +t |
| 04:25.18 | CoaxD | derobert: Pretty decently, yea |
| 04:25.35 | jskcr | Qwell Im working on a multiplatform timing device but its tough to get the nuclear materials to keep perfect time. |
| 04:25.38 | derobert | CoaxD: at least, as long as you turn up max-user-frequence to 1024, or whatever. |
| 04:26.10 | CoaxD | derobert: the wierdness surrounding zaprtc was that until semi-recently,they couldn't figure out how to make it SMP safe |
| 04:26.13 | Netgeeks | <PROTECTED> |
| 04:26.20 | Netgeeks | [21:23] CoaxD: derobert; A device capable of interrupting 1000 times per second, exactly, and very precisely |
| 04:26.22 | nwhit | lathos42 ... that sux |
| 04:26.38 | CoaxD | netgeek: Bwahahahahaha |
| 04:26.42 | CoaxD | netgeek: Yeah, mine too. *lol* |
| 04:27.10 | derobert | According to http://www.voip-info.org/tiki-index.php?page=Asterisk+zaprtc zaprtc on 2.6 is rather broken |
| 04:27.28 | CoaxD | derobert: dunno, never tried it. just used an x100p for timing |
| 04:27.30 | *** part/#asterisk Lathos42 (~Lathos42@h46091809.area4.spcsdns.net) |
| 04:27.39 | jskcr | Im a bit weary of using zaprtc in a production smp server |
| 04:27.51 | blitzrage | just hashing away some ideas here, if someone where to write a "cookbook" for Asterisk, what kinds of "recipes" would you like to see? |
| 04:28.01 | CoaxD | jskcr: It should work. Its just that the code to make it happen is a locking nightmare |
| 04:28.16 | Netgeeks | you don't need zaprtc in 2.6 kernels, use ztdummy |
| 04:28.46 | CoaxD | netgeeks: they figure out how to make the ztdummy module modify the high res timer on the system to beat at precisely 1000 times per second? |
| 04:29.02 | jskcr | Netgeeks: smp |
| 04:29.46 | drumkilla | blitzrage: something with ... butter! |
| 04:29.52 | twisted | boy butter! |
| 04:29.57 | twisted | http://www.boybutter.com |
| 04:29.58 | Netgeeks | I'm not sure, but I have three dual proc systems (one a dual AMD 2800MP system, a dual Xeon 3.06 system, and a Dual Opteron) all three using 2.6 kernel and I've tested both trunking and meetme with success |
| 04:30.11 | brookshire | thanks for that twisted |
| 04:30.11 | CoaxD | netgeeks: How many people in your meetme? |
| 04:30.26 | Netgeeks | on the opteron we had 18 tops |
| 04:30.30 | twisted | brookshire, you're welcome! |
| 04:30.32 | CoaxD | netgeeks: Thats a good test |
| 04:30.42 | derobert | Netgeeks: oh? I had only two or three people, and it was having problems with the audio dropping out... |
| 04:30.44 | Cresl1n | who wants to go drinking? |
| 04:30.49 | twisted | Cresl1n!!! |
| 04:30.50 | derobert | Netgeeks: it'd just transmit silence for several seconds |
| 04:30.52 | Cresl1n | what |
| 04:30.55 | twisted | Cresl1n, mee!!!! |
| 04:30.56 | blitzrage | Cresl1n: me! |
| 04:30.57 | Cresl1n | redbull...? |
| 04:30.58 | Cresl1n | :-) |
| 04:31.00 | blitzrage | bah! |
| 04:31.04 | twisted | Cresl1n!! |
| 04:31.06 | twisted | Cresl1n, mee!!!! |
| 04:31.08 | Cresl1n | what!? |
| 04:31.08 | Cresl1n | :-) |
| 04:31.09 | Cresl1n | ok |
| 04:31.12 | Netgeeks | strange, drobert |
| 04:31.20 | brookshire | you don't drink |
| 04:31.23 | derobert | Netgeeks: what version are you running? I've got 1.0.7 (debian sarge) |
| 04:31.26 | brookshire | don't tease me like that |
| 04:31.34 | Cresl1n | heh |
| 04:31.37 | blitzrage | brookshire: no kidding eh? :) |
| 04:31.52 | Netgeeks | i'M RUNNING cvs head FROM 09/29/2005 |
| 04:31.54 | brookshire | blitzrage: huh? |
| 04:31.55 | *** join/#asterisk techie (~gus@70.86.57.50) |
| 04:32.06 | derobert | Netgeeks: ummm, what? |
| 04:32.19 | CoaxD | netgeeks: 1.21 JIGAWATTS! |
| 04:32.28 | Netgeeks | I'm runnin gHEAD from sep 29, 2004... sorry |
| 04:32.30 | Netgeeks | not 2005 |
| 04:32.34 | brookshire | CoaxD: going back in time? |
| 04:32.36 | jskcr | wow can I have a flux capacitor |
| 04:32.37 | yipdw | heh, CVS from the future |
| 04:32.48 | derobert | Netgeeks: ok, 'cause running a future CVS head is impressive. Very impressive. |
| 04:32.53 | CoaxD | brookshire: Nahhh. but netgeeks was heading into the future.. |
| 04:33.12 | blitzrage | brookshire: Cresl1n teasing about drinking |
| 04:33.17 | brookshire | oh yeah |
| 04:33.20 | jskcr | cvs -D "ten days ago" here :) |
| 04:33.27 | CoaxD | just imagine.. the sacred cvs server that allowed you to view cvs trees just 1 year into the future.. |
| 04:33.37 | `Sauron | hum, now to find out whatt this thread is trying to do |
| 04:33.38 | yipdw | I'd love that |
| 04:33.43 | brookshire | what's sad.. is if crestl1n ever did decide one day to drink, i would be the one who corrupts him, lol |
| 04:33.46 | yipdw | it'd help out Inkboard development a lot |
| 04:33.50 | twisted | try cvs -D "two years ago" |
| 04:33.53 | twisted | there were very few bugs |
| 04:33.55 | derobert | OK, well I guess I'll try CVS head. Or should I give 1.0.9 a shot first? Anything to watch out for? |
| 04:34.04 | Cresl1n | always give head a shot |
| 04:34.10 | Cresl1n | that's my philosophy |
| 04:34.16 | brookshire | shot for head? |
| 04:34.21 | brookshire | hmmm |
| 04:34.22 | nwhit | it would make development alot faster |
| 04:34.23 | CoaxD | crestlin: Giving head isn't my cup of tea.. |
| 04:34.23 | Netgeeks | and give that date I listed a try, the past one not the future |
| 04:34.25 | drumkilla | head is always worth a shot. |
| 04:34.39 | CoaxD | crestlin: but if it works for you.. |
| 04:34.39 | brookshire | then winks |
| 04:34.41 | Cresl1n | ooh.... |
| 04:34.44 | jskcr | twisted: I dont think the extended odbc storage patch 4403 would work with that. |
| 04:34.47 | derobert | Netgeeks: doesn't a year-old snapshot have some potential security problems? |
| 04:34.49 | Netgeeks | I had major problems with HEAD after about 10-1-2004 |
| 04:34.49 | drumkilla | woah now |
| 04:34.53 | brookshire | <WINK> <WINK> |
| 04:34.58 | twisted | jskcr, you don't need that |
| 04:35.32 | `Sauron | Ah. |
| 04:35.34 | `Sauron | hehn. |
| 04:35.47 | jskcr | twisted: why. |
| 04:35.59 | Netgeeks | I was getting some bad SIP audio breaking up that I fixed by falling back from early 2005/late 2004 head to 9/29/04 |
| 04:36.11 | jskcr | has it been merged into the cvs? |
| 04:37.24 | *** join/#asterisk fugitivo (~ajf@201.255.101.206) |
| 04:37.28 | fugitivo | hello |
| 04:39.11 | derobert | hmmmmm.... http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg11995.html |
| 04:39.26 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 04:40.02 | Cresl1n | hrm.... |
| 04:41.11 | derobert | Wow, I never tried the q option.... |
| 04:41.17 | derobert | Sounds like I should give that whirl, too. |
| 04:42.14 | *** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au) |
| 04:43.00 | nwhit | th wireless sux in this hotel |
| 04:43.58 | twisted | cry me a fucking river |
| 04:44.03 | *** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net) |
| 04:44.26 | irv999 | has anyone had any issues with digum cards being way to sensitive with T1's and going into red alarm even though the T1 is up |
| 04:44.26 | irv999 | ? |
| 04:44.42 | jskcr | irv999 sounds like a smart jack problem |
| 04:44.52 | irv999 | jsk nope |
| 04:44.53 | nwhit | blahh...blahh...tear teat |
| 04:45.23 | irv999 | jsk well I dont think so.. phone company says t1 is fine.. and when we change the t1 dbounce to 20 from 0 it goes from 20 red alarms to 0 |
| 04:45.49 | asteriskmonkey | i have a t110p what pri card do you have? |
| 04:46.12 | irv999 | aster: is that the 4 port card? 3.3 V? that is what I have.. I thbink it is the 110 |
| 04:46.27 | asteriskmonkey | 110p is a single t1 card |
| 04:46.44 | asteriskmonkey | no matter check the actual jump or switch on the board becuase you can set it to e1/t1 |
| 04:46.44 | irv999 | aster: I have a 411 then |
| 04:47.06 | asteriskmonkey | it could be set up for the wrong type of switch bank (its a physical toggle switch) |
| 04:47.19 | irv999 | aster: the red moveable switch? |
| 04:47.31 | asteriskmonkey | should be labeled t1/e1 |
| 04:47.38 | irv999 | aster: I will check |
| 04:47.41 | asteriskmonkey | look at the pdf for that unit :D |
| 04:48.23 | irv999 | aster: will do.. thanks for the info.. |
| 04:48.29 | irv999 | night all... |
| 04:49.23 | asteriskmonkey | night |
| 04:53.10 | Barmal | stupid q: in sip.conf what does 'user=phone' stand for? |
| 04:53.53 | gambolputty | user= is for incoming/outgoing calls |
| 04:54.01 | gambolputty | what is allowed |
| 04:54.29 | Barmal | so user=<context for incomming calls> should be? |
| 04:54.44 | gambolputty | type= |
| 04:54.45 | gambolputty | whoops |
| 04:56.26 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:58.13 | SkramX | lalalalala |
| 04:58.21 | `Sauron | derobert: thanks for the help with the strace stuff earlier |
| 04:59.21 | `Sauron | apparently chan_oss isn't happy if you symlink /dev/dsp to a named pipe |
| 05:00.13 | *** join/#asterisk X-Rob (~rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 05:00.43 | hardwire | blah |
| 05:00.44 | hardwire | and |
| 05:00.45 | hardwire | 1/2 |
| 05:00.46 | hardwire | dookies |
| 05:01.09 | X-Rob | 1/2 dookies? |
| 05:01.30 | X-Rob | stupidly cheap. |
| 05:02.00 | hardwire | eh |
| 05:02.02 | hardwire | err |
| 05:02.02 | hardwire | heh |
| 05:02.31 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 05:02.50 | *** part/#asterisk Cresl1n (~Cresl1n@user-24-236-124-147.knology.net) |
| 05:03.31 | PyroSteve | hey guys |
| 05:03.31 | X-Rob | Quality of Service Support |
| 05:03.31 | X-Rob | The switch supports Layer 2 802.1p Priority Queue control to prioritize network packets. Classification of users data priorities can be based on a data packet Priority Queue. |
| 05:03.46 | PyroSteve | Ive gotten some real experince with a phone system other than |
| 05:03.48 | PyroSteve | asterisk |
| 05:04.03 | PyroSteve | Im troubleshooting a Samsung DCS50si |
| 05:04.39 | hardwire | how do you GotoIf DB Key Exists |
| 05:04.48 | twisted | DBGET |
| 05:04.54 | twisted | if it fails, it goes to n+101 |
| 05:05.16 | pauldy | wow doesn't just fall through? |
| 05:05.30 | hardwire | twisted: that works |
| 05:05.40 | hardwire | I guess |
| 05:05.42 | PyroSteve | where can I find an Application list for * 1.2 ? |
| 05:06.15 | hardwire | tell me if I am an idiot, I want to store the OUTBOUNDCID/1111/"TDX Blah Blahblah <9077277272>" in the database |
| 05:06.23 | hardwire | via a script for the CLI |
| 05:06.24 | hardwire | or an AGI |
| 05:06.24 | PyroSteve | can someone msg me the output from `show application` |
| 05:06.35 | PyroSteve | on * 1.2 |
| 05:06.55 | hardwire | and then use GotoIF when I am dialing out on a certain trunk.. to match the ext (1111) to the full CallerID |
| 05:07.43 | hardwire | oh well.. brb. |
| 05:09.52 | PyroSteve | whos running asterisk 1.2 ? |
| 05:10.58 | SkramX | is it out? |
| 05:10.59 | SkramX | heh |
| 05:11.19 | PyroSteve | well maybe not |
| 05:11.24 | PyroSteve | sorry |
| 05:12.09 | PyroSteve | well .. isn't the application list the same in the latest cvs head |
| 05:12.15 | PyroSteve | compared to 1.2 ? |
| 05:12.23 | *** join/#asterisk file[laptop] (~file[lapt@69.158.162.238) |
| 05:13.05 | *** join/#asterisk jiro5281 (~anton281@203.131.137.76) |
| 05:14.23 | X-Rob | PyroSteve, CVS HEAD will become 1.2 |
| 05:14.29 | X-Rob | 1.2 does not exist at the moment |
| 05:14.35 | PyroSteve | ok .. thanks |
| 05:14.49 | X-Rob | there's at least 3 or 4 applications in mantis at th emoment, so the application list is hazy. |
| 05:15.10 | PyroSteve | mantis ? |
| 05:15.30 | PyroSteve | manifest ? |
| 05:23.02 | *** join/#asterisk derrick_ (~derrick@blinky-lights.org) |
| 05:23.10 | derrick_ | w/dinw split on |
| 05:24.25 | derobert | `Sauron: I'd expect not, considering /dev/dsp has a lot of ioctl's that a pipe won't. |
| 05:24.29 | *** join/#asterisk zoo (nobody@ip-54-16.travedsl.de) |
| 05:26.13 | *** join/#asterisk Beccara (~Tristram@222-152-13-41.jetstream.xtra.co.nz) |
| 05:27.41 | derrick_ | file, i figured out how to get laid: http://www.flickr.com/photos/jmckible/sets/684076/ |
| 05:28.02 | file[laptop] | oh goody! |
| 05:28.27 | derrick_ | it's not just the gundam suit. the rollerblades are needed |
| 05:28.55 | derrick_ | color me surprised |
| 05:30.10 | SkramX | Ok, last time.. what are some CallingCard/Billing systems that you have used / heard of are good? |
| 05:31.00 | derrick_ | the ones you spend three days making? |
| 05:31.26 | SkramX | heh |
| 05:31.32 | SkramX | I dont have time to make my own |
| 05:31.32 | derrick_ | you can spend 75k on it, or 3k |
| 05:31.40 | SkramX | Im talking about free ones. |
| 05:32.17 | derrick_ | in my opinion, your best turnaround on that is hire someone to do it. you pay for them and the product in one |
| 05:32.34 | derrick_ | to customize that is |
| 05:33.06 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 05:33.06 | *** mode/#asterisk [+o twisted] by ChanServ |
| 05:33.20 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 05:35.33 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:38.53 | SkramX | ?? |
| 05:39.03 | SkramX | who can do it? |
| 05:40.47 | *** join/#asterisk atif_ (~atif_@202.163.66.8) |
| 05:40.55 | Beccara | santa |
| 05:41.10 | twisted | i can do it |
| 05:41.17 | twisted | i've done it many many times |
| 05:41.24 | twisted | many satisfied partis |
| 05:41.26 | twisted | parties |
| 05:41.39 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 05:41.58 | twisted | oh wait, what are we talking about again? |
| 05:43.53 | *** join/#asterisk cfrank (~cfrank@wsip-24-234-137-140.lv.lv.cox.net) |
| 05:44.51 | *** join/#asterisk blackgecko (~blackgeck@dsl-200-78-42-225.prod-infinitum.com.mx) |
| 05:45.38 | SkramX | Calling Card programs |
| 05:45.43 | SkramX | twisted, can ya? |
| 05:45.45 | *** join/#asterisk Malthus (~admin@port0088-aaw-adsl.cwjamaica.com) |
| 05:45.46 | SkramX | how much? |
| 05:45.50 | SkramX | its for a non-profit |
| 05:45.57 | SkramX | and its just to keep track of patrons |
| 05:46.00 | SkramX | no money will be made |
| 05:46.03 | twisted | SkramX, oh, i thought you were talking about something else... but my company can ;) |
| 05:46.10 | SkramX | well |
| 05:46.13 | SkramX | can they donate it? |
| 05:46.28 | PyroSteve | wow |
| 05:46.47 | PyroSteve | I just crashed an * box by using chanspy |
| 05:47.02 | PyroSteve | i couldn't even start another ssh session |
| 05:47.11 | PyroSteve | but a few minutes of panicing |
| 05:47.29 | PyroSteve | the asterisk process died |
| 05:47.38 | PyroSteve | and i was able to log back in and restart it |
| 05:47.58 | infinity1 | whats the new syntax for DBput in HEAD? this is invalid! s,1,DB(FW/${CALLERIDNUM}=${EXTEN:4}) |
| 05:48.32 | SkramX | twisted? |
| 05:48.39 | SkramX | whats the url? |
| 05:49.48 | SkramX | twisted? Steve?! |
| 05:51.02 | SkramX | Sorry, I mean Josh |
| 05:51.26 | harryvv | exten:4? |
| 05:52.18 | *** join/#asterisk KaBewM (~kabewm@24-180-28-208.dhcp.psdn.ca.charter.com) |
| 05:53.27 | RaYmAn-Bx | infinity1: read UPGRADE.txt. |
| 05:54.46 | SkramX | eh?! |
| 05:55.36 | infinity1 | RaYmAn-Bx: thanks. got it. |
| 05:57.51 | *** join/#asterisk Maarken (~panties@moist235.drizzle.com) |
| 06:00.51 | infinity1 | interesting. the application While isn't documented anywhre. |
| 06:01.08 | derrick_ | top secret |
| 06:01.38 | SkramX | indigent.sytes.net ? |
| 06:02.20 | infinity1 | derrick_: apparently. lets see if i can make it work. |
| 06:02.33 | SkramX | what yalll talkng about? |
| 06:04.49 | SkramX | twisted: id like to do business with you, contact me |
| 06:04.56 | SkramX | via pm or mark@mark-s.net |
| 06:05.44 | derrick_ | i recommend him for any work. very very reliable |
| 06:06.44 | harryvv | hello |
| 06:07.04 | SkramX | haha.. |
| 06:07.16 | SkramX | not right now |
| 06:07.23 | SkramX | i need a free EASY TO USE CDR or CallingCard System |
| 06:07.32 | harryvv | Just watched footage of the air france a380 crash on the news. |
| 06:07.42 | SkramX | yea |
| 06:07.44 | harryvv | everyone got out of it alive. |
| 06:07.52 | Qwell | harryvv: except file. he's pissed. |
| 06:08.04 | Qwell | made him get stuck in Toronto airport...heh |
| 06:08.38 | harryvv | If it was a microburst that pushed if off the runway, then it would not suprise me. I have experainced microburst while in my aircaft on the ground. It was suffenand did alot of damage to our aircraft. |
| 06:09.08 | harryvv | file should be gratefull for what he has. |
| 06:09.36 | Qwell | SkramX: there are a bunch of cdr apps |
| 06:09.41 | Qwell | builtin even |
| 06:10.25 | SkramX | yea |
| 06:10.34 | SkramX | but graphical, online is what my user wants |
| 06:10.36 | SkramX | they are a newb |
| 06:10.39 | SkramX | hahaha |
| 06:10.41 | infinity1 | heh. i think While() / EndWHile() is not wokring. |
| 06:10.44 | Qwell | write a simple php app |
| 06:10.51 | harryvv | I have a feeling when the airraft went into the gully, the ground ripped open the underside of the wing or ripped off the engine and fuel dumping out. I have refuled airliners before and thats a likly case. |
| 06:11.08 | infinity1 | wait. i found a bug in my syntax. |
| 06:11.39 | Qwell | SkramX: parsing a csv file from php is the easiest thing in the world to do |
| 06:11.51 | SkramX | Yea |
| 06:11.58 | Netgeeks | SkramX: http://www.voip-info.org/tiki-index.php?page=Asterisk+CDR+Areski+GUI |
| 06:12.04 | Maarken | well, not quite. parsing it in perl is easier. :) |
| 06:12.11 | *** join/#asterisk ZeeLax (~zeelax@rxgw.network.kz) |
| 06:12.12 | SkramX | :D |
| 06:12.18 | infinity1 | ah! it doesn't work. |
| 06:12.22 | Qwell | Maarken: ugh, don't remind me |
| 06:12.35 | Qwell | Maarken: I tookover a bunch of perl scripts at work, that do just that... |
| 06:12.47 | Qwell | assclowns are using substring, TWO HUNDRED times... |
| 06:12.59 | Qwell | to parse a csv file |
| 06:13.02 | Maarken | er |
| 06:13.03 | Beccara | anyone got any experence building the zaptel modules in debian? |
| 06:13.15 | Maarken | Qwell: that's not funny. |
| 06:13.20 | Qwell | Maarken: a file with 16,000 rows |
| 06:13.25 | Qwell | so, thats...how many times? |
| 06:13.33 | Maarken | "too many" |
| 06:13.36 | Qwell | erm, no, my mistake |
| 06:13.52 | Qwell | it was like 1,600 bytes per row, 500mb...something like 315,000 rows |
| 06:13.52 | Maarken | $10 says they used to be C/C++ programmers. |
| 06:14.01 | SkramX | hahaha |
| 06:14.25 | Qwell | shit takes 2.5 hours to run...per script :( |
| 06:15.09 | Maarken | perl is very good about letting you shoot yourself in the genitals. |
| 06:15.16 | infinity1 | anyone see something wrong : While($[${LOOP} < 4]) |
| 06:15.37 | Qwell | and, the damn script isn't even incrementing a counter or anything. its all hardcoded substrings |
| 06:15.54 | Qwell | so, what happens if one value grows in size? You guessed it! |
| 06:16.05 | Qwell | </rant? |
| 06:16.11 | Qwell | s/\?/\>/ |
| 06:17.06 | Maarken | I imagine this weekend I'll get to play with the AGI PMs for CIDname fu |
| 06:17.09 | harryvv | Beccara tried to make debian load the modeles every time i had to reboot it and never could. no one had any ideas why it did not load them. |
| 06:17.29 | Beccara | i can even compile them harryvv |
| 06:18.24 | *** join/#asterisk aiks (~aiks@159.148.227.104) |
| 06:18.46 | hardwire | heh |
| 06:18.54 | hardwire | I could store the Caller ID in DNS |
| 06:18.55 | hardwire | weird |
| 06:18.59 | hardwire | poor mans enum |
| 06:19.25 | aiks | hi, can i configure channels in zapata.conf in the form of channel => 0/2 => 0/15 |
| 06:19.30 | hardwire | I should really consider SQL at this point |
| 06:19.35 | hardwire | for quite a few things |
| 06:19.40 | aiks | sorry, channel => 0/2 - 0/15 |
| 06:19.48 | Qwell | aiks: 0-2, 3-15, 16-24, etc |
| 06:19.51 | Maarken | hardwire: especially if you're pondering doing unholy things to DNS, yeah. |
| 06:19.59 | hardwire | yeah |
| 06:20.03 | hardwire | its not unholy :) |
| 06:20.05 | hardwire | enum uses it :) |
| 06:20.11 | aiks | i have a configuration of bchans from 2-15 |
| 06:20.12 | Qwell | aiks: unless I'm misunderstanding the question |
| 06:20.21 | aiks | now in zapata i have |
| 06:20.30 | aiks | group=1 |
| 06:20.30 | aiks | channel => 2-15 |
| 06:20.42 | infinity1 | ok. i've tested WHile/EndWhile. It doesn't work. |
| 06:20.57 | Qwell | aiks: ok, and? |
| 06:20.57 | aiks | however asterisk message log says: Aug 2 20:39:54 WARNING[3417]: Ring requested on unconfigured channel 0/3 span 1 |
| 06:21.20 | Maarken | hardwire: I looked at SQL. I'll probably end up with just a flat text file. the upside of a 2 user system |
| 06:21.25 | aiks | so i wonder what does that leading zero before slash mean |
| 06:21.32 | hardwire | Maarken: not I |
| 06:21.35 | hardwire | I need to like |
| 06:21.43 | hardwire | have an sql system |
| 06:21.56 | hardwire | to store the snom sip phone information |
| 06:21.58 | hardwire | per mac address |
| 06:22.19 | hardwire | and then generate a sip.conf out of that somehow |
| 06:22.32 | hardwire | as well as some includes for my extensions.conf |
| 06:22.40 | Maarken | sounds...fun? |
| 06:22.43 | hardwire | not really |
| 06:22.47 | hardwire | sounds like its too damn custom |
| 06:23.01 | harryvv | hhe |
| 06:23.07 | aiks | :) |
| 06:23.26 | hardwire | blah |
| 06:23.27 | hardwire | yeh |
| 06:23.28 | hardwire | bye |
| 06:24.59 | harryvv | bye |
| 06:25.01 | harryvv | :) |
| 06:27.56 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 06:27.59 | SkramX | twisted? |
| 06:29.42 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 06:34.09 | derrick_ | i don't think humans are designed to be separated |
| 06:47.28 | aiks | finaly it runs |
| 06:47.46 | aiks | after 3 days of struggle with our telco to prove it was therio fault |
| 06:47.48 | aiks | their |
| 06:47.55 | aiks | it finally runs |
| 06:48.05 | SkramX | heh! |
| 06:48.42 | aiks | i even moved dchan to channel 1 of E1 to get it all working |
| 06:49.36 | aiks | as somwhere inbetween one of telcos hardware was craping that channel with their own stuff (not HDLCFCS at all) |
| 06:49.51 | *** part/#asterisk ady (~root@202.5.145.13) |
| 06:50.08 | *** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo) |
| 06:53.51 | Beccara | i've just installed asterisk and done a basic config with 1 extension and 1 trunk, whenever i try and dail anything i always get "499 not acceptable here" |
| 06:54.12 | Beccara | i have this in my extensions file "exten => 1,1,Dial(Zap/1/) |
| 06:54.13 | Beccara | " |
| 06:54.18 | aiks | what does /var/log/asterisk/messages says |
| 06:54.19 | aiks | ? |
| 06:54.20 | *** join/#asterisk r3d5un (~r3d5un@80.121.192.21) |
| 06:55.26 | Beccara | Aug 3 20:09:15 NOTICE[4775]: No compatible codecs! |
| 06:55.59 | aiks | hmmm |
| 06:58.08 | X-Rob | Beccara - now, what do you think that error means? |
| 06:58.50 | Beccara | i know, stupid me, the sip client was in g729 only mode |
| 06:59.27 | derobert | FYI: trying to 2.6 rtc patch for ztdummy, it seems to be helping with the SIP drop-outs |
| 07:01.12 | r3d5un | quick question. Do i need to purchase the g729 codec if i want to call from a Grandstream BT-100 to a Cisco 7960 (which both speak g729). Or is the free (passthrough) version enough for this scenario? |
| 07:02.01 | X-Rob | derobert, it shouldn't. |
| 07:02.11 | X-Rob | ztdummy _only_ affects moh and meetme |
| 07:02.22 | derobert | X-Rob: yeah, this is sip dropouts in meetme |
| 07:02.34 | derobert | X-Rob: sorry for not being clearer! |
| 07:02.53 | X-Rob | r3d5un, IF you want asterisk to do anything with the call apart from passing it through (eg, voicemail, attended trasnfers), you need a codec. |
| 07:03.25 | X-Rob | derobert - in that case, yes, use the RTC patch 8) Clients will slowly slide behind and end up with terrible latency |
| 07:03.31 | X-Rob | (without it) |
| 07:03.40 | aiks | :) |
| 07:03.42 | r3d5un | x-rob, thanx just for now i need no voicemal or other "special" features, just a simple call |
| 07:03.48 | aiks | it sounded like promise |
| 07:03.51 | aiks | of the patch |
| 07:03.53 | aiks | :)))) |
| 07:04.19 | derobert | X-Rob: well, I found that without it, occaisionaly asterisk would just go into "send everyone silence" mode for a few seconds... very annoying. Seems fixed with it. |
| 07:04.20 | X-Rob | r3d5un why not use the free one? |
| 07:04.29 | derobert | X-Rob: that patch needs to be merged, dang it. |
| 07:04.48 | X-Rob | http://www.aussievoip.com.au/wiki-G729 |
| 07:04.54 | *** join/#asterisk _ioscanner (~ioscanner@c-67-162-251-133.hsd1.tx.comcast.net) |
| 07:05.09 | X-Rob | At the bottom of both the G729 and G723.1 page is a codec_foo.so that's already been compiled |
| 07:05.21 | *** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no) |
| 07:05.38 | X-Rob | derobert, the RTC patch? Well, there's some grief with less than 2.6.13 - if you use 2.6.13 it _is_ turned on |
| 07:06.00 | derobert | X-Rob: what kind of grief with less than 2.6.13? |
| 07:06.08 | derobert | I'm on 2.6.8 + debian patches |
| 07:06.21 | X-Rob | No idea. But with 2.6.13 or greater the RTC stuff is automatically used |
| 07:08.16 | derobert | hmmm, wasn't applied to the zaptel modules in debian sarge.... |
| 07:08.21 | derobert | anyway, working great so far :-) |
| 07:08.27 | _ioscanner | I am getting an error with wcfxo (all driver in the zaptel set) I compile them fine but when I try to run them I get: |
| 07:08.49 | _ioscanner | zaptel: disagrees about version of symbol hdlc_open |
| 07:08.59 | _ioscanner | zaptel: Unknown symbol hdlc_open |
| 07:09.20 | _ioscanner | this repeats many time for different names ppp_output_wakeup etc.. |
| 07:09.28 | *** join/#asterisk razu (~razu@61kontor.ewn.ee) |
| 07:09.29 | _ioscanner | anyone knwo how to fix this problem |
| 07:13.12 | derobert | Well, now that it seems to working, want to try throwing some load at it... |
| 07:13.26 | derobert | sip:1000@planck.derobert.net |
| 07:13.42 | derobert | hit in option 4 for the meetme room |
| 07:14.29 | *** join/#asterisk Myshenka (~gunde@217.9.101.85) |
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| 07:37.34 | *** part/#asterisk pbxbart (~pbxbart@ip-80-226-238-235.vodafone-net.de) |
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| 07:41.30 | nounours_fr | hi |
| 07:43.01 | Zeeek | salut |
| 07:43.12 | _ioscanner | Howdy |
| 07:45.08 | Zeeek | wassup? |
| 07:45.24 | _ioscanner | broken... |
| 07:45.41 | _ioscanner | can't get cvs stable to work for some reason |
| 07:45.53 | Zeeek | why not? |
| 07:46.14 | _ioscanner | I get unknown symbol |
| 07:46.27 | _ioscanner | and unknown symbol |
| 07:46.30 | Zeeek | is this a new install or ? |
| 07:46.37 | _ioscanner | when I start zaptel and wcfxo or any zaptel |
| 07:46.50 | _ioscanner | I am installing from cvs stable |
| 07:47.15 | *** join/#asterisk cfrank (~cfrank@wsip-24-234-137-140.lv.lv.cox.net) |
| 07:47.21 | _ioscanner | says ppp_ input hdlc and many more have unresolved symbols |
| 07:47.45 | _ioscanner | I think I have seen this before about a year ago, but I can't find anything about it. Or haven't yet |
| 07:47.54 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 07:55.01 | *** join/#asterisk r3d5un (~r3d5un@80.121.192.21) |
| 07:55.50 | Zeeek | doesn't mean anythiong to me, sorry |
| 07:56.18 | *** join/#asterisk RoyK (~roy@142.80-203-78.nextgentel.com) |
| 07:57.09 | Beccara | hmm |
| 07:58.21 | _ioscanner | that is okay. I pulled some older zap drivers I build for this kernel. a week ago. seems like a zaptel driver problem. they seem to mess it up every other week so it will not build if you are using hdlc and zap net with t-1 cards |
| 08:00.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:07.32 | Beccara | is there anyway to try and clean up the sound from a zap line? |
| 08:24.21 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 08:24.49 | langals | Hi there - I am wanting to get a Digium card for meetme timing. Is any Digium card suitable for this? |
| 08:26.10 | *** join/#asterisk juliette (~Juliette@217.146.227.2) |
| 08:27.13 | juliette | hi |
| 08:28.33 | juliette | do somebody know iax ? |
| 08:29.06 | Zeeek | ask juliette and you shall see |
| 08:35.34 | juliette | i start with iax, and i don't know how send a simple message with. |
| 08:35.49 | juliette | i want just send un string |
| 08:35.52 | *** join/#asterisk PhreeStyle (me@cpe-24-221-52-165.az.sprintbbd.net) |
| 08:38.42 | Zeeek | first, you have to wear a string |
| 08:39.25 | opus_ | can i use manager api to put a sip phone on hold? |
| 08:39.33 | opus_ | like, is this setting a channel variable? |
| 08:39.45 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
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| 09:14.56 | clive- | does anyone know if its possible to perform packet loss concealemnt on G729? |
| 09:16.00 | juliette | does anyone can help me with iax ? |
| 09:16.48 | Zeeek | juliette pleaseexamplain exactly what you want to do |
| 09:18.40 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 09:19.27 | *** join/#asterisk basta (~sc@213-156-52-98.fastres.net) |
| 09:20.42 | juliette | i wish send a string to my phone, every 20s to display this string on the screen of my sip phone. And i must use IAX. |
| 09:21.37 | *** join/#asterisk mkl1525 (~mkl1525@212.80.239.159) |
| 09:21.38 | Zeeek | That would be a good mailing list question |
| 09:22.16 | Zeeek | I don't know if IAX2 can send arbitrary stringsd or whether you need a trick like CID manipulation. Have you seen the spec? |
| 09:22.40 | juliette | my problem is to send this string by IAX. |
| 09:23.20 | juliette | yes, i see the spec |
| 09:26.20 | mkl1525 | Hi! Trying to get MusicOnHold to play a mp3 file, have installed (the real) mpg123 added "exten => 18,3,WaitMusicOnHold(30) ;exten => 18,3,MP3Player(/usr/share/asterisk/mohmp3/musik.mp3)" in extensions.conf, musiconhold.conf has "default => quietmp3:/usr/share/asterisk/mohmp3" set. When I uncomment the MP§Player line I hear the file but when I use WaitMusicOnHold I just hear silence - any thoughts where the problem could be? |
| 09:28.53 | *** join/#asterisk PhreeStyle (me@cpe-24-221-52-165.az.sprintbbd.net) |
| 09:30.07 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 09:32.27 | *** join/#asterisk KeX_WorX (~chris@83-65-129-46.paris-lodron.xdsl-line.inode.at) |
| 09:32.39 | KeX_WorX | hi |
| 09:32.43 | r3d5un | hi |
| 09:32.44 | *** part/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg) |
| 09:32.53 | KeX_WorX | is it possible to print timestamps (ms) in asterisk ? |
| 09:35.40 | *** join/#asterisk forkqueue (~sam@2001:4bd0:2024:0:0:0:0:10) |
| 09:35.57 | opus_ | yes there is a variable for that |
| 09:36.08 | opus_ | you can use NoOp to display variables to the asterisk cli |
| 09:36.30 | opus_ | hmmm.. how do I send SIP packets like 'hold' with manager api or command clI? |
| 09:37.01 | *** join/#asterisk kks (~kks@202.73.8.130) |
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| 09:42.38 | *** join/#asterisk Delvar (~irc@host-83-146-53-34.bulldogdsl.com) |
| 09:43.05 | Delvar | ok im buggered, what does this error mean on my TDM card? 'ProSLIC 3210 version 2 is too old' |
| 09:43.40 | Delvar | i tried a google but didnt turn up anything usfull |
| 09:47.39 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
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| 09:51.31 | Beccara | anyone tell me quickly what ports i need to forward on my router to enable SIP clients to connect to me * box which is behind nat |
| 09:52.02 | mkl1525 | Is there some command to show what asterisk is doing at the moment (something like top) cause it's eating 70% of my cpu without any calls? |
| 09:59.12 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
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| 10:02.07 | *** join/#asterisk xylome (~asterisk@hg-msq-hol.levigo.de) |
| 10:08.07 | Delvar | ok im buggered, what does this error mean on my TDM card? 'ProSLIC 3210 version 2 is too old' |
| 10:08.27 | RoyK | it means get a new one .P |
| 10:08.55 | *** part/#asterisk derobert (~anthony@Maxwell.derobert.net) |
| 10:09.27 | *** join/#asterisk basta (~sc@213-156-52-98.fastres.net) |
| 10:10.11 | opus_ | hmmm. |
| 10:10.23 | opus_ | app_voicemail really doesn't support database password changes... |
| 10:10.59 | opus_ | or am I wrong? |
| 10:15.41 | *** join/#asterisk hennker (flullup@dsl-213-023-250-073.arcor-ip.net) |
| 10:16.32 | Delvar | RoyK: bugger... |
| 10:16.57 | Delvar | anything else that can cause this? |
| 10:17.18 | *** join/#asterisk Sadaat (sadat@203.215.180.254) |
| 10:17.23 | Delvar | need a new card or module? |
| 10:17.25 | *** join/#asterisk FITA1 (~m_ahmed@202.5.145.50) |
| 10:17.35 | FITA1 | hi all |
| 10:17.59 | Sadaat | hi everybody |
| 10:21.50 | FITA1 | I want to ask that can I change my callerid on outgoing call to pstn number in asterisk |
| 10:22.30 | Zeeek | how are yopu connected? ISDN, analog...? |
| 10:22.45 | FITA1 | I mean I don't want to show my telephone number, instead I want to send some other number ... |
| 10:22.49 | FITA1 | ISDN |
| 10:23.13 | *** join/#asterisk emp (~emp@70.57.239.37) |
| 10:23.17 | FITA1 | ISDN at both ends |
| 10:24.13 | Beccara | what ports do i have to forward to my ASTERISK box to get it working behind NAT? |
| 10:25.20 | FITA1 | <Beccara>: I think 4569, 5036, 5060 and 5038 |
| 10:25.36 | Beccara | thanks |
| 10:26.17 | *** join/#asterisk lilalinux (e-trolley@deepthroat.deswahnsinns.de) |
| 10:26.32 | lilalinux | Hey guys |
| 10:26.39 | lilalinux | does asterisk support ipv6? |
| 10:26.43 | RoyK | no |
| 10:26.49 | lilalinux | thx |
| 10:26.51 | RoyK | but does that matter? |
| 10:26.59 | RoyK | nothing else supports ipv6 :) |
| 10:27.37 | lilalinux | RoyK: http://www.sixxs.net/faq/ipv6/?faq=coolthings |
| 10:27.44 | FITA1 | RoyK u there, can you answer my question ??? |
| 10:28.42 | RoyK | FITA1: setcallerid? |
| 10:29.02 | FITA1 | It is not working ... |
| 10:29.38 | Zeeek | may depend on thephone company |
| 10:29.40 | Sadaat | FITA1 . . do you dial out by using your ISDN lines? |
| 10:29.51 | Sadaat | and you get ISDN Number at other end??? |
| 10:30.13 | RoyK | FITA1: you're usually not allowed to set another number than your own |
| 10:30.26 | FITA1 | I m using ast_request_and_dial and giving callerid in the argument |
| 10:30.26 | FITA1 | yeah sadaat I m |
| 10:30.55 | FITA1 | RoyK: why??? |
| 10:31.06 | Sadaat | that's not possible then... |
| 10:31.18 | Sadaat | because telco doesnt allow to do that.. |
| 10:32.09 | RoyK | FITA1: obviously they don't want you to spoof it |
| 10:34.01 | FITA1 | So, This mean I cann't change a callerid in an outgoing call ... am I right ??? |
| 10:35.03 | Sadaat | yes, until you are not a TELCO... |
| 10:35.20 | FITA1 | sadaat well said :) |
| 10:35.40 | FITA1 | Is there any possiblity to change callerid if I m not a telco ... |
| 10:36.27 | Sadaat | nope |
| 10:38.17 | juliette | :'( someone can help me with IAX ? |
| 10:38.39 | X-Rob | Nope. |
| 10:38.54 | X-Rob | Someone can possibly help you if you tell us explicitly and in great detail what your problem is. |
| 10:38.59 | X-Rob | but we can't help you 'wish IAX' |
| 10:39.06 | X-Rob | 'with IAX' |
| 10:39.08 | X-Rob | even |
| 10:39.12 | Sadaat | :-D |
| 10:39.15 | *** join/#asterisk kb1_kanobe (~jsmith@h24-207-96-50.cst.dccnet.com) |
| 10:40.32 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
| 10:41.00 | juliette | i want send a string via iax to my phone every 20s |
| 10:41.07 | *** join/#asterisk Megyu (balazs@b.tmit.bme.hu) |
| 10:41.12 | Megyu | hi |
| 10:41.17 | RoyK | <PROTECTED> |
| 10:41.17 | RoyK | Segmentation fault (core dumped) |
| 10:41.21 | RoyK | :S |
| 10:41.30 | X-Rob | Woo! You win! |
| 10:41.37 | kb1_kanobe | morning all. |
| 10:41.38 | Megyu | does anybody knows |
| 10:42.00 | X-Rob | juliette - you have a PA1688 phone I'm guessing, and you're shitty about the way the screen changes all the time? |
| 10:42.01 | Megyu | whether asterisk supports usb-phones? |
| 10:42.30 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 10:42.49 | Sadaat | usb-phones connected to asterisk box or any other machine? |
| 10:42.51 | kb1_kanobe | Any ideas why I wouldn't get audio across a call that's 'native bridged' at the IAX level? Works fine if the call goes out a zap interface and back in on a seperate channel. |
| 10:42.58 | bublbobl | Hi all. The beta site is down :-( . Anyone knows how to set a GrandStream budgetone to fullduplex (I have too many collisions in half :-$) |
| 10:43.38 | Megyu | well, both :) |
| 10:44.03 | Sadaat | bublbobl : Grandstream is already fullduplex (I use them)...u may have problem due to NAT or something else.. |
| 10:44.27 | X-Rob | Sadaat - if your GXP isn't working at full duplex, your switch doesn't support N-Way negotiation. |
| 10:44.38 | X-Rob | WHich is an IEEE/802.1 standard. |
| 10:44.50 | X-Rob | Buy a new switch. You're beating a dead horse if you try to work around it. |
| 10:45.02 | Sadaat | meee????!!!!! |
| 10:45.06 | Megyu | So I think remote machine with usb phone can work with a VoIP client |
| 10:45.06 | X-Rob | s/GXP/Budgettone/ |
| 10:45.17 | X-Rob | wups |
| 10:45.28 | X-Rob | I mean bublbobl |
| 10:45.41 | bublbobl | Sadaat> :-o So i shouldn't have collisions |
| 10:45.41 | Sadaat | X-Rob: I swear I dont have any problem with Asterisk...and grandstream . . .kekeke |
| 10:45.42 | X-Rob | bublbobl - if your Budgettone isn't working at full duplex, your switch doesn't support N-Way negotiation. |
| 10:45.54 | X-Rob | Sadaat - I love grandstream. Well. A but. |
| 10:45.55 | X-Rob | a bit |
| 10:46.00 | bublbobl | X-Rob> oki, I will have a look at 802.1 settings in the switch |
| 10:46.01 | X-Rob | I love snom _lots_ more tho |
| 10:46.09 | X-Rob | bublbobl - you have a managed switch? |
| 10:46.19 | X-Rob | eg, you can telnet to the switch? |
| 10:46.48 | bublbobl | X-Rob> Yes, I can, but it is private lan and I don't have access by now :-( |
| 10:46.55 | X-Rob | what sort of switch is it? |
| 10:47.25 | *** join/#asterisk phil0u (~philou@81.56.194.193) |
| 10:47.30 | phil0u | 'lo |
| 10:47.44 | X-Rob | hi. |
| 10:49.07 | bublbobl | It is manageable, a dlink 3226s |
| 10:49.17 | Megyu | so the question is that, can directly connected usb phone used with asterisk? |
| 10:49.26 | X-Rob | Megyu - the simple question is yes |
| 10:49.36 | Megyu | :) thanx |
| 10:49.37 | X-Rob | IF you're going to ask 'how?', that answer is far harder. |
| 10:49.54 | Megyu | hm |
| 10:49.59 | Megyu | isnt it simple? |
| 10:50.17 | X-Rob | how are you _planning_ on using it? |
| 10:51.18 | Sadaat | bublbobl: is your asterisk at public IP Address? behind the firewall? |
| 10:52.30 | memic | anybody has example config for asterisk acting as sip server which has 2 b channels for incoming calls and outgoing calls which shoudl forwareded to sip phones? |
| 10:52.40 | bublbobl | Sadaat> it is in a private LAN but outgoes to PSTN from this LAN, no NAT is used |
| 10:52.51 | Megyu | it is plugged to a machine which has an asterisk installed, and an other one which is plugged to an other asterisk, and the two are connected via vpn |
| 10:53.06 | *** join/#asterisk jiro5281 (~anton281@203.177.242.192) |
| 10:56.12 | jiro5281 | hi guys! just want to ask how the manager api works...4 sample...btw...im implementing this is php .i want to set cdruserfield=67 if channel:Local/913104925517@outgoing |
| 10:56.24 | FITA1 | when I make an outgoing call (pstn number) and request a channel * gives a channel named Zap/1-1. I have a pri line and 6 different number allocated on it. How can I get the number to which Zap/1-1 actually refers ??? |
| 10:56.55 | kb1_kanobe | FITA1: 'pri debug span 1' will show you the setup information. |
| 10:57.35 | kb1_kanobe | Remember that a PRI consists of channels and the channel-to-number mapping is arbitrary. |
| 10:57.35 | FITA1 | I mean is there any field in channel structure which can tell me the actual number |
| 10:58.53 | FITA1 | or any other method or application in aterisk which can return the actual numer ??? |
| 10:59.14 | kb1_kanobe | I'm not certain I understand your question. |
| 10:59.37 | kb1_kanobe | If you place an outgoing call on a PRI service then it will go out on one of many channels. |
| 11:00.20 | kb1_kanobe | Unless your service provider allows you to set the originating caller ID information then it will appear to come from the 'pilot number' of the PRI, regardless of the channel it's on. |
| 11:00.24 | *** join/#asterisk jtza8 (~jens@tbnb-165-198-84.telkomadsl.co.za) |
| 11:01.24 | jtza8 | I don't see anything in the topic, but where do I goto if I'm new to VoIP? |
| 11:01.25 | FITA1 | you mean for every channel the callerid will be the same |
| 11:01.29 | kb1_kanobe | If you instead have a T1 (ie. non-PRI) with a seperate line installed on each timeslot then, yes, you would be able to map out which number was on which timeslot. |
| 11:01.55 | kb1_kanobe | Yes, with PRI unless you can set your originating callerID (varies by provider) then it will appear to come from the pilot number. |
| 11:02.03 | Megyu | anyway, do I need 2 ASTERISK for both sites, when I connect them with VPN? And there are also analogue PBXs atthe sites, which must be connected. |
| 11:03.13 | kb1_kanobe | jtza8: Cisco have some very good, if cisco-centric, documentation on their website for their VoIP products. Most of it applies across all systems. |
| 11:03.54 | kb1_kanobe | check the wiki (http://www.voip-info.org) if you want to go in at the deep end. |
| 11:05.50 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 11:07.55 | kb1_kanobe | Any ideas why I wouldn't get audio across a call that's 'native bridged' at the IAX level? Works fine if the call goes out a zap interface and back in on a seperate channel but no audio passes when it's 'native bridged'... |
| 11:11.54 | kb1_kanobe | Is anyone here running the latest cvs-head? |
| 11:12.22 | r3d5un | nope not me |
| 11:14.03 | xming | yes |
| 11:14.13 | kb1_kanobe | How is it at the moment? Stable? |
| 11:14.51 | xming | I haven't stress test it but it isn't crshing |
| 11:15.23 | xming | I have always run CVS HEAD on lightly loaded servers |
| 11:16.18 | kb1_kanobe | Do you use zap and/or iax? |
| 11:18.20 | FITA1 | kb1_kanobe: pri_debug is not giving any info about number send as a callerid |
| 11:19.34 | kb1_kanobe | FITA1: Look for the 'Calling Number' item in the SETUP message. |
| 11:21.11 | xming | ~.~. |
| 11:23.17 | phil0u | one question: why does CVS HEAD disables BUSYDETECT by default, in the makefile ? Wasn't the case in 1.0.9 right ? Problem with busydetect ? |
| 11:24.14 | xming | I am using zap/sip/iax |
| 11:25.11 | kb1_kanobe | Good to know. I've got a wierdness and am still running a mid-May cvs head. Probably time to upgrade, assuming it doesn't deadlock when you hit tab or anything silly... ;-) |
| 11:30.47 | FITA1 | Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) |
| 11:30.47 | FITA1 | > Presentation: Number not available (67) '' ] |
| 11:31.02 | FITA1 | kb1_kanobe : I m getting this |
| 11:31.12 | kb1_kanobe | Ah, you're not even transmitting any callerID information then. |
| 11:31.55 | kb1_kanobe | Which means you're presenting the pilot number to whomever you're calling. |
| 11:32.16 | FITA1 | I did but * is not sending that caller id instead it sends the pilot number |
| 11:33.16 | kb1_kanobe | Hmmm.m.... your telco has to support you sending your own custom callerID. It's ususally a for-fee service. |
| 11:33.16 | FITA1 | I gave and got Presentation: Number not available (67) '123456' but on the recieving side I got the number 02070162222 |
| 11:34.04 | kb1_kanobe | yeah. That's because of your telcos restriction. Shortly after the '123456' you'll probably see a 'PROTOCOL ERROR' reply message - that's the exchange complaining about the rejected callerID. |
| 11:37.02 | FITA1 | let me check this ... whether i m getting this error or not |
| 11:42.27 | Hmmhesays | fuck you united |
| 11:42.33 | Hmmhesays | argh!@ |
| 11:55.47 | kb1_kanobe | g'night all |
| 12:00.04 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 12:04.25 | Mimmus | hi |
| 12:04.43 | Mimmus | I'm having problem with a recent buyed TE410P Digium T1/E1 card |
| 12:05.05 | Mimmus | driver is loaded correctly but no interrupts are seenn |
| 12:05.43 | *** join/#asterisk pa (~Paolo@pa.user) |
| 12:07.10 | X-Rob | mimmus - email support@digium.com |
| 12:07.46 | Mimmus | X-Rob: I emailed some hours ago... I hope in a response during USA working hours! |
| 12:08.27 | Mimmus | X-Rob: in the meanwhile, I'd like to investigate by myself |
| 12:10.28 | Mimmus | but all people are at PBX Developers Conference... |
| 12:13.01 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 12:13.40 | *** join/#asterisk |dennis| (~dennis@200.32.215.82) |
| 12:17.16 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:19.26 | *** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net) |
| 12:36.25 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
| 12:40.22 | *** part/#asterisk hennker (flullup@dsl-213-023-250-073.arcor-ip.net) |
| 12:43.05 | *** part/#asterisk asteriskmonkey (~phil@69.158.154.80) |
| 12:43.47 | RoyK | anyone here that knows where i can get cheap termination to .ru and .fi? |
| 12:44.52 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
| 12:48.24 | *** join/#asterisk tengulre (~tengulre@222.90.140.207) |
| 12:51.42 | *** join/#asterisk pdugas (~pdugas@h102.73.40.69.ip.alltel.net) |
| 12:52.29 | Dibbler_ | Anyone know why my Q931 isn't working :-( |
| 12:53.40 | mishehu | maybe you didn't ask nicely enough |
| 12:53.53 | RoyK | Dibbler_'s never nice |
| 12:54.22 | Dibbler_ | No more sausagesinabun for you ;) |
| 12:54.49 | X-Rob | C.M.O.T Dibbler |
| 12:58.12 | *** join/#asterisk RomDump (romdump@norge.freeshell.ORG) |
| 12:58.29 | *** join/#asterisk _omer (omer@203.215.180.254) |
| 12:58.54 | _omer | hi |
| 12:59.23 | _omer | have anybody seen this problem??? |
| 12:59.23 | _omer | Aug 3 16:33:12 NOTICE[27585]: app_dial.c:977 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) |
| 12:59.24 | _omer | <PROTECTED> |
| 12:59.26 | X-Rob | And, anyway, what did I do? |
| 12:59.45 | X-Rob | omer, show us the 'Dial' line |
| 13:00.25 | _omer | Executing Dial("SIP/00011284-aa2f", "SIP/0013602276003@outgoing") in new stack |
| 13:00.38 | *** join/#asterisk funxion (~nunya@mtnuser.icgws.com) |
| 13:00.54 | bjohnson | has the use of the term "meetme calling" been discussed here? http://www.digitalvoice.bell.ca/RatesAndPlans/ |
| 13:01.32 | _omer | X-Rob: exten => _X.,2,Dial(SIP/0013602276003@outgoing) |
| 13:01.35 | bjohnson | Bell Canada's new VOIP offering ^^ |
| 13:01.41 | *** join/#asterisk coppice (~chatzilla@125.166.17.210.dyn.pacific.net.hk) |
| 13:01.45 | X-Rob | omer, that's wrong |
| 13:03.16 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 13:03.40 | _omer | and correct syntax? ... it works when I dial out with FWD..... |
| 13:03.58 | X-Rob | with FWD you dial SIP/fwd/number |
| 13:04.36 | X-Rob | eg, you've got to tell asterisk where to send the sip traffic |
| 13:05.29 | _omer | and in my dial line.....Asterisk is dialing out from "outgoing" peer..... |
| 13:05.58 | X-Rob | eg, Dial(SIP/broadvoice/number) |
| 13:06.09 | X-Rob | or Dial(SIP/whateverthehellyoucalledthepeer/number) |
| 13:06.16 | X-Rob | I'm sure you can figure it out from here |
| 13:07.07 | _omer | alright...let me check plz |
| 13:08.20 | djin | _omer, what is 'outgoing'? |
| 13:08.33 | X-Rob | Dial(SIP/user:password@sip.provider.com/number) |
| 13:08.48 | _omer | [outgoing] |
| 13:08.59 | djin | didn't you mean exten => _X.,2,Dial(Local/0013602276003@outgoing) |
| 13:10.10 | mishehu | bah. |
| 13:10.53 | djin | hab |
| 13:11.48 | _omer | same problem... |
| 13:12.31 | *** join/#asterisk Mavvie (edwin@252-131-222-203.rev.techex.net.au) |
| 13:12.34 | RoyK | http://www2.gamesville.lycos.com/html_poke/poke_penguin.htm |
| 13:13.13 | djin | Unable to create channel of type 'SIP' (cause 3) ??? |
| 13:13.19 | _omer | correct |
| 13:13.39 | djin | The re-read X-Rob's explaination |
| 13:13.47 | _omer | I have two different SIP Service providers....and it works with one but not with other... |
| 13:14.06 | X-Rob | try 'sip show registry' |
| 13:14.21 | X-Rob | the first google result for that says it's coz it's not registering |
| 13:15.24 | _omer | 213.170.92.166:5065 00011284 1785 Registered |
| 13:15.24 | _omer | 203.128.7.14:5060 2106436 105 Registered |
| 13:15.39 | mut | hmm |
| 13:15.44 | _omer | hmmmmm |
| 13:15.49 | mut | where in a resume do you usually put salary requirements? |
| 13:16.07 | gambolputty | wouldn't that go in a cover letter? |
| 13:16.08 | _omer | it works with 2106436 ...but not with 00011284 |
| 13:16.08 | newl | nowhere |
| 13:16.11 | gambolputty | for a specific job? |
| 13:16.17 | mut | ya for a specific job |
| 13:16.22 | mishehu | mut: you assume that asterisk users know how to write resumes? |
| 13:16.25 | gambolputty | and only if they ask you for what your requirements are? |
| 13:16.30 | mishehu | we know how to run voip systems! |
| 13:16.30 | newl | salary is usually dicussed after you're shortlisted :) |
| 13:16.32 | mut | mishehu: i assume people do... |
| 13:16.49 | mishehu | mut: you'd be surprised then. I've done hiring before. |
| 13:16.50 | mut | it says salary requirement required in resume |
| 13:16.59 | gambolputty | unreal |
| 13:17.17 | gambolputty | then email a word file of the resume |
| 13:17.21 | gambolputty | just for that one job |
| 13:17.24 | mut | yea |
| 13:17.25 | newl | If they need to ask, they can't afford you. 8) |
| 13:17.28 | gambolputty | but it usually goes in cover letter |
| 13:17.34 | mut | thats what i'm goin to do |
| 13:17.37 | newl | heh nah, do it in OO.org Writer format. :) |
| 13:17.58 | gambolputty | yes, then force all the secretaries to upgrade overnight |
| 13:18.03 | mut | newl: probly but i'm willing to pack up on this job for any other at this point |
| 13:18.09 | mishehu | I use pdf |
| 13:18.15 | gambolputty | what kind of job is it? |
| 13:18.17 | gambolputty | a * job? |
| 13:18.27 | mut | no |
| 13:18.37 | mut | developer job |
| 13:18.37 | gambolputty | job description? |
| 13:18.40 | gambolputty | ok |
| 13:18.44 | mut | web dev mainly |
| 13:18.48 | gambolputty | how much they want? |
| 13:18.56 | mut | ? |
| 13:19.14 | gambolputty | your salary |
| 13:19.42 | newl | web devs typically get $45-55k here in Western Australia |
| 13:19.47 | mut | i should probqbly put like 30k |
| 13:19.57 | gambolputty | ok |
| 13:20.09 | newl | doing market research for your geographic location will help too |
| 13:20.18 | mut | it's in the middle of detroit |
| 13:20.24 | newl | heh |
| 13:20.24 | newl | shit |
| 13:20.29 | newl | another detroit person |
| 13:20.40 | mut | i'm in northern michigan right now |
| 13:20.43 | newl | take it at any cost..I hear unemployment there sucks. :) |
| 13:20.47 | X-Rob | bloody amercians |
| 13:20.48 | *** part/#asterisk pigpen (~mark@fw.seamans.cc) |
| 13:20.49 | mut | it does |
| 13:20.56 | Rienzilla | dammit |
| 13:20.59 | mut | job market in michigan really sucks |
| 13:21.05 | newl | mut: I saw the writing on the wall, that's why I left. :) |
| 13:21.07 | X-Rob | newl, if you're on an iiSlam I hate you already. |
| 13:21.08 | Rienzilla | damn asterisk keeps segfaulting :/ |
| 13:21.20 | newl | X-Rob: Eh :) |
| 13:21.32 | gambolputty | due to what rienzilla? |
| 13:21.50 | X-Rob | nayway. Bedtime for me. |
| 13:21.53 | X-Rob | night orl. |
| 13:21.58 | Rienzilla | I don't know exactly, but I suspect it has got to do with chan_misdn |
| 13:22.02 | newl | 'night X-Rob |
| 13:22.14 | _omer | thanks X-Rob .....and bye :) |
| 13:22.22 | Rienzilla | sometimes after or while it calls out to an isdn phone it suddenlyu segfaults |
| 13:22.41 | X-Rob | 557hp with a standard clutch? |
| 13:22.52 | X-Rob | Dude, you _enjoy_ buying clutches and bell housings? |
| 13:22.58 | newl | X-Rob: I'll probably settle for the Phase III though. :) |
| 13:24.04 | newl | X-Rob: hey, stock engine. It's amazing what they can get that 4L straight 6 to do without so much as removing the rocker cover. hehe |
| 13:24.13 | X-Rob | is that the standard intake on the phase iii? |
| 13:24.19 | newl | yep |
| 13:24.58 | *** join/#asterisk exonic (~exonic@209.172.11.54) |
| 13:25.04 | exonic | Hey folks, I have some questions |
| 13:25.04 | X-Rob | so all they're doing is a turbo and (I guess an intercooler) and a chip and getting 200+nm of torque out of it? |
| 13:25.13 | lilalinux | the asterisk installation tells me to: "YOU MUST READ THE SECURITY DOCUMENT" |
| 13:25.16 | lilalinux | Where can I find it? |
| 13:25.23 | newl | X-Rob: they don't touch the turbo. |
| 13:25.40 | X-Rob | lilalinux - /usr/src/asterisk/SECURITY |
| 13:26.15 | newl | X-Rob: well, hardware wise anyway, the ECM data gets altered by that interface controller they install. |
| 13:26.48 | exonic | i'm attempting to call my SIP phone from asterisk using sip URL, asterisk rejects the call saying "handle_request: Failed to authenticate user" |
| 13:26.58 | X-Rob | newl, I'm over car horsepower these days. I've got my diesel prado which would pull-start a kenworth 8) |
| 13:27.06 | SkramX | what are you dialing/ |
| 13:27.13 | SkramX | What's the SIP URL? |
| 13:27.40 | newl | X-Rob: haha In theory, I _should_ be over it too. I'm just a big kid with expensive toys though I guess. 8) |
| 13:27.44 | exonic | Dial(SIP/exonic@<ip>) |
| 13:28.20 | lilalinux | X-Rob: oh, that one :) |
| 13:28.28 | newl | X-Rob: if it were a younger person, mods would be externally visable and the like..us older people like to make sleepers. :) |
| 13:28.48 | exonic | The thing is the user is registered on one of two sip servers, I am using openser to route correctly but asterisk doesn't accept the calls. |
| 13:29.26 | X-Rob | newl - There's a 'VOLVISSAN' roaming around here |
| 13:29.44 | X-Rob | It's a 240GLE Volvo with an GTR-R32 engine and gearbox. |
| 13:29.47 | *** join/#asterisk azrishahril (~azrishahr@60.50.193.76) |
| 13:29.51 | X-Rob | totally stealth. |
| 13:29.56 | newl | X-Rob: haha wicked |
| 13:30.34 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 13:30.39 | SpaceBass | morning |
| 13:30.46 | newl | hmm 9:30..wonder if channel 7 is on schedule tonight or not |
| 13:31.26 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 13:32.47 | *** join/#asterisk Darwin35 (~richard@ip70-179-215-116.dl.dl.cox.net) |
| 13:33.44 | *** join/#asterisk funxion (~nunya@mtnuser.icgws.com) |
| 13:33.54 | Darwin35 | man building a new system is fun but time consu |
| 13:33.54 | Darwin35 | ming |
| 13:34.40 | *** join/#asterisk likwid-- (~likwid@nc-65-41-163-67.dyn.sprint-hsd.net) |
| 13:36.16 | exonic | I'm attempting to dial a user by dialing a sip proxy. asterisk keeps rejecting the call with "handle_request: Failed to authenticate user "6164506862"" the user it's trying to authenticate is my cell phone #, it makes no sense |
| 13:37.46 | thal | exonic: paste your extenstions.conf Dial Statement |
| 13:37.49 | SpaceBass | whats the dial command look like? |
| 13:38.32 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 13:38.33 | Darwin35 | use pastebin |
| 13:38.39 | Darwin35 | not the channel |
| 13:38.43 | thal | whatever |
| 13:38.52 | thal | a single line is perfectly in the channel |
| 13:39.03 | exonic | i would but it's only one line. exten => 6169803444,1,Dial(SIP/office@216.65.177.10|25) |
| 13:39.41 | exonic | 216.65.177.10 is an openser proxy. it is properly routing the call to asterisk but asterisk trys to authenticate my cell #. |
| 13:40.58 | *** join/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 13:42.23 | ChArLeS___ | hey |
| 13:42.32 | ChArLeS___ | does anybody got H323 to work ? |
| 13:42.47 | exonic | give it some work boots |
| 13:43.38 | SpaceBass | so you want that to match the Caller ID of the cell and then dial Office@.... ? |
| 13:45.30 | *** join/#asterisk tla (~tl@almestien.com) |
| 13:45.34 | exonic | it's doing what I want it to, asterisk is just not accepting the call |
| 13:45.42 | exonic | i'll keep diging |
| 13:47.44 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 13:48.57 | ManxPower | ~docs |
| 13:48.57 | jbot | i heard docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 13:49.00 | ManxPower | ~mailinglist |
| 13:49.00 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 13:49.28 | ManxPower | exonic, sounds like you need "insecure=very" |
| 13:49.43 | *** part/#asterisk Maksim (~max@213.142.207.20) |
| 13:49.51 | ManxPower | exonic, see the mailing list archive and sip.conf.sample for more information on that option. |
| 13:50.01 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 13:52.46 | SkramX | has anyone one done SMS with their asterisk box? |
| 13:53.00 | r3d5un | if yes, i would be interested too |
| 13:53.02 | r3d5un | :) |
| 13:53.10 | bublbobl | What value of payload (for g711) would you set to talk to a Grandstream Budgetone, my current setting is CodecPayloadSize: 1280 (bits) and i'm not sure it is ok ? |
| 13:53.17 | *** join/#asterisk jiro5281 (~jiro5281@203.131.137.76) |
| 13:54.37 | Cybertoy | SMS as in the messages on the mobile phones? |
| 13:54.59 | Cybertoy | I'm not aware that this is implemented in asterisk? |
| 13:56.39 | coppice | there is some support for one of the two ETSI standards for SMS over landlines |
| 13:56.46 | lilalinux | Where can I find a tutorial on setting up asterisk? |
| 13:57.17 | djin | isn't it possible to talk G729 with the phone and use G711 with the PSTN provider (conversion on *)? |
| 13:57.28 | *** join/#asterisk jiro5281 (~anton@203.131.137.76) |
| 13:57.37 | Cybertoy | lilalinux, check www.voip-info.org |
| 13:57.43 | bublbobl | lilalinux> if you also want AMP (asterisk management portal) I relied on the README or the INSTALL supplied w/ the tarball |
| 13:57.55 | jiro5281 | hi guys..is SetCDRUserField of manager api working...? |
| 13:58.01 | Cybertoy | djin, only if you have a G729 license on the asterisk box. otherwise not. |
| 13:58.15 | lilalinux | thx |
| 13:58.37 | djin | I have the licence, but get: Dropping incompatible voice frame on Local/31651439140@net-out-3d83,1 of format ulaw since our native format has changed to g729 |
| 13:58.49 | jiro5281 | tried SetCDRUserField in php and doestn reflect in mysql |
| 13:59.07 | djin | and: Aug 3 15:55:47 WARNING[11631]: channel.c:2127 ast_channel_make_compatible: No path to translate from SIP/212.4.194.106-2a8b(4) to Local/31651439140@net-out-3d83,2(256) |
| 13:59.08 | Cybertoy | djin, you have the license installed on * or on the phone? |
| 13:59.15 | djin | Asterisk |
| 13:59.23 | djin | ans phone :) |
| 13:59.25 | djin | and |
| 13:59.38 | Cybertoy | that error message doesn't look like that though. |
| 14:00.07 | djin | ok, perhaps I did something wrong with the installation of the Digium license |
| 14:00.09 | memic | are zaptel configureed via modem.conf?? |
| 14:00.20 | djin | memic, no |
| 14:00.25 | memic | but how |
| 14:00.28 | djin | /etc/zaptel.conf |
| 14:00.39 | djin | ans /etc/asterisk/zapata.conf |
| 14:00.41 | djin | and |
| 14:00.45 | djin | *sigh* |
| 14:00.51 | ManxPower | djin, "show translation" or "show translations" Is there a number for g729? |
| 14:01.11 | djin | mmm, no |
| 14:01.28 | ManxPower | djin, then the g729 license was not installed correctly. |
| 14:01.43 | twisted | ManxPower, or the codec isn't loaded |
| 14:01.44 | djin | Ok, lemme check |
| 14:02.28 | djin | i see |
| 14:02.43 | memic | i cant dialout via my sipphones |
| 14:02.47 | memic | they are registert |
| 14:02.54 | memic | @asterisk |
| 14:03.03 | djin | ManxPower, it now does :) |
| 14:03.03 | *** join/#asterisk Akelavlk (~jansun@82.119.239.141) |
| 14:03.37 | ManxPower | djin, there ya go |
| 14:03.46 | djin | forgot to copy the .so to modules :) |
| 14:03.54 | Akelavlk | Hello, I want have one-two PSTN and 10-20 analog phones, what hardware shall I buy? |
| 14:04.00 | djin | thanks (and Cybertoy) |
| 14:04.13 | memic | how to configure asterisk that i can dialout via my siphones over isdn |
| 14:04.31 | djin | memic, question is much shorter then answer |
| 14:04.51 | memic | lol |
| 14:05.00 | memic | %) |
| 14:05.08 | memic | i have both running |
| 14:05.18 | Maarken | heh. it's questions like that make me glad I trunk via IAX. :D |
| 14:05.18 | memic | zaptel & sip phone |
| 14:05.19 | djin | ah, that makes things easier :) |
| 14:05.30 | *** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net) |
| 14:05.40 | *** join/#asterisk IPD (~asteriski@vpn.expetel.com) |
| 14:05.42 | memic | i can call from siphone to siphone |
| 14:05.43 | focks | how do I unlock a Cisco 7960? |
| 14:06.20 | Akelavlk | Hello, I want have one-two PSTN and 10-20 analog phones, what hardware shall I buy? |
| 14:06.54 | ChArLeS___ | Akelavlk: pay for consultation |
| 14:07.00 | focks | what he said |
| 14:07.18 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 14:07.28 | Cybertoy | ake, why 10-20 analog phones? it'd probably be easier to deploy IP phones ... |
| 14:07.58 | Akelavlk | Cybertoy, Because companyes has analog phones. |
| 14:08.03 | SpaceBass | focks depending on the firmware its **# or cisco |
| 14:08.20 | SpaceBass | ebay the analog phones, buy IP ones... |
| 14:08.29 | focks | SpaceBass, if I gave you the firmware would you know? |
| 14:08.42 | memic | eh djin how? ;) |
| 14:08.45 | djin | can G729 licenses be transferred to a new server (perhaps once) |
| 14:08.47 | Cybertoy | I would agree with space... you'd have to get ATA's for the analog phones... and then you might as well just buy IP phones. |
| 14:08.57 | focks | Application Load: P003AM30 Boot Load: PC030300 |
| 14:09.15 | SpaceBass | focks sounds like the early early... its **# |
| 14:09.16 | focks | that's SCCP right? |
| 14:09.35 | SpaceBass | yeah |
| 14:09.37 | focks | SpaceBass, hmm, that doesn't seem to do anything. must it be plugged into a network to work? |
| 14:10.19 | Beirdo | Cybertank: I would disagree :) |
| 14:10.23 | Akelavlk | I think, VoIP is good, but if company has already builded phone network, haven't sence change it to VoIP. |
| 14:10.27 | Beirdo | get a 24-port FXS channel bank |
| 14:10.30 | ManxPower | djin, I THINK your license key will work up to three times |
| 14:10.34 | Cybertoy | akelavlk, unless you have those 10-20 analog phones connected to an analog PBX .. in that case you can only connect the pbx to asterisk |
| 14:10.41 | *** part/#asterisk Darwin35 (~richard@ip70-179-215-116.dl.dl.cox.net) |
| 14:10.46 | Akelavlk | Beirdo, Where can I buy it? |
| 14:10.49 | Beirdo | ebay |
| 14:10.53 | Beirdo | where else? :) |
| 14:10.54 | IPD | adit 600 makes a 48 port channel bank |
| 14:10.59 | ManxPower | djin, the G7829 license is tied to the MAC address in the computer. |
| 14:11.00 | IPD | get the mgcp card |
| 14:11.10 | Beirdo | you'd need a T1 card for the asterisk box though if I remember correctly |
| 14:11.21 | memic | ManxPower change your mac ;) |
| 14:11.23 | Akelavlk | eBay is not working in my country.. |
| 14:11.24 | focks | SpaceBass, do I have to be in any certain menu when I issue the **#? |
| 14:11.28 | djin | ManxPower, yeah and I thought I read something about limited transferring. |
| 14:11.52 | djin | Well, donating 50 to Digium isn't that bad ;) |
| 14:11.58 | Cybertoy | beirdo, ok .. but for that money you can still get ip phones, no? |
| 14:12.11 | Beirdo | it would be close |
| 14:12.16 | Akelavlk | Is there some card what has PSTN and also T1/E1? |
| 14:12.19 | Beirdo | 20 IP phones is a lot of money |
| 14:12.39 | djin | isn't E1/T1 PSTN? |
| 14:12.39 | Akelavlk | Sure IP phones are too expensive.. |
| 14:12.40 | focks | $4000 for 20 Polycom 501s |
| 14:12.41 | opus_ | about $4400 |
| 14:12.58 | SpaceBass | focks I'm not 100% sure since I rarely encounter that... it might even be *## |
| 14:12.58 | opus_ | with advance replacement:) |
| 14:12.58 | Akelavlk | djin, no it's not same.. |
| 14:13.12 | Beirdo | and a channel bank is about $600US, plus a T1/E1 card for around the same... |
| 14:13.16 | djin | Are you sure you not mistaken with POTS? |
| 14:13.18 | SpaceBass | focks I think if you are in the main menu and do **# you'll see the lock open |
| 14:13.19 | IPD | im setting a adit600 channel bank up now.. lan connection to asterisk - asterisk communicates to adit via mgcp - adit channel bank converts it to up to 40 POTS lines FXS fxo OR t1 |
| 14:14.22 | Akelavlk | So what hardware should I buy? |
| 14:15.05 | Beirdo | I think adtran is one of the most respected channel banks. I dunno, I only have 3 phones hooked up |
| 14:15.10 | Beirdo | I use ATAs |
| 14:15.12 | SpaceBass | i'm still working on email to fax with *... anyone know how I can take an attachment out of my mailbox /var/spool/mail/asterisk and copy it to a directory? |
| 14:15.25 | Akelavlk | What I need exactly is some card from digium what has 20 FXS an 2 FXO |
| 14:15.40 | Beirdo | you won't find one |
| 14:15.44 | djin | 20 FXS is a channelbank |
| 14:15.52 | juliette | <PROTECTED> |
| 14:16.27 | Akelavlk | Aha ok.. |
| 14:17.30 | memic | how to configure a hcfi card to listen to one special msn? |
| 14:17.39 | memic | is this done via extensions in asterisk ? |
| 14:17.45 | memic | or in zaptel.conf |
| 14:17.49 | *** part/#asterisk Akelavlk (~jansun@82.119.239.141) |
| 14:17.57 | memic | ore zapata.conf |
| 14:18.02 | SuPrSluG | i keep getting this odd message on one of my boxes. |
| 14:18.07 | SuPrSluG | <PROTECTED> |
| 14:18.09 | SuPrSluG | <PROTECTED> |
| 14:18.20 | forkqueue | SuPrSluG: Do you have monit installed? |
| 14:18.28 | djin | Akelavlk, you could google for ADTRAN TSU-600 |
| 14:18.35 | SuPrSluG | doesn't seem to affect anything. just annoying |
| 14:18.43 | SuPrSluG | what's that? |
| 14:18.50 | IPD | does this look familar to anyone? chan_mgcp.c:2281 handle_response: Transaction 2 timed out |
| 14:18.57 | forkqueue | SuPrSluG: Program that monitors daemons and restarts if necessary |
| 14:18.58 | funxion | SuPrSluG I think its just your level of chatiness |
| 14:19.25 | forkqueue | SuPrSluG: Or anything else that connects to the * manager? |
| 14:19.27 | SuPrSluG | i'm running safe_asterisk. would that use the program? |
| 14:19.47 | *** join/#asterisk trickyrick77 (~rsegrest@207.111.174.1) |
| 14:20.31 | SuPrSluG | funxion:verbose? |
| 14:20.35 | funxion | yes |
| 14:20.36 | *** join/#asterisk aiks (~aiks@159.148.227.104) |
| 14:21.00 | aiks | hi, once again |
| 14:21.01 | SuPrSluG | i hav it a 4. lemme drop it to 3 |
| 14:21.14 | lilalinux | As I don't have real hw sip phones, can I use a sw client on the same machine as the asterisk is running on? |
| 14:21.59 | thal | lilalinux: sure |
| 14:22.26 | aiks | another stupid question: i have around 40 analog phones, which i would like to connect to asterisk pbx; so what i basicly need is at least double span e1 card and a multiplexer supporting FXO -> E1 ? |
| 14:22.36 | lilalinux | I have a sipgate account, where do I tell asterisk that it should use that? |
| 14:23.16 | SuPrSluG | forkqueue:manager isn't running |
| 14:24.12 | thal | lilalinux: read the documentation. www.voip-forum.org, the asterisk handbook |
| 14:24.26 | thal | lilalinux short: in sip.conf with register => |
| 14:25.03 | IPD | question: has anyone seen this message? chan_mgcp.c:2281 handle_response: Transaction 2 timed out |
| 14:25.27 | IPD | ive done all I can to not come ask this... but HELP! |
| 14:25.47 | *** join/#asterisk DeeJayTwo (~deejay2@office.abi.ca) |
| 14:26.19 | lilalinux | thx |
| 14:28.32 | *** join/#asterisk tengulre (~tengulre@219.144.170.174) |
| 14:28.34 | *** join/#asterisk krisguy (~krisguy@h216-170-039-057.adsl.navix.net) |
| 14:29.04 | *** join/#asterisk tclark (~TC@S0106000c413a1c61.gv.shawcable.net) |
| 14:29.50 | *** part/#asterisk pointer (pointer@aj.catt.com) |
| 14:29.58 | memic | wtf |
| 14:30.03 | memic | how does that work? |
| 14:30.07 | *** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
| 14:30.17 | memic | how can i bind asterisk to a msn? |
| 14:30.31 | ManxPower | ~docs |
| 14:30.31 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 14:31.08 | *** join/#asterisk loud (~roots@cypher.punk.net) |
| 14:31.47 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfl3e.dialup.mindspring.com) |
| 14:33.21 | SuPrSluG | memic: what's an msn? |
| 14:34.07 | ManxPower | SuPrSluG, ISDN BRI thing |
| 14:34.10 | memic | just a isdn nummer |
| 14:34.51 | *** join/#asterisk cypromis (~michael@83.149.70.59) |
| 14:35.04 | SuPrSluG | have'nt had to deal with isdn yet |
| 14:35.48 | coppice | everyone live in an ISDN country |
| 14:35.51 | essobi | What was the name of that asterisk benchmarking program that controlled another * box from the management port? |
| 14:36.53 | Cybertoy | yeah .. I moved from Switzerland to USA ... here it seems like ISDN is only something big corporations can afford... |
| 14:37.19 | Cybertoy | then again I only have a dsl line at home now and route all calls with * through VoIP ... |
| 14:37.28 | Nugget | where in switzerland? |
| 14:37.39 | Cybertoy | was in zurich |
| 14:37.48 | Nugget | cool. I lived in basel on and off for a long time |
| 14:37.49 | *** join/#asterisk Darwin35 (~richard@ip70-179-215-116.dl.dl.cox.net) |
| 14:38.01 | Cybertoy | I was born in Basel ... :) |
| 14:38.08 | Nugget | great place, basel. I love it there |
| 14:40.25 | riksta | how can we push for a kind of important feature to make it into 1.2 ?? ( http://bugs.digium.com/view.php?id=4766 ) |
| 14:41.01 | ManxPower | riksta, Um, too late. 1.2 is in Feature Freeze |
| 14:41.18 | riksta | that really sucks |
| 14:41.29 | riksta | this is pretty critical |
| 14:41.37 | ManxPower | Ah! It was submitted before the feature freeze, maybe it will be included. |
| 14:41.39 | riksta | (for the stuff i wanna do anyway :) |
| 14:41.50 | riksta | well, i hope it will then |
| 14:42.18 | riksta | the "maybe" is what concerns me, who can I ask? |
| 14:42.24 | riksta | :) |
| 14:42.29 | essobi | Riksta You can always patch. :P |
| 14:42.39 | essobi | Mmm. |
| 14:42.40 | riksta | no |
| 14:42.54 | essobi | I don't suppose SIPP builds and tears down RTP ehh? |
| 14:42.57 | riksta | then everyone who uses the app will have to patch thats just stupid |
| 14:43.09 | essobi | Umm. |
| 14:43.13 | essobi | Then get over it. :) |
| 14:43.27 | riksta | there's no reason for it not to be included in 1.2 |
| 14:43.30 | essobi | And stupid for whom? You? Them? Developers? |
| 14:43.46 | riksta | all? |
| 14:44.00 | essobi | you != all. ;) |
| 14:44.17 | essobi | Like me.. I frankly could care less.. if you did have an app I wanted to use.. I'd patch it then. |
| 14:44.17 | *** join/#asterisk anthm (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 14:44.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:44.18 | riksta | you don't even understand |
| 14:44.18 | Rienzilla | hmm |
| 14:44.35 | Cybertoy | I have a problem with ChanSpy ... it's garbled and on asterisk I get ast_queue_spy_frame: Too Many frames queued |
| 14:44.35 | Cybertoy | at once, flushing cache. |
| 14:44.45 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 14:44.59 | essobi | riksta, okay.. what.. you got a .net application that allows someone to log in and out of a phone queue? |
| 14:45.06 | essobi | Or is it php? |
| 14:45.11 | essobi | Maybe perl? |
| 14:45.12 | riksta | no |
| 14:45.19 | blitzrage | gotta love power outages |
| 14:45.26 | essobi | blitzrage :) |
| 14:45.32 | Rienzilla | bweh |
| 14:45.44 | blitzrage | essobi: zup :) |
| 14:45.49 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 14:45.50 | SkramX | hehe |
| 14:45.52 | SkramX | that wasnt fun |
| 14:45.55 | SkramX | fucking, RoadRunne |
| 14:45.57 | SkramX | r |
| 14:46.03 | essobi | blitzrage All power corrupts. |
| 14:46.11 | riksta | hehe |
| 14:46.12 | essobi | blitzrage But we need it to run *. :) |
| 14:46.16 | blitzrage | SkramX: I hear that! (and I'm not even in the states :)) |
| 14:46.31 | blitzrage | essobi: you need a bicycle and some batteries :) |
| 14:46.44 | essobi | You'd have to be fast to fuck a roadrunner. |
| 14:46.54 | mut | yes |
| 14:46.55 | SkramX | haha, you know what I mean.. |
| 14:47.03 | mut | *bow chica bow chica bow* |
| 14:47.06 | essobi | Haha.. |
| 14:48.09 | *** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1) |
| 14:48.40 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
| 14:53.05 | blitzrage | Cresl1n: morning! |
| 14:53.20 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 14:53.56 | RoyK | BOOOOOORING |
| 14:54.20 | Cresl1n | bliztrage!!!! |
| 14:54.21 | Cresl1n | hey |
| 14:54.59 | SkramX | eeeeeee! |
| 14:55.43 | puzzled | has addmailbox been removed from today's HEAD? can't find it anymore |
| 14:55.59 | puzzled | and morning all off course |
| 14:56.16 | *** join/#asterisk jmacz (~jmacz@63.245.86.173) |
| 14:56.58 | Cresl1n | I thought you didn't have to use addmailbox anymore? |
| 14:57.10 | puzzled | that would be a good reason to remove it :) |
| 14:57.13 | Cresl1n | that's the userland app for making mailboxes right? |
| 14:57.18 | puzzled | yup |
| 14:57.21 | Cresl1n | yeah |
| 14:57.27 | Cresl1n | IIRC, you haven't had to use that for a long time |
| 14:57.40 | Corydon-w | You haven't had to use it for over two years |
| 14:57.53 | puzzled | I just have it in my rpm and updated this morning, rebuild and got an error cause it was missing |
| 14:58.04 | puzzled | time to fix up the specfile |
| 15:01.44 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
| 15:02.54 | *** part/#asterisk Darwin35 (~richard@ip70-179-215-116.dl.dl.cox.net) |
| 15:04.18 | *** join/#asterisk Hmmhesays (~Neg@24-117-213-113.cpe.cableone.net) |
| 15:06.48 | *** join/#asterisk leandro_it (~leandro@lan.texnet.it) |
| 15:07.15 | leandro_it | hello |
| 15:07.37 | leandro_it | anyone try the chan_bluetooth module? I am looking for any help |
| 15:08.34 | *** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218) |
| 15:09.29 | Cresl1n | -EAGAIN |
| 15:10.33 | blitzrage | wow, apple is finally offering a multi button mouse |
| 15:11.08 | *** join/#asterisk mjman (~mikem@205.158.42.66.ptr.us.xo.net) |
| 15:11.10 | mjman | Hello |
| 15:11.27 | bublbobl | blitzrage> I enjoyed apple b4 makin no difference whether you are left or right handed :-) |
| 15:11.34 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 15:11.34 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 15:11.35 | RomDump | can anyone get to http://asteriskathome.sourceforge.net? |
| 15:11.38 | blitzrage | bublbobl: I hate apple :) |
| 15:12.10 | bublbobl | RomDump> down :-( |
| 15:12.23 | RomDump | I need the pdf handbook :( |
| 15:12.43 | *** join/#asterisk Darwin35 (~richard@ip70-179-215-116.dl.dl.cox.net) |
| 15:12.49 | bublbobl | RomDump> have an email adress ? |
| 15:13.41 | mjman | I am having an issue with dropped calls. This is the log output when the call is dropped: http://pastebin.com/328427 It appears that asterisk is (perhaps mistakenly) receiving a hangup signal from the far end of the call. My question is this: Can I configure asterisk to NOT hangup the call when it receives a 'hangup' from the far end? Essentially, I want asterisk to always wait for the near-end to hang up before ending the call. Any help |
| 15:13.41 | mjman | <PROTECTED> |
| 15:14.33 | *** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
| 15:14.37 | TripleFFF2sdf | hey all |
| 15:15.05 | bublbobl | bybye all |
| 15:15.09 | TripleFFF2sdf | wondering if any have or kow about a pluging for eiter naggios or standalone that can monitor asterisk... i mean like place a call etc |
| 15:16.03 | ManxPower | mjman, sounds like you have busydetect=yes or callprogress=yes |
| 15:16.26 | ManxPower | They are both aliases for the option randomlydisconnectcalls=yes |
| 15:16.43 | RomDump | Anyone know where I can get an x100p clone card in Toronto, Ontario, Canada? |
| 15:16.50 | *** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca) |
| 15:17.00 | Qwell | RomDump: I'd look in the trashcan...you're bound to find some there. |
| 15:17.02 | ManxPower | RomDump, ebay |
| 15:17.06 | mjman | ManxPower, lol i'll check |
| 15:17.15 | mjman | is it in zapata.conf? |
| 15:17.21 | ManxPower | yes |
| 15:17.30 | RomDump | ManPower: I really just want to go pick it up. |
| 15:17.36 | ManxPower | I assume your calls are going out a Zap port. |
| 15:17.39 | RomDump | Are they really junk? |
| 15:17.44 | Qwell | RomDump: yes |
| 15:17.45 | ManxPower | RomDump, good luck with that. |
| 15:17.53 | ManxPower | RomDump, no they are not "junk" |
| 15:17.56 | RomDump | What do you recommend then? |
| 15:18.11 | RomDump | With out breaking my arm |
| 15:18.25 | ManxPower | RomDump, I recommend a Te110P and a channel bank w/FXO ports, but that's somewhat more expensive. |
| 15:18.28 | Qwell | Why don't people ever understand that telephony ain't cheap? |
| 15:18.37 | RomDump | Let me look it up |
| 15:18.41 | Cresl1n | yeah |
| 15:18.50 | Qwell | RomDump: the tdm400p is a cheaper solution |
| 15:19.06 | Cresl1n | Qwell: I get a kick out of it everytime tha somebody on the users lists balks at the price of a T1 card |
| 15:19.08 | Qwell | That'll give you up to 4 FXO ports |
| 15:19.17 | Maarken | even good digium cards are cheaper than a standard digital PBX system though. |
| 15:19.18 | RomDump | You mean an actual Digium card |
| 15:19.24 | Qwell | Maarken: exactly |
| 15:19.29 | ManxPower | I've not been happy with the TDM400P |
| 15:19.29 | Qwell | RomDump: uhh, yeah |
| 15:19.39 | ManxPower | And yes, a TE110P is CHEAP. |
| 15:19.43 | Beirdo | heh |
| 15:19.48 | Qwell | ManxPower: better then the x100p, and cheaper then a TE110P (for 1 port) |
| 15:20.03 | Beirdo | not if you compare it to the cost of the parts on it, it sure isn't |
| 15:20.07 | ManxPower | On a Nortel you can expect to pay $4,000 just for the T-1 card plus the cost of the software to enable PRI on the PBX |
| 15:20.23 | Qwell | ManxPower: You have to pay for software to get a PRI? Thats silly |
| 15:20.25 | ManxPower | Qwell, Actually, my X100P has been more reliable than my TDM400P |
| 15:20.26 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
| 15:20.29 | Qwell | but, I guess I'm spoiled |
| 15:20.37 | RomDump | What about the tigerjet cards? |
| 15:20.57 | ManxPower | RomDump, get a X100P clone, but don't pay more than $20 for it. |
| 15:21.10 | Maarken | ebay has them for about $15+S&H |
| 15:21.24 | Qwell | I paid 12, with shipping |
| 15:22.04 | *** join/#asterisk mago2-cn (~maxglucks@200.109.166.83) |
| 15:22.05 | RomDump | I am trying voipdepot.ca but they are out of stock |
| 15:22.16 | RomDump | I really dislike ebay |
| 15:22.37 | RomDump | $20 US? |
| 15:22.38 | Maarken | there are couple storefronts on ebay, you just hit buy it now and go |
| 15:24.37 | SkramX | Is there a way to like look up and see what company/telco owns a certain TOLL FREE #? |
| 15:24.59 | Cresl1n | yeah, get SS7 access |
| 15:25.00 | Cresl1n | :-) |
| 15:25.03 | Cybertoy | search on google? |
| 15:25.12 | Cresl1n | do the database lookup |
| 15:25.35 | Qwell | Anybody happen to know how I can get ahold of oej? |
| 15:25.58 | *** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
| 15:25.59 | SkramX | Creslin, who has that kind of access |
| 15:26.09 | Qwell | SkramX: your telco |
| 15:26.13 | Cresl1n | skramx: your telco :-) |
| 15:26.13 | SkramX | uh |
| 15:26.15 | Cresl1n | jinx! |
| 15:26.23 | Qwell | Cresl1n: beat you by three seconds. :p |
| 15:26.24 | mjman | ManxPower, callprogress and busydetect are both off |
| 15:26.26 | RomDump | My PAP-NA cost me $150CDN ever since normal distribution stopped. Should ahve bought a sipura. |
| 15:26.49 | TripleFFF2sdf | but romdump cheap clones have false hangups.. + get non cid detections in 50% of the cases |
| 15:26.54 | mjman | this is going out over a PRI T1 by the way |
| 15:26.57 | TripleFFF2sdf | so uslsess with ivr if u need cid |
| 15:27.12 | ManxPower | TripleFFF2sdf, I never had those problems |
| 15:27.18 | TripleFFF2sdf | on clones ? |
| 15:27.20 | TripleFFF2sdf | lol ok |
| 15:27.22 | TripleFFF2sdf | just me then |
| 15:27.30 | TripleFFF2sdf | btw i can have a carrier push SS& |
| 15:27.32 | TripleFFF2sdf | ss7 |
| 15:27.34 | RomDump | Yeah but I really don't want to fork over that much money now for an FXo |
| 15:27.37 | TripleFFF2sdf | ihere or h323 |
| 15:27.47 | TripleFFF2sdf | what is better.. i prefered SIP but hey they cant it seem |
| 15:28.01 | *** part/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
| 15:28.26 | *** part/#asterisk xylome (~asterisk@hg-msq-hol.levigo.de) |
| 15:28.27 | TripleFFF2sdf | and my other question.. any monitoring program for asterisk made yet ? |
| 15:28.36 | TripleFFF2sdf | like that places a call or somthing every xx seconds to test |
| 15:28.54 | *** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
| 15:29.28 | *** join/#asterisk _ranger_ (~bgmilne@vesuvius.obsidian.co.za) |
| 15:30.24 | SkramX | anyone ever used internet, via their Sprint PCS phone? We have a Treo 650... |
| 15:30.57 | TripleFFF2sdf | no |
| 15:31.01 | SkramX | damn |
| 15:31.18 | RomDump | Maybe I should just build my own FXO since the X100p clone cards suck. |
| 15:31.46 | Qwell | RomDump: Are you intimately familiar with how one works? If not, I'd not suggest it... |
| 15:32.37 | _ranger_ | anyone know what to use in South Africa ? |
| 15:32.45 | TripleFFF2sdf | water |
| 15:32.45 | RomDump | There is reference design already up for the FXO, (Check tigerjet website) |
| 15:32.46 | TripleFFF2sdf | ;) |
| 15:32.50 | _ranger_ | (ie in loadzone in zaptel.conf) |
| 15:32.52 | mishehu | SkramX: I've seen some bluetooth dialing docs on the web before. |
| 15:33.02 | SkramX | yueya |
| 15:33.05 | SkramX | i dont have the cd tho |
| 15:33.23 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 15:34.13 | *** join/#asterisk Cherebrum (jgarland@72.36.136.226) |
| 15:34.25 | Mimmus | hi, I'm having problems with a recently buyed TE410P Digium card |
| 15:34.39 | Cherebrum | Someone here is running an evil twin AP here at ClueCon |
| 15:34.43 | Mimmus | no interrupts in cat /proc/interrupts |
| 15:34.51 | mishehu | Mimmus: did you try calling digium? |
| 15:35.33 | Cherebrum | Thier MAC address is 00-09-5B-53-0C-5A |
| 15:35.33 | Mimmus | mishehu: yes, but I'd like to solve by myself in the meanwhile... |
| 15:35.59 | mishehu | Mimmus: it might be in the wrong type of pci slot. |
| 15:36.17 | Cresl1n | Mimmus: what drivers are you using? |
| 15:36.21 | Mimmus | mishehu: no, no: slot is a PCI-X and it is right |
| 15:36.36 | Mimmus | Cresl1n: I'm using wct4xxp |
| 15:36.46 | Cresl1n | Mimmus: no, what version? |
| 15:36.54 | mishehu | Mimmus: is that card a pci-x card? |
| 15:36.59 | TripleFFF2sdf | any monitoring program for asterisk made yet ? |
| 15:37.00 | mishehu | or just a pci one |
| 15:37.30 | Mimmus | mishehu: yes, digium support confirmed this |
| 15:37.54 | Cresl1n | TripleFFF2sdf: probably not if nobody has answered yet |
| 15:38.00 | TripleFFF2sdf | k |
| 15:38.03 | TripleFFF2sdf | how about h323 |
| 15:38.06 | mishehu | Mimmus: it's a long pci slot on the card? I don't have a te4xx series, I only have a te1xx series, and it's not a pci-x card. |
| 15:38.21 | TripleFFF2sdf | is that hard to implement on * and is it better worst then SIP |
| 15:38.29 | mishehu | I've had problems with other digium cards in pci-x slots, I'd had to put them in standard pci slots. |
| 15:38.30 | mjman | I ask again. Does anyone know of a way to tell asterisk to keep a channel open (i.e. not hangup the call) until ONLY the internal phone ends the call??? If the external phone hangs up, I want asterisk to ignore it. Thanks. |
| 15:38.37 | Mimmus | Cresl1n: tried also with latest cvs |
| 15:39.15 | Cresl1n | mjman: use analog signalling :-) |
| 15:39.19 | Mimmus | mishehu: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P |
| 15:39.21 | Cresl1n | mjman: that |
| 15:39.31 | Cresl1n | mjman: that'll do it for you |
| 15:39.32 | Cresl1n | :-) |
| 15:39.35 | Mimmus | Cresl1n: uh? |
| 15:39.49 | Cybertoy | mjman, what are you trying to achieve? If I hangup on someone my switch disconnects and there's no way for the other person to keep the line open. |
| 15:40.20 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 15:40.40 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
| 15:40.44 | MikeJ[Laptop] | simulcast from cluecon!!! |
| 15:40.45 | MikeJ[Laptop] | mark is speaking |
| 15:40.47 | mjman | Hm. I didn't know that that was impossible to do. I guess you're right though. The problem is that asterisk receives the signal that the external end has hungup, even though it hasn't, and the call is dropped |
| 15:40.52 | Cresl1n | how do we listen? |
| 15:41.15 | MikeJ[Laptop] | no, I type |
| 15:41.20 | Cresl1n | ah, ok |
| 15:41.22 | Cresl1n | where? |
| 15:41.31 | mishehu | Mimmus: *shrug* |
| 15:41.32 | MikeJ[Laptop] | he says some people think dual license is evil |
| 15:41.36 | mishehu | can't help you unfortunately |
| 15:41.42 | RomDump | The Wildcard TE110P costs $595.. WHo was the brainiact who suggested that |
| 15:41.45 | mjman | Cresl1n, what is analog signalling |
| 15:41.59 | MikeJ[Laptop] | the roadmap is driven by the developers.... |
| 15:42.03 | MikeJ[Laptop] | and other stuff |
| 15:42.06 | MikeJ[Laptop] | ;) |
| 15:42.30 | lilalinux | how do i quit asterisk fro mthe cli? |
| 15:42.39 | *** join/#asterisk Exstatica (exstatica@65.119.22.200) |
| 15:42.40 | Lee__ | the brainiact with $595 |
| 15:42.43 | Cybertoy | mjman, dunno what your problem is without output from the asterisk console... sorry ... |
| 15:42.47 | Cresl1n | mjman: use FXOs w/loop start, w/o disconnect supervision |
| 15:42.48 | Cresl1n | :-) |
| 15:42.49 | Qwell | lilalinux: quit the CLI, or stop it? |
| 15:43.06 | RomDump | I was looking for a cheap fxo... |
| 15:43.06 | TripleFFF2sdf | anyone tried the new echo cancelers one ? |
| 15:43.08 | ChArLeS___ | lilalinux: pull the power cord |
| 15:43.13 | ChArLeS___ | lilalinux: pull it out |
| 15:43.14 | TripleFFF2sdf | are they worth extra 600$ on quad's |
| 15:43.24 | lilalinux | of course i could kill it, but I expected a command like "quit" :) |
| 15:43.29 | Saaib | morning all |
| 15:43.33 | Qwell | lilalinux: There is a quit command. |
| 15:43.35 | TripleFFF2sdf | try exit |
| 15:43.39 | lilalinux | too late |
| 15:43.39 | Cresl1n | TripleFFF2sdf: if you have echo it might be :-) |
| 15:43.43 | lilalinux | but i did before |
| 15:43.43 | Qwell | lilalinux: There is also stop now, if you want to stop it |
| 15:44.02 | TripleFFF2sdf | i mean * has Ecancel built in..but software..is hardware better? |
| 15:44.19 | mjman | Cresl1n, here is the output from asterisk when the call is dropped: http://pastebin.com/328427 We are using a PRI card, so no FXO's |
| 15:44.30 | RomDump | I can only afford the asterisk sticker from Digium :) |
| 15:44.35 | Cresl1n | mjman: you're out of luck then |
| 15:44.42 | Cresl1n | mjman: no cookie for you |
| 15:44.42 | mjman | hm |
| 15:44.43 | TripleFFF2sdf | lol |
| 15:44.45 | mjman | =( |
| 15:44.50 | TripleFFF2sdf | the sticker is what is work 600 to 2.4k |
| 15:44.55 | TripleFFF2sdf | the board is like 5$ |
| 15:45.00 | TripleFFF2sdf | pay for the name my friend |
| 15:45.01 | mog_home | ? |
| 15:45.16 | TripleFFF2sdf | and qual |
| 15:45.25 | Qwell | I wonder if stickers and such come with support too |
| 15:45.38 | RomDump | The sticker cost $10 Plus shipping |
| 15:45.39 | Qwell | "uhh, yeah...hi...I can't seem the get the backing off of this sticker." |
| 15:45.51 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 15:45.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:45.53 | TripleFFF2sdf | yep 100$ per 10stickers for 1 year |
| 15:46.23 | TripleFFF2sdf | hehe |
| 15:46.46 | leandro_it | anyone use the chan_bluetooth channel? |
| 15:47.05 | TripleFFF2sdf | no |
| 15:47.07 | TripleFFF2sdf | hmm |
| 15:47.16 | TripleFFF2sdf | ok so AGI is my way i need to go |
| 15:47.49 | TripleFFF2sdf | if call rings on destination i just get a return that is non error right ? so by making a call filei should be able to see if asterisk can dial out.. therefore if all is ok |
| 15:48.02 | RomDump | I guess the Asterisk sticker is TM so I can't make bootleag copies to sell :( |
| 15:48.11 | ChArLeS___ | hey |
| 15:48.15 | mog_home | lol |
| 15:48.17 | ChArLeS___ | does anybody got H323 to work ? |
| 15:48.23 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:48.27 | TripleFFF2sdf | charles i wish |
| 15:48.30 | ChArLeS___ | does anybody have openh323 (v1.13.5) ? |
| 15:48.31 | TripleFFF2sdf | whats needed for it ? |
| 15:49.09 | RomDump | TripleFF2sdf: What there is a one year license for the stickers? |
| 15:49.15 | TripleFFF2sdf | hehe |
| 15:49.21 | RomDump | 10 stickers for one year? |
| 15:49.45 | TripleFFF2sdf | afk |
| 15:49.53 | RomDump | and I thought Micro$oft was bad... |
| 15:50.20 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 15:51.46 | RomDump | "These professionally designed black-and-white stickers boast "Asterisk The Open Source PBX" and all proceeds go directly to further its development. |
| 15:51.52 | *** join/#asterisk dacleric (~dacleric@p54829CAB.dip0.t-ipconnect.de) |
| 15:52.36 | mog_home | yup |
| 15:53.07 | mago2-cn | hi, firmware files for cisco phones on sccp in astersik are the same found on cisco call manager's tftp directory? |
| 15:53.08 | RomDump | I am always up for a good cause |
| 15:53.54 | mog_home | it does help pay the rent |
| 15:56.30 | blitzrage | RomDump: those stickers are great for IDing your luggage at the airport |
| 15:56.51 | blitzrage | plus I've had numerous hot chicks ask me what the sticker meant - chicks dig Asterisk |
| 15:56.57 | mog_home | lol |
| 15:57.12 | *** join/#asterisk zoo (nobody@ip-54-16.travedsl.de) |
| 15:57.15 | MikeJ[Laptop] | mog! |
| 15:57.24 | mog_home | MikeJ |
| 15:57.27 | MikeJ[Laptop] | event system |
| 15:57.31 | MikeJ[Laptop] | radius |
| 15:57.35 | MikeJ[Laptop] | realtime |
| 15:57.39 | mog_home | all working? |
| 15:59.04 | funxion | a bit off topic but would anyone happen to kno how I can get a non prrivledged user to be able to restart a process on a remote machine using ssh user@ip /etc/init.d/program restart I've tried sudo it prompts for pass and this is for a script so it doesnt werk |
| 15:59.17 | blitzrage | OT: anyone know if I "remove" FireFox from my Add/Remove programs if it'll leave the data (bookmarks, etc...) and just remove the program? I'm getting crashes everytime I close firefox out (but works great when loaded) and its quick annoying. |
| 15:59.28 | RomDump | I can see it now at the airport... What country is that symbol for?... Ahh.. Daaa... Asterisk land.. |
| 15:59.44 | blitzrage | funxion: you need to setup keys on the machine so that you dont' authenticate with a password |
| 15:59.55 | funxion | blitzrage i think wheen you remove it it asks if you wan to remove the bookmarks etc just leave them at tat point |
| 15:59.56 | Delvar | SSH will always prompt for password, try using a keys.. to get round that |
| 16:00.04 | blitzrage | funxion: thanks! I will try that |
| 16:00.13 | funxion | blitzrage i did that it prompts for pass to goto sudo |
| 16:00.29 | forkqueue | funxion: NOPASSWD in sudoers |
| 16:00.37 | funxion | ahhh |
| 16:00.38 | funxion | thnx |
| 16:00.47 | funxion | I've done it before long time ago |
| 16:00.51 | funxion | just couldnt remember |
| 16:00.54 | funxion | thnx a lot |
| 16:00.55 | TripleFFF2sdf | or use keychain |
| 16:01.01 | TripleFFF2sdf | along with keys |
| 16:01.07 | funxion | ? |
| 16:01.15 | TripleFFF2sdf | google gentoo keychains |
| 16:01.19 | funxion | ok |
| 16:01.21 | TripleFFF2sdf | no's on chain |
| 16:01.21 | funxion | thnx |
| 16:01.38 | TripleFFF2sdf | u need passphrase oce per reboot..if u use one |
| 16:01.45 | TripleFFF2sdf | then all shells are loaded with one |
| 16:01.53 | TripleFFF2sdf | i can set it up @ 60$ per hour |
| 16:02.24 | mog_home | il do it at 59.95 |
| 16:02.29 | TripleFFF2sdf | 59.98 |
| 16:02.31 | TripleFFF2sdf | 59.94 |
| 16:02.31 | *** join/#asterisk riemensc (~riemensc@83-169-155-92-dynip.superkabel.de) |
| 16:02.37 | TripleFFF2sdf | this is a reverse ebay |
| 16:02.40 | mog_home | damn to cheap for my blood |
| 16:02.51 | TripleFFF2sdf | yeah the extra penny is all it took |
| 16:02.56 | ChArLeS___ | $90.00 Guaranteed |
| 16:03.05 | TripleFFF2sdf | well it takes about 30 minutes to compile setup and test both boxes |
| 16:03.10 | ChArLeS___ | $120.00 Guaranteed with insurance! |
| 16:03.15 | mog_home | lol |
| 16:03.18 | TripleFFF2sdf | 180. i make dishes |
| 16:03.25 | denon | $5k and you wont have to deal with these wankers |
| 16:03.30 | denon | :) |
| 16:03.30 | mog_home | heh |
| 16:03.32 | TripleFFF2sdf | now we talking |
| 16:03.42 | TripleFFF2sdf | 120k i buy you out and you never need to touch a keyboard |
| 16:03.46 | RomDump | Do Sipura devices use OpeenSSL or there own propriety Certificates for provisioning? |
| 16:03.58 | TripleFFF2sdf | RomDump weird q's this morning |
| 16:04.07 | blitzrage | woohoo! FireFox doesn't crash on close anymore! |
| 16:04.39 | TripleFFF2sdf | wondering if a fingerprint reader can server as API for asterisk sip phone password lol |
| 16:04.46 | TripleFFF2sdf | thats a RomDump qustio |
| 16:04.47 | TripleFFF2sdf | ;0 |
| 16:05.04 | RomDump | Just figuring out something I have been working on for some time... |
| 16:05.58 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 16:06.07 | TripleFFF2sdf | anyone know brian roy |
| 16:07.08 | TripleFFF2sdf | wich on 120,000 ? |
| 16:07.29 | RomDump | From $60 to $120,000.... |
| 16:07.41 | riemensc | everbudy use sipgate? |
| 16:07.55 | TripleFFF2sdf | i assume s/ever/any |
| 16:08.05 | RomDump | Talk about price gouging... |
| 16:08.38 | RomDump | So let me see I talk to you on the and it is $60/hr and you come to my bussiness and charge me $120,000... |
| 16:08.40 | TripleFFF2sdf | http://www.freedownloadscenter.com/Utilities/Password_Management_Utilities/Asterisk_Password_Recovery_Screenshot.html |
| 16:08.41 | RomDump | :) |
| 16:08.41 | *** part/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985234.sympatico.ca) |
| 16:08.50 | TripleFFF2sdf | time to get the names unmixed |
| 16:09.35 | Mimmus | any help with exact settings in zaptel.conf and zapata.conf for E1 PRI line in Italy? |
| 16:09.53 | *** join/#asterisk TheEmperor (TheEmperor@60.49.109.150) |
| 16:10.25 | leandro_it | Mimmus, wait that I recall the conf |
| 16:11.24 | TheEmperor | hi, what's a good gsm modem that's compatible with asterisk? |
| 16:11.41 | SkramX | hmm |
| 16:11.43 | TripleFFF2sdf | also looking for sms ;) |
| 16:12.07 | RomDump | My father just retired from Bell Canada after 30 Year's of service. I tried to sell him on the idea of of setting up Asterisk in third world countries. |
| 16:12.32 | TheEmperor | yeah, i would like something that can accept an sms and then call the mobile number that just smsed the system |
| 16:12.37 | TripleFFF2sdf | LIBISUP s |
| 16:12.56 | RomDump | I demo's voip to him and he wasn't impressed with the quality |
| 16:13.17 | leandro_it | RomDump, what codec do you use? |
| 16:13.29 | RomDump | G.729 wasn't any help in the presentation |
| 16:13.58 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 16:14.05 | RomDump | It sounded like I was underwater |
| 16:14.33 | TripleFFF2sdf | oh RomDump.. u need to get out the swimmiming pool first ! |
| 16:14.33 | leandro_it | g.729 and gsm are very good in our installation... |
| 16:14.53 | RomDump | I am trying to make it up by making an asterisk box and demoing it locally |
| 16:14.55 | TheEmperor | so can anyone recommend me a good gsm modem that can handle sms? :) |
| 16:15.17 | TripleFFF2sdf | Emperor ;) no.. but let me know if you find one |
| 16:15.22 | RomDump | g.729 over gprs right? |
| 16:15.26 | TripleFFF2sdf | the SMS function of asterisk does not work uin USA |
| 16:15.27 | TheEmperor | ok... |
| 16:15.35 | *** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no) |
| 16:15.43 | leandro_it | gprs? why do you want to use gprs? |
| 16:16.01 | leandro_it | gsm as codec |
| 16:16.42 | RomDump | I don't follow, You are VOIP over GSM? |
| 16:17.22 | TripleFFF2sdf | so anyone know best way to test if asterisk is all ok ? like a test caller agi or somethign ? |
| 16:17.45 | Maarken | TripleFFF2sdf: two softphones. :) |
| 16:17.53 | forkqueue | TripleFFF2sdf: You mean automated test like via Nagios? |
| 16:17.58 | TripleFFF2sdf | i mean ever 60 seconds automaticaly |
| 16:18.01 | Maarken | ah |
| 16:18.06 | TripleFFF2sdf | forkqueue yes |
| 16:18.17 | forkqueue | TripleFFF2sdf: You can use my check_sip plugin - http://www.bashton.com/content/nagiosplugins |
| 16:18.45 | TripleFFF2sdf | hmm |
| 16:18.52 | TripleFFF2sdf | what king of tests does it run ? |
| 16:19.10 | forkqueue | TripleFFF2sdf: Sends a SIP option packet |
| 16:19.15 | TripleFFF2sdf | ok |
| 16:19.19 | leandro_it | RomDump, gsm is one of the codec you can use, along with g.729 |
| 16:19.22 | TripleFFF2sdf | but doesnt actually make a call |
| 16:19.36 | TripleFFF2sdf | i need somethign that would place a call to test basically all sub routines.. |
| 16:19.50 | TripleFFF2sdf | like ODBC connection , trunks, my pri etc |
| 16:19.57 | TripleFFF2sdf | and make it call like 8004444444 |
| 16:20.00 | forkqueue | TripleFFF2sdf: The plugin is GPL, feel free to submit a patch :) |
| 16:20.01 | TripleFFF2sdf | or anynumber.. |
| 16:20.05 | TripleFFF2sdf | k |
| 16:20.10 | *** part/#asterisk FITA1 (~m_ahmed@202.5.145.50) |
| 16:20.50 | RomDump | leandro_it: I know GSM is a codec as well has G.729 you were talking about talking from G.729 to GSM? |
| 16:21.28 | RomDump | or takling about using G.729 on GPRS? |
| 16:21.41 | leandro_it | No, I am saying yo to try GSM codec if G.729 is bad for you |
| 16:21.56 | RomDump | I follow now... |
| 16:23.11 | RomDump | I onced worked for a major cellphone carrier in Canada. The president of the company would smash his phone if he would not get service and blame it on the phone. |
| 16:25.30 | TripleFFF2sdf | forkqueue wwhats a example cmd line. |
| 16:25.33 | TripleFFF2sdf | i dont get it |
| 16:26.06 | forkqueue | TripleFFF2sdf: more README |
| 16:26.26 | forkqueue | TripleFFF2sdf: Pricing for consultancy is on my site :) |
| 16:26.35 | TripleFFF2sdf | lol |
| 16:27.05 | TripleFFF2sdf | well user to test is USER in sip:USER@myproxy.com ? |
| 16:27.21 | Mimmus | msg leandro_it LBO=2, perche'? |
| 16:27.38 | TripleFFF2sdf | Invalid Extension |
| 16:27.50 | forkqueue | TripleFFF2sdf: check_sip -u sip:100@example.com |
| 16:28.03 | TripleFFF2sdf | oh not user |
| 16:28.07 | TripleFFF2sdf | it checks extension |
| 16:28.08 | TripleFFF2sdf | ok |
| 16:28.19 | forkqueue | TripleFFF2sdf: OK that should be clearer in the docs :) |
| 16:28.34 | TripleFFF2sdf | SIP 200 OK: 0.01 second response time |
| 16:28.35 | TripleFFF2sdf | you rock |
| 16:28.41 | forkqueue | TripleFFF2sdf: Thanks :) |
| 16:28.46 | forkqueue | TripleFFF2sdf: And with that, I'm off |
| 16:28.47 | forkqueue | l8rs |
| 16:28.55 | TripleFFF2sdf | j |
| 16:28.56 | TripleFFF2sdf | k |
| 16:30.15 | Rienzilla | hmm |
| 16:30.25 | Rienzilla | if I dial a number on a normal telephone |
| 16:30.34 | Rienzilla | when does asterisk decide that I'm finished dialing? |
| 16:30.50 | leandro_it | digittimeout |
| 16:31.40 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
| 16:32.33 | Rienzilla | before the tiemout |
| 16:32.43 | Rienzilla | how does it judge that a number is finished |
| 16:32.48 | Rienzilla | a match in extensions? |
| 16:33.13 | *** join/#asterisk junbug (junya@adsl-065-013-044-139.sip.mia.bellsouth.net) |
| 16:33.40 | leandro_it | no, first it timeout, then a match is searched. The first (not the best) is selected |
| 16:33.50 | *** join/#asterisk Bhaal (bhaal@bhaal.staff.freenode) |
| 16:33.57 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
| 16:33.59 | Rienzilla | so I _always_ wait for the timeout? |
| 16:34.37 | leandro_it | yes, this is true on Zap channel. On sip and iax2 channel this is not true |
| 16:34.46 | Rienzilla | It's an misdn channel |
| 16:35.01 | leandro_it | I don't know this kind of channel, sorry |
| 16:36.26 | Mimmus | +modalita' pass-trough (inizialmente) |
| 16:38.48 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 16:39.05 | *** join/#asterisk felipex (~dsfdsf@host250-98.pool8545.interbusiness.it) |
| 16:40.28 | *** join/#asterisk jtodd (~jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 16:41.39 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 16:46.12 | *** part/#asterisk zoo (nobody@ip-54-16.travedsl.de) |
| 16:47.48 | leandro_it | anyone use the chan_bluetooth? |
| 16:50.13 | *** join/#asterisk cfrank_ (~cfrank@bi01p1.co.us.ibm.com) |
| 16:52.47 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 16:56.41 | RomDump | Has anyone dealt with IAXtalk in buying hardware? |
| 16:56.55 | *** join/#asterisk YoYo (YoYo@dilbert.psknet.com) |
| 16:59.31 | *** join/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu) |
| 17:00.33 | *** join/#asterisk Techieb0y (~techieb0y@adsl-68-23-77-165.dsl.applwi.ameritech.net) |
| 17:06.22 | *** join/#asterisk shimi (~shimi@shimi.user) |
| 17:06.44 | RomDump | Doe Asterisk@home include ARI? |
| 17:07.36 | shimi | I read all over the google and couldn't understand anything. Is asterisk possible as a solution for running an office phone system with extensions, and if so, what do I need to buy and what do I need from the telco? |
| 17:08.20 | Cybertoy | how many extensions do you want to connect? you might be better off using SER (SIP express router) ... |
| 17:08.28 | RomDump | How much Lines infocoming and how much extentions? |
| 17:08.53 | shimi | I don't know exactly, let's say 20-30. |
| 17:09.12 | RomDump | Incoming lines? |
| 17:09.21 | shimi | extensions |
| 17:09.23 | Cybertoy | 20-30 extensions shouldn't be a problem. |
| 17:09.31 | *** join/#asterisk SplasPood (~jwb@dementia.paravolve.net) |
| 17:09.40 | shimi | the question is - how is this "hubbed" to the telco |
| 17:09.50 | junbug | who has an voicepulse IP i can ping .... |
| 17:09.57 | RomDump | Get 20-30 Sip phones then... |
| 17:10.18 | Cybertoy | shimi, you can plug it into a VoIP provider... |
| 17:10.22 | shimi | I was reading stuff about E1's etc. and that sounds weird because the phone company does only analog lines (so I think?) |
| 17:10.28 | *** join/#asterisk ayano (~erik@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 17:10.34 | shimi | but that's over the internet, and our internet links are really bad |
| 17:10.40 | ayano | Hi all |
| 17:11.09 | RomDump | shimi: Use the sip phones as extentions to asterisk |
| 17:11.23 | shimi | and how is asterisk connecting to the telco? |
| 17:11.30 | RomDump | Use a T1/E1 as a trunk to to telco |
| 17:11.39 | RomDump | Or FXO card |
| 17:11.44 | shimi | that's a data link |
| 17:11.47 | shimi | no? |
| 17:11.48 | RomDump | to use normal line |
| 17:12.04 | RomDump | Yeah it's a data link |
| 17:12.23 | shimi | is there a place where this thing is _explained_ ? because all what you're saying are things that I know, but not related to telephony. any good document would do... |
| 17:13.09 | mut | whats a good place to look for cool gadgets other than thinkgeek? |
| 17:13.15 | shimi | specificly, how data connection gets connected to the telco's carrier to make normal calls... |
| 17:14.48 | RomDump | shimi How much outgoing lines? |
| 17:15.24 | shimi | man, I don't know. I was just sent by the people giving the money to search for a good telephony solution to the office |
| 17:15.39 | shimi | I need to present the pro's and con's of whatever possible, and then they'll decide |
| 17:16.19 | shimi | if I knew how this worked, it would be easy. but I can't find any document explaining how these things work against regular phone lines (like the ones we got at home, for instance) |
| 17:17.19 | shimi | like: I dial 9 on the phone to get outside line. how is this linked to the telco? over what? if not on analog line, who does this conversion on the other side? what do I need to ask my phone company to do? |
| 17:17.51 | RomDump | If you have a small amount of incoming lines, (like 2-4) buying FXO card's and pluging normal phone lines are good |
| 17:18.13 | *** join/#asterisk pnviking (~pnlarsson@c83-248-2-153.bredband.comhem.se) |
| 17:18.26 | RomDump | FXO <=> Asterisk <=> SIP Phone |
| 17:18.33 | _T3_ | hi!, i need help from a developer, anybody? |
| 17:18.39 | shimi | if I use asterisk, I must use SIP phones? |
| 17:18.59 | RomDump | No you can use and ATA adapter also |
| 17:19.07 | ayano | _t3_: Most of them are eating lunch at cluecon right now |
| 17:19.26 | shimi | how much does an SIP phone cost? (rough estimate...) |
| 17:19.35 | ayano | shimi: no |
| 17:20.27 | ayano | shimi: dont go to cheap, you will regret it. I would go with like the IP300 which I think is about 150, but not sure |
| 17:20.55 | SkramX | damn it |
| 17:21.01 | SkramX | i really need to downgrade from cvs |
| 17:22.39 | wrmem | shimi: You can use your existing phones if you purchase a channel bank and a T1/E1 card for your server. Also, T1/E1 can be used for data, but also can be used for voice. Look up "ISDN PRI", "E&M", "trunk", etc. |
| 17:23.29 | shimi | wrmem, thanks, I already understood that, the part I didn't understand is what I need to _ask_ from the telco to install in my office |
| 17:23.40 | *** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl) |
| 17:24.06 | *** join/#asterisk loick (~loick@APuteaux-151-1-54-123.w82-120.abo.wanadoo.fr) |
| 17:24.17 | ayano | shimi: It is more what they "will" install. |
| 17:24.22 | shimi | but anyways, RomDump is explaining me in privmsg, so don't bother doing it twice :) |
| 17:25.10 | Maarken | generally the less you tell a telco the better. |
| 17:25.21 | Maarken | and use small words, spoken slowly. |
| 17:25.45 | wrmem | I'm in the US, so using the local words, an "ISDN PRI provisioned with xx DID numbers, configured for voice", "a channelized T1 with xx DS0 voice lines". You can also just use normal lines if you want to present a "you have reached company ABC, please enter the extension of the person..." |
| 17:25.50 | Cybertoy | skramX, I have CVS and it's working fine for me ... what features are not good for you? |
| 17:26.06 | SkramX | for some reason it keeps disconnected the server |
| 17:26.13 | SkramX | skram*CLI> |
| 17:26.13 | SkramX | Disconnected from Asterisk server |
| 17:26.19 | SkramX | does that every so often |
| 17:26.27 | Cybertoy | never here. |
| 17:26.31 | SkramX | fuck |
| 17:28.06 | Cybertoy | I have a prob with ChanSpy though ... |
| 17:28.15 | Cybertoy | it's garbled.... anyone have experience with that? |
| 17:29.36 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
| 17:31.52 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 17:31.58 | *** join/#asterisk mago2-cn (~maxglucks@200.109.166.83) |
| 17:32.13 | cursor | zzzzzz |
| 17:32.56 | mago2-cn | Hello, could anyone please let me know how to login into voicemail without having to dial the mailbox and using a diffeent context than deffault? |
| 17:33.09 | SkramX | what do you mean? |
| 17:33.14 | mago2-cn | I'm trying: exten => 123,1,VoicemailMain(${CALLERIDNUM}@different-context) |
| 17:33.22 | SkramX | ok |
| 17:33.44 | SkramX | so do 124,1,VoicemailAdmin.. blah blah blah |
| 17:33.45 | mago2-cn | But keeps me asking for mailbox. If I use default asks for password directly |
| 17:34.16 | mago2-cn | VoicemailAdmin will allow for context? |
| 17:35.28 | SkramX | I think |
| 17:35.42 | SkramX | Try it man. |
| 17:37.03 | cursor | VoicemailMain(${CALLERIDNUM}@different-context) will just ask for the password (not the mailbox) |
| 17:37.05 | cursor | if it exists |
| 17:37.28 | SkramX | OK/ |
| 17:38.50 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 17:40.25 | blitzrage | note: first 50 people to register for Astricon (http://www.astricon.net) get a free IAXy! |
| 17:40.35 | cursor | free :-) |
| 17:40.49 | blitzrage | :) |
| 17:41.20 | mut | what if we register then back out |
| 17:41.24 | mut | O_o |
| 17:41.33 | blitzrage | mut: won't be able to pick up your IAXy at the Digium booth then :) |
| 17:42.10 | mut | i could get someone else ot |
| 17:42.41 | cursor | far too many cons this year |
| 17:42.44 | blitzrage | mut: yah, but how are you going to pick it up if you're not registered - you wont' get into the conference :) |
| 17:42.51 | *** join/#asterisk astoria (~cluecon@67.107.50.40.ptr.us.xo.net) |
| 17:44.22 | inspired | hmm, even though I have notransfer=yes, I see this message: |
| 17:44.23 | inspired | <PROTECTED> |
| 17:44.35 | mago2-cn | Please correct if wrong because something is happening: if a dash is included in the sip name, it removes when passing mailbox to VoicemailMain right? |
| 17:44.36 | inspired | has anyone else seen this? |
| 17:44.47 | mago2-cn | Like different-context to differentcontext |
| 17:45.16 | mago2-cn | Talking abaout callerID |
| 17:45.25 | cursor | I doubt it |
| 17:45.29 | cursor | What do you see when you try it? |
| 17:46.22 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 17:46.22 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 17:46.54 | Darwin35 | well I hope the kids are going to have fun in Chicago |
| 17:47.02 | astoria | <-- in chicago |
| 17:47.20 | drumkilla | woooooooooo |
| 17:47.43 | cursor | Wow - I managed to find the con's registration page |
| 17:47.44 | mago2-cn | -- Executing VoiceMailMain("SIP/context-1001-787a", "context1001@context-phones") in new stack |
| 17:47.58 | *** join/#asterisk iswm (iswm@iswm.user) |
| 17:48.07 | Darwin35 | the other one bit the dust |
| 17:48.49 | mago2-cn | Let me put it with a different name: -- Executing VoiceMailMain("SIP/company-1001-787a", "company1001@company-phones") in new stack |
| 17:49.45 | mago2-cn | the second argument should be: company-1001@company-phones |
| 17:50.29 | *** join/#asterisk anthm (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 17:50.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:51.03 | cursor | Darwin35: what happened to the old system? |
| 17:52.16 | Darwin35 | wellI had it open cleaning it up anddusting. my cat desided he did not like my glass of icetea on the desk |
| 17:52.40 | *** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 17:52.53 | Darwin35 | <PROTECTED> |
| 17:53.06 | cursor | eeeew - good cat |
| 17:53.09 | cursor | iced tea is evil |
| 17:53.21 | Darwin35 | saw a few sparks and it died |
| 17:53.28 | cursor | When my tea goes cold, I make my way back to the kettle to make some more |
| 17:53.55 | Darwin35 | I love my green icetea |
| 17:54.32 | cursor | I prefer boiling hot tea |
| 17:54.38 | hardwire | blah |
| 17:55.07 | cursor | Did you damage any expensive cards? |
| 17:55.43 | Darwin35 | some days I do but green icetea is good as a blood thiner antioxiden |
| 17:56.19 | cursor | so is paint stripper |
| 17:57.03 | cursor | and a broadsword |
| 17:57.18 | cursor | that thins the blood quite effectively |
| 17:57.38 | Darwin35 | lol |
| 17:57.54 | cursor | reduces blood pressure too :-) |
| 17:58.28 | Darwin35 | I prefer to live |
| 17:58.36 | cursor | :-) |
| 17:59.41 | Darwin35 | man this xp 2600 is nice |
| 17:59.55 | Rienzilla | hmmm |
| 17:59.55 | *** join/#asterisk pbxbart (~pbxbart@proxy.prodyna.com) |
| 18:00.03 | cursor | AMD? |
| 18:00.06 | Darwin35 | and it did not cost me a dime |
| 18:00.21 | Darwin35 | who else makes a xp2600 |
| 18:00.31 | cursor | Microsoft :-) |
| 18:00.34 | cursor | 2600 is the version |
| 18:00.35 | Darwin35 | ? |
| 18:00.36 | cursor | haha |
| 18:00.38 | SkramX | ahhh |
| 18:00.44 | *** part/#asterisk pbxbart (~pbxbart@proxy.prodyna.com) |
| 18:00.46 | Darwin35 | hahha\ |
| 18:00.52 | _T3_ | thank ayano |
| 18:00.58 | Darwin35 | I dont do windows |
| 18:00.59 | _T3_ | sorry i had to run |
| 18:01.08 | Darwin35 | I do X |
| 18:01.10 | cursor | I only do windows when it's warm |
| 18:01.22 | cursor | or if someone farts |
| 18:01.47 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 18:02.14 | cursor | so then, what's the topic for today? |
| 18:02.16 | astoria | ha ha, look at all the people from dsl.chcgil.ameritech.net :) |
| 18:02.36 | cursor | with their laptops |
| 18:02.45 | *** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:03.06 | Darwin35 | cluecon if for those without a clue |
| 18:03.18 | Darwin35 | clues for 200 pls Alex |
| 18:03.21 | cursor | That's a lot of registrations then |
| 18:03.32 | outtolunc | typos for 1000 |
| 18:04.28 | cursor | Get a money-off cluepon |
| 18:04.54 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 18:05.09 | Darwin35 | it comes on a cd and when loaded it sends algarithiums to do specific functions ? |
| 18:05.21 | Darwin35 | sorry . |
| 18:05.24 | Darwin35 | not ? |
| 18:05.48 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 18:06.05 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 18:06.16 | harryvv | Darwin35 thats what I thought at first :) |
| 18:06.34 | Darwin35 | clock is running out |
| 18:06.43 | Darwin35 | pls ring ing with your answer |
| 18:07.17 | Darwin35 | eeet |
| 18:07.31 | blitzrage | f00d! |
| 18:07.38 | Darwin35 | the answer was WHat is a operating System? |
| 18:08.27 | outtolunc | i'd say there are other things that fit that also |
| 18:08.43 | cursor | Not all OSs come on a CD |
| 18:09.56 | *** join/#asterisk pfn (~pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 18:10.21 | jontow | does the iaxy support gsm? |
| 18:10.51 | drumkilla | jontow: ulaw and adpcm |
| 18:10.57 | *** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com) |
| 18:11.01 | Darwin35 | back in a bit break time |
| 18:13.01 | jontow | ouch, damn :) |
| 18:14.50 | cursor | Get a SIP device instead, if you need compression and don't really need IAX |
| 18:15.05 | cursor | Of course, if you need IAX then you need it |
| 18:15.31 | Maarken | sipura 1001 is cheaper anyway |
| 18:15.59 | jontow | this isn't a question of need right now.. its a question of we have the hardware and are beta testing a setup for a CLEC offering VoIP services in the upstate NY area.. |
| 18:16.11 | Maarken | ah |
| 18:16.38 | Maarken | well, on the upside with ulaw faxing might work sometimes. |
| 18:17.06 | jontow | i've got a sipura SPA-2000 at my place; a cow-orker is taking the IAXy home, and my boss has a softphone (iaxcomm) on his laptop |
| 18:17.40 | *** join/#asterisk Assid (~assid@203.115.64.59) |
| 18:17.56 | jontow | so far, iaxcomm is the best ;P |
| 18:18.15 | jontow | all using them on cablemodems, which is really the most widespread form of broadband in this area, by far |
| 18:19.06 | jontow | and GSM is what has been working best so far as well; without compromising voice quality |
| 18:19.10 | *** join/#asterisk |Vulture| (~V@c-66-177-92-4.hsd1.fl.comcast.net) |
| 18:20.33 | Maarken | yeah, GSM is pretty nice |
| 18:21.12 | Assid | i thought ilbc worked better than GSM |
| 18:21.13 | jontow | little vulnerable to popping/flaking but best so far |
| 18:21.20 | cursor | iLBC is nice |
| 18:21.23 | Maarken | I'll have to try iaxcomm. my users are mostly sjphone right now |
| 18:21.24 | |Vulture| | are they planning on releasing 1.2 durring astericon? |
| 18:21.28 | |Vulture| | cursor: yea i use ilbc |
| 18:21.42 | tzanger | iLBC gets nothing but complaints from my office |
| 18:21.50 | |Vulture| | really... |
| 18:21.52 | jontow | tzanger; how so? |
| 18:21.53 | tzanger | gsm and g729 are fine, but iLBC gets complaints :-( |
| 18:21.56 | tzanger | jontow: I wish I knew |
| 18:22.07 | jontow | voice quality or just overall bleh? :) |
| 18:22.12 | tzanger | they all (30+ people) claim that voice quality blows |
| 18:22.12 | |Vulture| | g729 sounds good as long as there is no music or making recordings |
| 18:22.12 | cursor | have you tried SpeeX? |
| 18:22.30 | tzanger | |Vulture|: I have no problem hearing music on g729... it's not PRETTY but it's not horrible either |
| 18:22.32 | |Vulture| | gsm I think sounds much worse than ilbc |
| 18:22.37 | tzanger | (thinking of on-hold music specifically) |
| 18:22.56 | jontow | i liked speex.. but it seems its less of an option in this case |
| 18:22.59 | |Vulture| | I just know I have to switch to ulaw when I make recordings remotly |
| 18:23.03 | tzanger | whee I get to write my first ever linux kernel driver |
| 18:23.18 | jontow | not so many marketed end-user devices (ATAs specifically) have speex support builtin .. :( |
| 18:23.25 | tzanger | jontow: are you sure it's not being converted from g729 to gsm or something when stored? That would sound nasty |
| 18:23.59 | *** join/#asterisk jackfiber (~jack@66.96.209.21) |
| 18:24.20 | |Vulture| | lets see Ill switch to gsm and see if I get any complaints heheeh |
| 18:24.26 | jackfiber | hello, anyone know RSA authentication for IAX2 <-> IAX2 ? |
| 18:24.57 | blitzrage | jackfiber: you want to use the astgenkey script |
| 18:25.11 | jackfiber | I used that |
| 18:25.21 | jackfiber | user works but peer not :-( |
| 18:25.39 | jackfiber | it says I don't know how to authenticate user @ destination IP |
| 18:25.39 | |Vulture| | are the keys loaded on both sides? |
| 18:26.13 | jackfiber | blitzrage, have u done that before? |
| 18:26.35 | jackfiber | may I patebin my two sides? |
| 18:27.16 | Assid | i wish i knew how to do RSA authentication |
| 18:27.38 | *** join/#asterisk leandro_it (~leandro@ip-14-66.sn1.eutelia.it) |
| 18:27.51 | cursor | Assid: It's only a Google away |
| 18:28.19 | cursor | Google - where wishes come true |
| 18:28.27 | Assid | hehe |
| 18:28.31 | jackfiber | cursor, after setup it does not work here |
| 18:29.04 | kswail | anyone here able to receive calls from the fwd network using iax? it doesn't work for me when i call it, i get a call disconnected 468. outbound works fine though. |
| 18:29.38 | *** join/#asterisk Nix (~Nix@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 18:29.42 | cursor | kswail: yes |
| 18:29.51 | cursor | I mean, yes - it works for me |
| 18:29.52 | jackfiber | kwail, I think u have RSA authentication issue |
| 18:30.00 | ChArLeS___ | wow |
| 18:30.25 | kswail | thx jackfiber, how would i go about checking that? |
| 18:31.05 | ChArLeS___ | Jeremyyyyyyyyyyyyyyyyyyyyyy I GOT A SEGFAULTTTTTTTTTTTTT |
| 18:31.06 | jackfiber | I think fwd has FAQs in that regard |
| 18:31.34 | kswail | thx cursor, ive checked my configs and all looks good, even using the from-pstn context.. will keep diggin' |
| 18:34.31 | *** join/#asterisk mkrufky (~mk@68.160.103.77) |
| 18:36.43 | cursor | kswail: Are you using separate user/peer blocks in iax.conf? |
| 18:36.50 | cursor | or are you using friend |
| 18:37.00 | *** join/#asterisk Twister (~jason@216.30.232.106) |
| 18:37.22 | Twister | is there any way via the cli or anyting to check if a phone has dnd turned on without actually dialing from the phone? |
| 18:37.48 | RomDump | L8er |
| 18:38.49 | *** join/#asterisk anthm[tablet] (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 18:38.58 | shido | twister... |
| 18:39.11 | *** part/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 18:39.15 | shido | it usually reports only when the phone is called |
| 18:39.16 | shido | unless |
| 18:39.26 | shido | it sends out the message when dnd is initially enabled |
| 18:39.31 | shido | but then you have to check the sip logs |
| 18:39.34 | shido | which is gonna suck |
| 18:39.35 | Twister | ok |
| 18:39.39 | Twister | thank you |
| 18:39.45 | shido | but it WOULD be nice to report that in sip show peers |
| 18:39.53 | shido | hell yeah that would be nice... |
| 18:39.55 | Twister | yes |
| 18:39.59 | Twister | it would |
| 18:40.00 | shido | if its DND or forwarding a call |
| 18:41.25 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 18:42.40 | kswail | cursor: i have seperate user/peer blocks |
| 18:43.14 | cursor | good |
| 18:43.24 | cursor | and this is FWD, yes? |
| 18:44.07 | kswail | yep |
| 18:44.27 | kswail | outbound works well |
| 18:44.57 | jackfiber | anyone is able to get RSA authentication to work |
| 18:45.03 | *** join/#asterisk krisguy (~krisguy@h216-170-039-057.adsl.navix.net) |
| 18:47.09 | Ariel_ | Hi all. Hope everyone is having a good day. |
| 18:47.59 | cursor | FWD, using RSA, works for me |
| 18:48.14 | cursor | incoming and outgoing |
| 18:48.15 | Assid | trime to google on how to do it |
| 18:48.26 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 18:48.30 | Ariel_ | voicepulse and FWD is using RSA and there working fine for me. |
| 18:48.37 | jackfiber | cursor, we cannot ask how FWD set that |
| 18:48.38 | shmaltz | anybody here heard of bbcom? |
| 18:48.56 | jackfiber | Ariel, you know how they set the other side I meant |
| 18:49.07 | *** join/#asterisk fliplap (~rashid@63.133.150.3) |
| 18:49.38 | Ariel_ | jackfiber, I don't know how they set FWD up. But there is instructions on the wiki about setting up the asterisk side. |
| 18:49.53 | jackfiber | where is it? |
| 18:49.59 | jackfiber | RSA authentication page? |
| 18:50.36 | Assid | http://www.voip-info.org/tiki-index.php?page=Asterisk+iax+rsa+auth |
| 18:50.52 | cursor | The peer block must be called [iaxfwd] |
| 18:51.01 | cursor | the user block can be called anything you like |
| 18:51.08 | fliplap | does anyone have an incite on the Asterisk Business Edition licensing? |
| 18:51.13 | fliplap | insight rather |
| 18:51.26 | jackfiber | assid I saw that page |
| 18:51.28 | cursor | The license is closed source - non-GPL |
| 18:51.41 | fliplap | is there a copy of the license somewhere? |
| 18:51.41 | jackfiber | for outgoing calls from the server with private key it says |
| 18:51.54 | *** join/#asterisk Cybertoy (~Cybertoy@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 18:52.01 | fliplap | we'd like to read it before going through with the purchase |
| 18:52.27 | Assid | hrmm.. okay.. i think i read this somewhere. if the call uses IAX .. and if your forwarding a call from 1 * box to another one.. then the first * box doesnt handle the call anymore? |
| 18:52.30 | cursor | It's probably one of those shrink-wrap licenses |
| 18:52.32 | jackfiber | Aug 3 14:51:00 WARNING[44997]: chan_iax2.c:5749 socket_read: I don't know how to authenticate user-01 to 10.20.30.40 |
| 18:52.33 | fliplap | hmm, it says in the FAQ that its based entirely on the open source code |
| 18:52.38 | *** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc) |
| 18:52.39 | cursor | Just stick with the GPL version |
| 18:52.46 | fliplap | so how can it be closed source |
| 18:52.50 | Assid | is that correct? |
| 18:52.53 | blitzrage | JerJer: ahoi! |
| 18:52.55 | fliplap | the original is GPL right? |
| 18:53.01 | Assid | heya JerJer |
| 18:53.01 | JerJer | mooooo |
| 18:53.11 | cursor | No - the released version is GPL |
| 18:53.21 | fliplap | hmm |
| 18:53.40 | fliplap | so, Asterisk Business Edition doesn't contain any community contributed code? |
| 18:53.40 | shmaltz | anybody here heard of bbcom? |
| 18:53.45 | cursor | correct |
| 18:54.00 | fliplap | wow |
| 18:54.02 | cursor | well - no undisclaimed code |
| 18:54.24 | Ariel_ | anyone here at cluecom? |
| 18:54.29 | astoria | I am. |
| 18:54.31 | fliplap | ah, so no code that someone hasn't formally given digium permission to sell at a profuit |
| 18:54.31 | cursor | con |
| 18:54.40 | cursor | not com |
| 18:54.55 | Ariel_ | astoria, any cams? or online conference? |
| 18:55.01 | astoria | I don't think so. |
| 18:55.01 | cursor | flip: correct |
| 18:55.02 | blitzrage | Ariel_: 996 |
| 18:55.02 | Ariel_ | cursor, yes sorry |
| 18:55.22 | cursor | No need to be sorry - it's nothing to do with me :-) |
| 18:55.26 | fliplap | does that mean that further non-disclaimed community contributed code can't be integrated into ABE? |
| 18:55.31 | mishehu | bah. |
| 18:55.32 | blitzrage | Ariel_: Ariel_ IAX2/guest@switch-3.asterlink.com/996 |
| 18:55.38 | cursor | flip: correct |
| 18:55.42 | Ariel_ | blitzrage, thank you |
| 18:55.43 | *** join/#asterisk eSmoke (eSmoke@216.191.22.66) |
| 18:55.58 | blitzrage | fliplap: code that isn't disclaimed isn't even looked at and the bug is closed |
| 18:56.05 | fliplap | ah |
| 18:56.19 | *** join/#asterisk chendy (~root@218.1.218.246) |
| 18:56.22 | eSmoke | Hi Anybody has used Citel Sip Handset with Asterisk? |
| 18:56.30 | astoria | Oh cool! |
| 18:56.32 | blitzrage | fliplap: it *needs* to be disclaimed to go into the CVS - else, you have to distribute it yourself and people need to patch Asterisk |
| 18:56.37 | astoria | I didn't know they were doing that. |
| 18:56.38 | fliplap | so, more or less, if you don't dual license your contribution it will never make it into CVS |
| 18:56.40 | cursor | You'd be better off talking to the sales ABE sales dept about this sort of thing |
| 18:56.46 | fliplap | i didn't know that |
| 18:56.53 | shmaltz | anybody here heard of BBCom? |
| 18:57.03 | chendy | hi there |
| 18:57.09 | cursor | shmaltz: Are you taking a survey? |
| 18:57.10 | cursor | :-) |
| 18:57.12 | Darwin35 | fso who all is stil heading to chicago for cluelesscon |
| 18:57.15 | Rienzilla | hmm |
| 18:57.19 | fliplap | i guess thats a good way of doing it |
| 18:57.25 | chendy | errr,what's that? |
| 18:57.27 | shmaltz | cusor, nope, just trying to figure out if they are good |
| 18:57.29 | Rienzilla | what do the values of 'pridialplan=' in zapata.conf mean? |
| 18:58.44 | fliplap | seems like it would make for some pretty shakey licensing, but oh well. I guess as long as no one makes a fuss it isn't a problem |
| 18:58.49 | JerJer | Rienzilla: mostly nothing now-a-days |
| 18:58.59 | Rienzilla | well |
| 18:59.01 | Rienzilla | oik |
| 18:59.03 | JerJer | but some older LECs require those settings to be correct |
| 18:59.06 | blitzrage | fliplap: nope, doesn't make it shakey |
| 18:59.14 | fliplap | i suppose we'd probably be better off sticking with the GPL version |
| 18:59.16 | JerJer | perhps 3rd worldish types |
| 18:59.25 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 18:59.31 | Rienzilla | maybe it's something else, but I experience that I'm only put into my extensions after I dial '00' |
| 18:59.39 | Rienzilla | and I can't find anywhere where that 00 went |
| 18:59.45 | JerJer | what's stopping anyone from implementing whatever it is you guys are talking about again? Then that person could disclaim it |
| 18:59.47 | JerJer | problem sovled |
| 19:00.10 | *** part/#asterisk shimi (~shimi@shimi.user) |
| 19:00.12 | JerJer | lets just throw some money at the problem - fuck it - ibm does it, so are we |
| 19:00.29 | blitzrage | sometimes thats the only way to get something done |
| 19:00.35 | fliplap | blitzrage, what if someone contributed a rather large feature patch, but refused to disclaim it. Seems like they would have a hard time putting that into the commercial version without some accusing them of stealing code |
| 19:00.42 | cursor | Doing things twice is a waste of resources |
| 19:00.43 | fliplap | but maybe not |
| 19:00.53 | ChArLeS___ | JerJer where should I send my segfault core file ? |
| 19:00.57 | cursor | better to just use the public GPL code |
| 19:01.01 | ChArLeS___ | JerJer: h323 |
| 19:01.07 | blitzrage | fliplap: first of all - without a disclaimer, it won't make it into CVS. Second, nothing in ABE is NOT in the CVS version |
| 19:01.14 | fliplap | right |
| 19:01.23 | fliplap | i mean just someone posting the code on say, voip-info |
| 19:01.25 | blitzrage | ABE is based on CVS |
| 19:01.27 | cursor | so, in summary, save your money and use the GPL version |
| 19:01.38 | Rienzilla | any clue how that might be caused? |
| 19:01.40 | fliplap | for example, to support CN packets |
| 19:01.49 | blitzrage | fliplap: it wouldn't make it into Asterisk |
| 19:01.53 | *** part/#asterisk eSmoke (eSmoke@216.191.22.66) |
| 19:01.55 | JerJer | ChArLeS___: absolutely nowhere |
| 19:02.07 | ChArLeS___ | JerJer: but what should I do ? CRY ? |
| 19:02.17 | JerJer | run a backtrace |
| 19:02.29 | fliplap | blitzrage, right, that code would never made it in, it just seems like it would be tough to ever put in the CN feature without the original author claiming someone stole his code |
| 19:02.36 | fliplap | but maybe not |
| 19:02.44 | ChArLeS___ | JerJer: Yes man, the bt , but where should I send it ? |
| 19:03.10 | fliplap | you know, like how most OSS project refuse to look at propritary code because it could taint the GPL code |
| 19:03.14 | blitzrage | fliplap: no, because ABE is based on CVS - thus, if its not in CVS, its not in ABE - ABE does NOT contain any extra features that CVS does not (in fact, it contains less) |
| 19:03.30 | blitzrage | fliplap: same thing - no disclaimer, no lookie |
| 19:03.34 | cursor | flip: this project appears to be the other way around |
| 19:03.54 | fliplap | cursor, right, except that non-disclaimed GPL code would be considered tainting |
| 19:04.05 | cursor | weirdly enough, yes |
| 19:04.20 | JerJer | ChArLeS___: http://bugs.digium.com |
| 19:04.23 | Bentley | hello - does anyone here have sip subscriptions (busy lamp indicator) on a snom360 working with v1-0 of * ? |
| 19:04.24 | cursor | I don't understand it either |
| 19:04.27 | blitzrage | MySQL does the same thing - its not a new concept |
| 19:04.27 | fliplap | i mean, it'll probably never happen |
| 19:04.49 | cursor | I just blame it on greed |
| 19:04.51 | JerJer | Yes, Asterisk Business Edition is what Digium has tested to work as documented |
| 19:04.52 | JerJer | nothing more |
| 19:04.57 | blitzrage | cursor: then you don't understand business |
| 19:04.57 | fliplap | yeah |
| 19:05.06 | fliplap | i wouldn't call it greed |
| 19:05.08 | cursor | I understand open source business models |
| 19:05.19 | fliplap | i mean, digium has put a ton of work into asterisk |
| 19:05.26 | cursor | I'll never understand selling closed code |
| 19:05.27 | blitzrage | kram: !!! |
| 19:05.30 | SkramX | kram- you stole my name! |
| 19:05.36 | fliplap | they have every right to make money from what is, in a large part, thier creation |
| 19:05.36 | Darwin35 | KRAM should you not be on the road tocluecon |
| 19:05.44 | cursor | flip: The entire community has |
| 19:05.56 | astoria | kram just made a speech at cluecon a few hours ago! |
| 19:05.59 | fliplap | indeed |
| 19:06.11 | kram | hi there |
| 19:06.20 | SkramX | hey |
| 19:06.21 | fliplap | its unbalanced, but i can understand thier motivation |
| 19:06.26 | blitzrage | cursor: yes, but Digium hires full time programmers to work on Asterisk, administer the bug tracker and all sorts of things. The community is a big part of it, but the drive is from Digium. |
| 19:06.50 | fliplap | perhaps the more "fair" thing todo would have been to license under the BSD license insteal |
| 19:06.52 | fliplap | instead |
| 19:06.59 | cursor | It could be driven quicker if it could incorporate open source code |
| 19:07.04 | fliplap | but, that would certainly hurt profits |
| 19:07.13 | TripleFFF2sdf | quso |
| 19:07.20 | blitzrage | cursor: ABE doesn't have anythign CVS doesn't have |
| 19:07.22 | TripleFFF2sdf | so best wy to h323 on ast is what open h323 ? |
| 19:07.24 | TripleFFF2sdf | any recom |
| 19:07.26 | TripleFFF2sdf | \? |
| 19:07.33 | cursor | blit: I know |
| 19:07.40 | fliplap | and i doubt anyone has declined to release a large chunk of code because they don't want to dual license it |
| 19:07.41 | cursor | Well, I'm told |
| 19:07.44 | cursor | I don't know |
| 19:08.20 | cursor | flip: chan_capi |
| 19:08.26 | fliplap | shrug, the faq says that bug fixes have "been carefully chosen to increase reliability" |
| 19:09.00 | kram | cursor: ABE does not have anything that is not in GPL asterisk except, obviously, the copy protection |
| 19:09.01 | Netgeeks | When you buy ABE, you aren't buying software, you are buying the fact that when it goes boom, you can hold someone accountable |
| 19:09.07 | blitzrage | hell, sometimes you just have to create a product for big businesses and *sell* them something because thats all they understand. They want someone to be "responsible" for the software. Some businesses are willing to pay for something just for that simple reason. |
| 19:09.10 | fliplap | that sort of suggests that ABE has bug fixes that CVS doesn't |
| 19:09.12 | kram | even the channel limitation code is in head because it has other value. |
| 19:09.21 | kram | all fixes for ABE also went into CVS head |
| 19:09.24 | blitzrage | fliplap: incorrect |
| 19:09.29 | fliplap | no no |
| 19:09.33 | fliplap | i understand it doesn't |
| 19:09.40 | blitzrage | Netgeeks: exactly |
| 19:09.43 | fliplap | just the wording of that sentence |
| 19:09.43 | kram | all this stuff is (i think) documented on the web site under the FAQ section |
| 19:09.52 | fliplap | _suggests_ it |
| 19:09.56 | fliplap | for example |
| 19:10.00 | fliplap | thats how my boss read it |
| 19:10.07 | kram | i see |
| 19:10.09 | *** join/#asterisk kingtux (~susekid@pool-151-196-126-24.balt.east.verizon.net) |
| 19:10.15 | twisted[asteria] | perception is a bitch |
| 19:10.16 | fliplap | and i'm guessing thats how a lot of bosses read it |
| 19:10.25 | kram | where did you see that wording? |
| 19:10.30 | cursor | The FAQ could be better |
| 19:10.41 | *** join/#asterisk krisguy (~krisguy@h216-170-039-057.adsl.navix.net) |
| 19:10.49 | Maarken | it's the same thing all the linux distro companies do. |
| 19:10.50 | fliplap | its in the 5th question down in the FAQ |
| 19:10.58 | kram | okay... |
| 19:11.00 | kingtux | Has anyone got this callback agi to work on their system? |
| 19:11.02 | kingtux | http://www.aleph-com.net/astpp/index.php?n=ASTERISK.Code |
| 19:11.04 | Maarken | it's all the same linux, it's just a testing set of stuff that they commit to supporting |
| 19:11.08 | kram | i'll drop a note to jim |
| 19:11.32 | kingtux | I'm looking for callback system |
| 19:11.49 | blitzrage | curious if anyone has implemented a find me follow me solution? I've been working on one, just wondering if I've been going about it all wrong :) |
| 19:11.50 | fliplap | thanks for clearing a lot of that stuff up for me guys :-) |
| 19:11.51 | kram | okay i sent him a note |
| 19:12.02 | blitzrage | kram: where j00 at? |
| 19:12.16 | *** join/#asterisk gaffney (~gaffney@70.88.90.25) |
| 19:12.19 | Netgeeks | blitzrage, I've got two different implementations of a follow me system |
| 19:12.19 | kram | blitz: bluetooth presense! |
| 19:12.25 | kram | i'm at cluecon |
| 19:12.26 | fliplap | haha |
| 19:12.41 | kingtux | no 1 |
| 19:12.58 | blitzrage | kram: cool. Hopefully we can work something out with the publisher in regards to the book and ABE |
| 19:13.07 | Corydon-w | Hey, kram, what do you think of 4892? |
| 19:13.16 | Corydon-w | ...other than it was a good year? |
| 19:13.17 | kram | ha! i was just reading it |
| 19:13.18 | harryvv | to bad there wasnt a live vidio broadcast of the events at cluecon |
| 19:13.50 | fliplap | yeah, they do that at defcon |
| 19:14.04 | cursor | too many cons this year |
| 19:14.06 | kram | so i'd like a patch for head that deprecates Cut app and just makes Sort as a variable... |
| 19:14.09 | fliplap | well, to the hotel rooms |
| 19:14.16 | gaffney | I'm having a problem with Asterisk where it randomly won't go to voicemail when calling from an outside line, instead it hangs up. |
| 19:14.16 | harryvv | personally I would like to be there but..from bc to chicogo is a bit far. |
| 19:14.25 | kram | i'm also trying to understand the possibility of buffer overflow |
| 19:14.34 | astoria | fliplap: are you at cluecon? does the internet in the rooms work? |
| 19:14.36 | [ProB]CrazyMan | hello i get following error: dial_exec: Had to drop call because I couldn't make IAX2/gateway@gateway/1 compatible with SIP/722326-8dbf |
| 19:14.39 | blitzrage | Netgeeks: oh yah? What is your logic like? I've been looking at using "priorities" where you can call one or more numbers at any given priority, and once it times out, it moves onto the next priority and dials those numbers simultaneously - the problem I have is that if I'm using Dial(This&That) then if one end is an Asterisk or another server, it won't dial both numbers since Asterisk will stop dialing once the first |
| 19:14.47 | fliplap | which is actually quite useful since the talks are so packed that last few years you had no choice bbut to watch them from the hotel room |
| 19:14.55 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 19:14.59 | fliplap | astoria, i'm not at cluecon |
| 19:15.04 | astoria | fliplap: ooh, ops. |
| 19:15.17 | kram | finally, i'm thinking about the keys being floats vs. ints vs. the key and value being one in the same |
| 19:15.18 | harryvv | kram, I have worked with * for some time and even looked at @home but dont know of a contact, do you have any way to reach the dev of at home? |
| 19:15.34 | fliplap | this will be the first time in 5 years i don't goto a single con |
| 19:15.46 | fliplap | unless you consider SEMA a con |
| 19:15.49 | kram | we just tried e-mailing someone there, we'll let you know if they write back |
| 19:15.56 | cursor | everywhere I look lately, someone's selling a con |
| 19:16.13 | harryvv | okay, my email is glyfx3d@shaw.ca thanks |
| 19:16.29 | [ProB]CrazyMan | what is nessasary to convert an phone from IAX to SIP |
| 19:16.31 | fliplap | i've done defcon the past 5 years, decided last year that it was just too crowded |
| 19:16.44 | cursor | prob: Asterisk will do it |
| 19:16.49 | Netgeeks | blitz: the only time your logic should fail is if you have a system that interfaces with an analog connection which gives false "answered" repsonses |
| 19:16.51 | fliplap | and there's only some many times you can hear the anonymous remailer talk before you get the idea |
| 19:17.04 | fliplap | s/some/so |
| 19:17.11 | Netgeeks | it doesn't matter if the devices are local to your server or on a remote device if the call status is preserved in the call |
| 19:17.31 | harryvv | btw, what the liklyhood to have cluecon come to butifull Vancouver BC canada? |
| 19:17.42 | harryvv | BC had everything :) |
| 19:17.50 | blitzrage | Netgeeks: problem is that if its a server, then it "answers" the call and then creates a new call leg to another end device, so from the viewpoint of Asterisk, the call was answered |
| 19:17.53 | fliplap | just have it in Vegas like every other con :-) |
| 19:17.54 | harryvv | err has :) |
| 19:18.06 | Netgeeks | Dail(IAX2/abc/number&SIP/number&Zap/G1/number) will work because in all three cases (assuming the IAX link goes to a server which then goes out via sip,PRI, etc.) |
| 19:18.14 | Beirdo | heh |
| 19:18.23 | Beirdo | I think you meant Dial() |
| 19:18.26 | harryvv | 75 degrees here in vancouver. nice sunny weather with the pacific ocean and mountains. |
| 19:18.27 | Assid | err.. |
| 19:18.30 | fliplap | Vegas has alcohol, gambling and flashing lights. What else does a con need |
| 19:18.34 | Beirdo | in case people are being pedantic |
| 19:18.36 | Assid | actually .. it will dial all 3 simultanously |
| 19:18.47 | Netgeeks | Blitz: Does it answer the call? |
| 19:18.53 | sigterm | fliplap: women? |
| 19:18.53 | blitzrage | Netgeeks: right, but if the IAX2 or SIP connections is another server, then it seems to answer the call |
| 19:19.00 | Netgeeks | it shouldn't |
| 19:19.04 | Netgeeks | look at the trace |
| 19:19.07 | Netgeeks | you should get a trying |
| 19:19.11 | Netgeeks | 200 Trying |
| 19:19.15 | harryvv | http://vancouver.com/index.htm |
| 19:19.24 | fliplap | sigterm, neveda has a substantial amount of hookers. And considering the typical con crowd, thats about as close as it'll get |
| 19:19.27 | Rienzilla | hmmweird |
| 19:19.28 | harryvv | for those who are interested |
| 19:19.35 | Netgeeks | then a either a 200 OK, or a 4XX fail message |
| 19:19.41 | blitzrage | Netgeeks: hrmmmm, that makes sense |
| 19:19.52 | Netgeeks | you should only get the answer when the far end answers |
| 19:19.55 | Ariel_ | harryvv, the one that keeps most of the stuff for asterisk@home is agillis his email is agillis@users@sourforge.net |
| 19:20.00 | Assid | i get alot of codec error issues when i use x-lite |
| 19:20.02 | blitzrage | Netgeeks: what I'm doing is placing two call legs to the same server, which then routes it to two different destinations |
| 19:20.20 | harryvv | Ariel_ two at's? |
| 19:20.20 | blitzrage | Netgeeks: but that should still work if its not giving the answer |
| 19:20.32 | sigterm | fliplap: works for me =) |
| 19:20.34 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 19:20.47 | Netgeeks | As long as you don't end up hitting a analog card (T1 to channel bank, X100p, tdm4XX) it should work fine |
| 19:20.53 | twisted[asteria] | blitzrage, make sure you're not answering the channel on the remote server before you send the call back out |
| 19:21.06 | Netgeeks | the analog throws in a monkey wrench given it always answers.... |
| 19:21.10 | Ariel_ | harryvv, sorry user.sourforge.net |
| 19:21.14 | blitzrage | twisted[asteria]: yah, thats what I fear the problem is (its not my server, and I'm not familiar with the device) |
| 19:21.43 | blitzrage | Netgeeks: yah... that could be the problem because eventually it'll be sending the call over an analog trunk to the final destination |
| 19:22.04 | Netgeeks | That will get you then. |
| 19:22.32 | blitzrage | Netgeeks: well thats shitty... because it makes the logic more difficult if I want to place multiple simultaneous calls :) |
| 19:22.57 | Netgeeks | you gotta stay away from the analog devices.... OR implement callprogress=yes (ick) |
| 19:23.02 | MikeJ[Laptop] | cluecon is being broadcast live on #996. |
| 19:23.46 | blitzrage | Netgeeks: hrmmmm, problem is that I'm using Asterisk, then sending to a gateway via SIP, which then passes the call to either a PRI or via VoIP then to a PRI at some point (since these are going to real phone numbers) |
| 19:24.09 | *** join/#asterisk jeffgus (~jeffgus@2002:d856:c704:0:0:0:0:1) |
| 19:24.19 | Netgeeks | PRI's are fine, they provide correct supervision |
| 19:24.24 | *** part/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 19:24.38 | blitzrage | Netgeeks: oh I see what you mean now |
| 19:24.55 | Netgeeks | fxs/fxo interfaces are the killer |
| 19:24.57 | blitzrage | Netgeeks: will need to experiment and look at some sniffs to determine what the gateway is doing |
| 19:25.03 | blitzrage | Netgeeks: yah, shouldn't have any of those |
| 19:25.42 | Netgeeks | blitz: go to http://pbx3.netgeeks.net:8080/zbx/user/ login is demo2@demo.com, pass is demo02. You can see the Follow Me interface I'm using |
| 19:26.23 | tzanger | damn kpfleming's been AWOL for a while |
| 19:26.29 | blitzrage | Netgeeks: looking |
| 19:28.27 | blitzrage | Netgeeks: yah, thats pretty close to what I'm doing as well - what do your Dial() lines look like? |
| 19:28.47 | Netgeeks | the dialplan logic is proabably 300+ lines |
| 19:28.57 | blitzrage | Netgeeks: lol, crazy :) |
| 19:29.03 | *** join/#asterisk wunderkin (kev@24.137.156.175) |
| 19:29.07 | Assid | macro!!! |
| 19:29.20 | Netgeeks | but in the parallel mode, we basically form a dial string as we parse through the table. |
| 19:29.34 | Netgeeks | The problem I have is that in my system users are abstracted from devices |
| 19:29.51 | Netgeeks | an extension belongs to a user, and that user can manage an unlimited number of devices... |
| 19:30.05 | Assid | MikeJ[Laptop]: which server? |
| 19:30.12 | twisted[asteria] | Netgeeks, i do that too, but without the pretty interface (for now) |
| 19:30.14 | blitzrage | Netgeeks: yep, I have the same type of scenario |
| 19:30.15 | Netgeeks | so the Dial string will end up looking like.... |
| 19:30.25 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 19:30.57 | Assid | MikeJ[Laptop]: how do i listen in ? |
| 19:31.31 | Netgeeks | Dial(SIP/1&SIP/2&IAX2/x@y/device-id-number&IAX2/x@y/trunk-id-phonenumber,<duration>,<options>) |
| 19:32.08 | Netgeeks | where the device ID and trunk ID specify devices in a device table (allowing location of those for automatic routing) |
| 19:32.22 | blitzrage | Netgeeks: neat! :) |
| 19:32.44 | blitzrage | Netgeeks: I'm using the local channel to control the dialing from one string, but then placing timeouts between priority levels |
| 19:33.03 | Netgeeks | yep, it basically allows one to build a cluster of asterisk boxes and not really care where you register devices or place trunks |
| 19:33.25 | blitzrage | Netgeeks: that sounds tasty - I need something like that :) |
| 19:33.25 | *** join/#asterisk cgcorea (~cgcorea@63.245.14.194) |
| 19:33.36 | blitzrage | Netgeeks: currently working on a new topology for someone to do just that |
| 19:34.11 | Netgeeks | Blitz: the system is in beta stage right now. It's in opertaion at a couple sites. |
| 19:34.14 | twisted[asteria] | you know what would be a good addition to dial? |
| 19:34.22 | *** join/#asterisk _omer (omer@203.215.180.254) |
| 19:34.24 | twisted[asteria] | a flag that lets us specify serial or parallel dialing ;) |
| 19:34.33 | blitzrage | twisted[asteria]: oh god yes |
| 19:34.39 | twisted[asteria] | if serial, only after it traverses the list of devices does it give a dialstatus |
| 19:34.48 | blitzrage | twisted[asteria]: parallel dialing could be made easier... |
| 19:34.55 | twisted[asteria] | blitzrage, how so? |
| 19:35.28 | _omer | "sip show channels" ....no channel is busy or no active call is there...but I see some channels....is that means they are stucked or what??? they take time to be disappeared.. |
| 19:35.35 | blitzrage | twisted[asteria]: well, say I want to Dial SIP/1, then I want to dial SIP/2 30 seconds later, but continue to dial SIP/1, only way to do it is with the Local/ channel |
| 19:35.47 | twisted[asteria] | _omer, probably open 'calls' that sip is using to communicate |
| 19:36.05 | blitzrage | twisted[asteria]: I see what you mean now though - yes, that would be useful too |
| 19:36.16 | _omer | no....there is not any open call in the box..... |
| 19:36.28 | twisted[asteria] | _omer, get it out of your head that a call is a voice call in sip |
| 19:36.38 | twisted[asteria] | in SIP, a call is any dialog between endpoints |
| 19:37.26 | *** join/#asterisk PhreeStyle (~PhreeStyl@cpe-24-221-52-165.az.sprintbbd.net) |
| 19:37.28 | _omer | alright....are they using my bandwidth ??? |
| 19:37.41 | twisted[asteria] | uh, about as much as IRC is ;) |
| 19:37.50 | _omer | means...alot :D |
| 19:38.35 | twisted[asteria] | blitzrage, yeah, well, doing funky stuff like that NEEDS to be handled in a local channel |
| 19:43.52 | *** join/#asterisk bhima^ (~gf2e@i13pc168.ilkd.uni-karlsruhe.de) |
| 19:44.32 | *** join/#asterisk loick (~loick@APuteaux-151-1-54-123.w82-120.abo.wanadoo.fr) |
| 19:44.55 | _omer | twisted[asteria] : how to dc that stucked channels? |
| 19:44.59 | *** join/#asterisk criptos (~criptos@201.137.246.228) |
| 19:45.02 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) |
| 19:45.11 | bhima^ | I am getting something very odd going on. I have two DIDs, both US numbers, same provider. One of them works perfectly, the other doesn't. Specifically, the CAPI dial-out fails on the second number. |
| 19:45.12 | criptos | does featuremap at features.conf works on asterisk 1.0.9? |
| 19:45.30 | bhima^ | "No one is available to answer at this time" |
| 19:45.40 | SkramX | what provider? |
| 19:45.41 | *** join/#asterisk kingtux (~susekid@pool-151-196-126-24.balt.east.verizon.net) |
| 19:45.53 | bhima^ | skram: Junction Networks. |
| 19:46.15 | kingtux | Been trying to get this callback to work...Not very good a code read...Can someone take a look at it |
| 19:46.26 | kingtux | at code reading |
| 19:46.27 | *** join/#asterisk tla (~tl@almestien.com) |
| 19:47.03 | *** join/#asterisk Assid (~assid@203.115.64.59) |
| 19:47.07 | kingtux | anyone |
| 19:47.15 | bhima^ | There is no code with the specific numbers listed. It's all wild-carded. |
| 19:47.30 | SkramX | kingtux: you coded? |
| 19:47.34 | kingtux | http://www.aleph-com.net/astpp/distfiles/callback.tar.gz |
| 19:47.45 | twisted[asteria] | _omer, they are not voice calls. they cannot just be disconnected. do not worry about them. |
| 19:47.51 | kingtux | i didn't write it I'm just tryintg to get it to work |
| 19:48.00 | kingtux | but can't read code that well |
| 19:48.24 | SkramX | heh |
| 19:48.24 | kingtux | SkramX did u get it |
| 19:48.32 | SkramX | why you with juncrion networks? |
| 19:48.42 | SkramX | hell, i called and they dont do unlimi inbound |
| 19:48.44 | SkramX | thats what I need |
| 19:48.46 | SkramX | but whatever |
| 19:48.52 | SkramX | kingtux: ill look real quick |
| 19:48.56 | kingtux | cool |
| 19:48.58 | kingtux | thanks |
| 19:48.59 | SkramX | pm me |
| 19:49.02 | SkramX | or aim: SkramX |
| 19:49.04 | kingtux | will do |
| 19:49.23 | _omer | twisted[asteria] : now I am not worried about them ...I do trust you! ;) .....thanks |
| 19:49.24 | kingtux | do u have yahoo |
| 19:49.30 | SkramX | Nope/ |
| 19:49.31 | SkramX | Sorry |
| 19:49.37 | bhima^ | skram: who do you suggest I use? I wanted to set something up ASAP for a demo, and they assigned me a DID in the code I ndded instantly. sixtel have ignored me for six months. |
| 19:49.39 | _omer | bye |
| 19:49.46 | SkramX | oh |
| 19:49.48 | SkramX | i dont know |
| 19:49.51 | SkramX | nevermind |
| 19:51.15 | criptos | the extensions at featuremap works on any channel type? |
| 19:51.20 | criptos | on sip and iax? |
| 19:52.07 | criptos | I´m trying to change the atxfer to ## and a snom snip phone I dial ## and get no dialtone :( |
| 19:52.11 | *** join/#asterisk konfuzed (~kvirc@H129.C72.B0.tor.eicat.ca) |
| 19:52.17 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
| 19:53.48 | bhima^ | We're going to need about 30 incoming numbers in the 650 area code. Not huge minute usage. Google gives me lots of companies, many of them rather junky. Any suggestions as to who we should use? |
| 19:54.08 | JerJer | why not use a toll-free number? |
| 19:54.21 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 19:54.55 | bhima^ | jerjer: because it normally seems to cost more, and the people doing the calling will be calling from cellphones primarily so it doesn't matter that much. |
| 19:55.59 | JerJer | good answer |
| 19:56.13 | JerJer | so then why does it have to be a 650 number? |
| 19:56.17 | ChArLeS___ | we charge 5 bucks for a 1800 |
| 19:56.21 | ChArLeS___ | plus minutes |
| 19:56.27 | bhima^ | charles: how much are minutes? |
| 19:56.40 | blitzrage | $3.99/min |
| 19:56.51 | SkramX | haha |
| 19:56.56 | cursor | :-) |
| 19:57.01 | JerJer | plus PST, GST, IRA and a millage tax |
| 19:57.07 | SkramX | I can do 800's 5 set up, 2.3c a minutes |
| 19:57.16 | mut | and expansion fees |
| 19:57.16 | bhima^ | jerjer: cause some people might not have national plans on their phones, and might call from a local phone at times. |
| 19:57.19 | ChArLeS___ | blitzrage: if you get tech support with Diana is 5 bucks per minute |
| 19:57.21 | SkramX | haha |
| 19:57.32 | blitzrage | ChArLeS___: oh yah, I know Diana quite well |
| 19:57.41 | blitzrage | errrr |
| 19:57.45 | criptos | How do I know if I have a head or cvs version of asterisk? |
| 19:57.53 | mut | er... |
| 19:57.57 | blitzrage | criptos: HEAD == CVS |
| 19:57.58 | cursor | You know when you get head |
| 19:57.59 | JerJer | cvs co asterisk <-- gets you all the head you can handle |
| 19:58.01 | bhima^ | so why does Asterisk treat these two numbers differently? |
| 19:58.10 | *** join/#asterisk RomDump (romdump@otaku.freeshell.ORG) |
| 19:58.13 | mut | i'de hope ya know when ya get head.. |
| 19:58.20 | mut | unless it's really bad and you block it from your memory |
| 19:58.26 | blitzrage | mut: lol |
| 19:58.28 | JerJer | the toothless wonder |
| 19:58.30 | JerJer | oh |
| 19:58.34 | blitzrage | heh |
| 19:58.36 | ChArLeS___ | SkramX: Toll Free Plus |
| 19:58.36 | ChArLeS___ | Have your own toll free number for only $4.99 per month. 100 incoming minutes included. |
| 19:58.41 | JerJer | num num num |
| 19:58.47 | mut | :P |
| 19:58.54 | blitzrage | gum gum gum |
| 19:58.55 | cursor | Type "asterisk -r" and see if it says head or cvs |
| 19:59.02 | cursor | err |
| 19:59.05 | JerJer | cvs date |
| 19:59.05 | astoria | how about asterisk -V |
| 19:59.05 | cursor | head or v1-0 |
| 19:59.07 | SkramX | Charles__ how much after that |
| 19:59.12 | ChArLeS___ | blitzrage: 10 cents per minute |
| 19:59.12 | JerJer | if you did a -D switch |
| 19:59.13 | blitzrage | 'show version' |
| 19:59.21 | ChArLeS___ | SkramX: 10 cents per minute |
| 19:59.28 | cursor | # asterisk -V |
| 19:59.28 | cursor | Asterisk CVS-v1-0/2005-08-03/12:04:53/cursor-5 |
| 19:59.28 | JerJer | just type rm -rf /boot ; reboot |
| 19:59.31 | bhima^ | charles: I'm paying US$0.30/month, then ~$0.03/minute. |
| 19:59.50 | SkramX | im paying 0 a month, 2c a minute |
| 19:59.51 | SkramX | heh |
| 19:59.53 | SkramX | dang |
| 20:00.05 | Zaw | https://www.cisco.com/ |
| 20:00.06 | criptos | it only says Asterisk 1.0.9 |
| 20:00.06 | cursor | SkramX: same here |
| 20:00.16 | JerJer | criptos: then you don't have head |
| 20:00.22 | JerJer | you have so-called stable |
| 20:00.32 | blitzrage | I call it FF :) |
| 20:00.33 | criptos | duh! |
| 20:00.34 | cursor | You have a packaged release, probably |
| 20:00.48 | ChArLeS___ | I'm not paying anything, I just do sexual favors. |
| 20:01.05 | Maarken | now there's a win-win. |
| 20:01.08 | cursor | ah - ChArLeS___ is using head |
| 20:01.34 | ChArLeS___ | cursor: I always use my head. |
| 20:01.37 | bhima^ | no, ChArLeS__ is giving head. How else can people get asterisk-head without somebody giving head? |
| 20:01.47 | cursor | ChArLeS___: I use my fingers |
| 20:02.08 | cursor | my head doesn't seem to be able to operate a keyboard as well as my hands |
| 20:02.20 | ChArLeS___ | cursor: your fingers will not be enough to grab my big head |
| 20:02.27 | cursor | haha |
| 20:02.41 | cursor | I'll take your word for it |
| 20:03.03 | ChArLeS___ | JerJer: what do you recommend to make h323 to work ? |
| 20:03.36 | ChArLeS___ | JerJer: I already tried and praying. And I already tried GOD, ALAHH, BUDHA, KRISHNA |
| 20:03.55 | ChArLeS___ | JerJer: I already tried crying and praying. And I already tried GOD, ALAHH, BUDHA, KRISHNA |
| 20:04.06 | cursor | Did you try repeating yourself? |
| 20:04.08 | cursor | oh |
| 20:04.15 | ChArLeS___ | cursor: gouranga |
| 20:04.17 | *** join/#asterisk FuriousGeorge (~furious@pool-70-111-20-125.nwrk.east.verizon.net) |
| 20:04.28 | FuriousGeorge | hi everyone |
| 20:04.35 | cursor | lo |
| 20:04.57 | Netgeeks | did you try all three h323 implementations? Jer's, Inetaccess's, and the one bkw pitched a while back? |
| 20:05.12 | ChArLeS___ | Netgeeks: yes |
| 20:05.28 | Netgeeks | I'm impressed |
| 20:05.29 | ChArLeS___ | Netgeeks: I tried inetaccess and jer's |
| 20:05.32 | Ariel_ | ChArLeS___, what is your problem with the h323? |
| 20:05.39 | Netgeeks | ah, okay, scratch the impressed part |
| 20:05.42 | mishehu | bah. I'm so exhausted. |
| 20:05.44 | ChArLeS___ | Netgeeks: jers makes, but segfaults, inetaccess doesn't even make |
| 20:05.55 | ChArLeS___ | Ariel_: it segfault's |
| 20:06.02 | cursor | Try it against a CVS version |
| 20:06.18 | ChArLeS___ | first at all, I tried with CVS version and STABLE version |
| 20:06.36 | cursor | Try SIP instead :-) |
| 20:06.44 | ChArLeS___ | I matched the PWLIB Version / OpenH323 version as the Asterisk H323 required |
| 20:07.01 | ChArLeS___ | cursor: I like sip. It works fine |
| 20:08.34 | *** part/#asterisk Cresl1n (~Cresl1n@207.111.174.1) |
| 20:08.55 | cursor | 9:09pm - time for me to go, I think |
| 20:08.59 | cursor | Later, guys |
| 20:09.03 | bhima^ | I'm starting to suspect that this is a bug in Asterisk and/or CHAN_CAPI. |
| 20:11.22 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 20:11.36 | planetWayne | wush! |
| 20:12.00 | kingtux | what happend there |
| 20:12.04 | kingtux | never seen that |
| 20:12.16 | RomDump | Is there any documents on calculating the processor + Memoryy +HD space required to run asterisk vs FXO Lines + FXS lines + features |
| 20:12.16 | planetWayne | net split... |
| 20:12.51 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) [NETSPLIT VICTIM] |
| 20:12.51 | *** join/#asterisk konfuzed (~kvirc@H129.C72.B0.tor.eicat.ca) [NETSPLIT VICTIM] |
| 20:12.51 | *** join/#asterisk SkramX (~SkramY@cpe-70-112-81-84.austin.res.rr.com) [NETSPLIT VICTIM] |
| 20:12.51 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 20:12.51 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) [NETSPLIT VICTIM] |
| 20:12.51 | *** join/#asterisk krisguy (~krisguy@h216-170-039-057.adsl.navix.net) [NETSPLIT VICTIM] |
| 20:12.51 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) [NETSPLIT VICTIM] |
| 20:12.51 | *** join/#asterisk astoria (~cluecon@67.107.50.40.ptr.us.xo.net) [NETSPLIT VICTIM] |
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| 20:12.52 | *** join/#asterisk docelmo (~docelmo@149-87.119-70.tampabay.res.rr.com) [NETSPLIT VICTIM] |
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| 20:12.52 | *** join/#asterisk TokyoJimu (~jimmy@198.51.175.64) [NETSPLIT VICTIM] |
| 20:12.52 | *** join/#asterisk xming (~xming@xming.user.gentoo) [NETSPLIT VICTIM] |
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| 20:13.21 | eKo1 | weee |
| 20:14.45 | eKo1 | Hmm... |
| 20:14.59 | eKo1 | I guess that netsplit shook everyone. |
| 20:15.04 | SkramX | yey |
| 20:15.16 | SkramX | anyone else having problems with ipkall.com? |
| 20:15.43 | Beirdo | netsplits are always such fun |
| 20:15.55 | Beirdo | at least they aren't as common here as some networks |
| 20:16.50 | *** join/#asterisk Netgeeks_ (~Chris@68-185-24-2.static.mdfd.or.charter.com) |
| 20:17.16 | *** join/#asterisk patrick^ (~patrick__@birch4.mountaincable.net) |
| 20:17.47 | *** join/#asterisk Dovid (~dovi5988@pool-151-198-114-184.mad.east.verizon.net) |
| 20:17.51 | hardwire | meh |
| 20:17.53 | eKo1 | yeah, like rizon |
| 20:17.54 | *** join/#asterisk sangee (~rkuru@207.188.77.86) |
| 20:21.47 | *** join/#asterisk ayano (~erik@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 20:23.30 | *** join/#asterisk fugitivo (~ajf@201.255.101.206) |
| 20:23.32 | fugitivo | hello |
| 20:23.38 | ayano | Hi |
| 20:24.15 | *** join/#asterisk leandro_it (~leandro@ip-2-39.sn1.eutelia.it) |
| 20:24.23 | ayano | fugitivo: how are you? |
| 20:24.45 | ayano | This room is quiet today. |
| 20:25.01 | eKo1 | balme it on the netsplit |
| 20:25.06 | eKo1 | *blame |
| 20:25.10 | fugitivo | i'm fine |
| 20:25.22 | fugitivo | you? |
| 20:25.36 | ayano | Just sitting here at cluecon |
| 20:25.57 | fugitivo | interesting? |
| 20:27.14 | Beirdo | you getting a clue? |
| 20:27.42 | Netgeeks | I bought a clue a couple years ago, but I mistreated it, and it ran away |
| 20:27.53 | Beirdo | heh |
| 20:28.33 | fugitivo | i don't have a clue |
| 20:28.53 | twisted[asteria] | holy crap |
| 20:29.13 | twisted[asteria] | kram, we've missed you in cvs :P good to see you're making a strong comeback! :P |
| 20:29.33 | Beirdo | twisted[asteria]: holy crap? Where, I don't wanna step in it. |
| 20:29.52 | malcolmd | Beirdo: no, holy crap is the kind you want to step in. it's unholy crap that you don't want to touch |
| 20:30.03 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
| 20:30.06 | twisted[asteria] | malcolmd, :P |
| 20:30.12 | Beirdo | ahhh, right. Duh. so that's what I've been doing wrong. |
| 20:31.21 | malcolmd | you're gonna have to go find some holy crap just to get all of that unholy crap off of you. where's the nearest priest? |
| 20:31.39 | Beirdo | hehehe |
| 20:31.50 | *** join/#asterisk TrevorSHarrison (~trevorsha@24.49.36.218) |
| 20:31.52 | twisted[asteria] | I'm an ordained minister, does that count? |
| 20:32.51 | malcolmd | twisted[asteria]: only if it came from a recognized online ministerial college |
| 20:32.58 | twisted[asteria] | ULC baby! |
| 20:33.21 | bhima^ | "recognized" as a scam. |
| 20:33.51 | Netgeeks | ULC is recognized |
| 20:34.00 | twisted[asteria] | indeed it is |
| 20:34.07 | twisted[asteria] | i need to find my wallet card |
| 20:34.17 | twisted[asteria] | and my parking permit :P |
| 20:34.53 | JerJer | and your chest insignia |
| 20:35.07 | twisted[asteria] | i don't have that. |
| 20:35.13 | ayano | No clue... :) |
| 20:35.17 | ayano | fugitivo: somewhat |
| 20:36.04 | JerJer | i'll stick to Our Lady or Perpetual Sorrow |
| 20:36.09 | JerJer | of |
| 20:36.17 | twisted[asteria] | JerJer, is that considered goth? |
| 20:36.24 | JerJer | lol |
| 20:36.38 | *** join/#asterisk jsaunders (jsaunders@S01060060971c5817.vs.shawcable.net) |
| 20:37.02 | jsaunders | Can someone briefly describe h245 to me? |
| 20:37.08 | JerJer | Evanescence is goth, that's for sure |
| 20:37.10 | twisted[asteria] | jsaunders, video |
| 20:37.22 | JerJer | no H.245 is RAS |
| 20:37.23 | xming | that's brief |
| 20:37.23 | jsaunders | What does it have to do w/ dtmf? |
| 20:37.24 | twisted[asteria] | oh |
| 20:37.26 | twisted[asteria] | n/m |
| 20:37.50 | JerJer | H.245 is a control protocol between two multimedia endpoints |
| 20:38.02 | *** join/#asterisk hound (tor@97c13b58ed5a9c1b.session.tor) |
| 20:38.12 | jsaunders | Any relation to dtmf? |
| 20:38.49 | jsaunders | Or rfc2833 for that matter? |
| 20:38.51 | xming | digital time multimedia format? |
| 20:38.57 | xming | :) |
| 20:39.00 | JerJer | You can send H.245 strings that could be interpeted as dtmf |
| 20:39.05 | jsaunders | Aha. |
| 20:39.12 | *** join/#asterisk anthm (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 20:39.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 20:39.14 | JerJer | a UserIndiction |
| 20:39.17 | JerJer | +a |
| 20:39.25 | jsaunders | So, rfc2833 is dtmf thru rtp stream, or you could pass dtmf thru H245 channel? |
| 20:39.53 | JerJer | rfc2833 sends the DTMF info as a special packet in the rtp stream, yes |
| 20:40.03 | jsaunders | Well ain't that some.... |
| 20:40.10 | JerJer | H.323 endpoints can do H.245 UserIndications |
| 20:40.14 | jsaunders | JerJer, you are too helpful. |
| 20:40.23 | JerJer | but you won't get H.245 on like SIP |
| 20:40.29 | xming | does anyone knows how to set TON=unknown and NPI=e164? I've hacked the code code to do that but I don't know if it is correct |
| 20:40.30 | jsaunders | Gotcha. |
| 20:42.18 | JerJer | (16:40:11) jsaunders: JerJer, you are too helpful. |
| 20:42.29 | JerJer | most people seem to think I bitch too much - at least lately |
| 20:42.47 | SkramX | ~jerjer |
| 20:42.47 | jbot | hmm... jerjer is the guy who runs nufone |
| 20:42.53 | tzanger | JerJer: you do bitch a lot |
| 20:42.56 | SkramX | ~nufone |
| 20:42.56 | jbot | well, nufone is Visit http://www.nufone.net for an excellent, native IAX termination service. |
| 20:43.09 | SkramX | wonder who added that one |
| 20:43.10 | tzanger | but we take it in stride since you hook us up with the phat beats |
| 20:43.16 | SkramX | ~sixtel |
| 20:43.28 | tzanger | SkramX: actually I think I added something along those lines. And no, I don't work for nufone |
| 20:43.38 | SkramX | o ok |
| 20:43.59 | *** part/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
| 20:44.02 | hardwire | blah and 1/2 |
| 20:44.59 | Beirdo | I'm sure JerJer hears enough bitching to drive any of us over the edge anyways. People are ingrates |
| 20:45.32 | *** join/#asterisk Lathos42 (~Lathos42@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 20:45.50 | JerJer | looks like Lathos42 made it to Illinois |
| 20:45.54 | JerJer | Beirdo: aye |
| 20:46.32 | Lathos42 | JerJer: Yep, finally got my rental car yesterday at 6pm EST |
| 20:46.34 | Beirdo | JerJer: just so you know, I'm a very happy customer :) |
| 20:46.37 | JerJer | and how many other players in here can take a half a million dollar loss and survive ? |
| 20:46.45 | JerJer | Lathos42: yay |
| 20:47.01 | bhima^ | ouch. How did you get the half million loss? |
| 20:47.18 | Beirdo | ouchie |
| 20:47.22 | tzanger | bhima^: do you have any idea what the 1-900 charges are for his particular tastes in girls?? |
| 20:47.26 | JerJer | http://voxilla.com/voxstory166.html |
| 20:47.27 | Beirdo | hang in there, you'll make it back |
| 20:47.37 | JerJer | Cost of doing business |
| 20:48.51 | Beirdo | sorry to hear it, JerJer |
| 20:48.59 | bhima^ | hmm. so you're basically not able to get a definite promise of rates for particular number patterns? |
| 20:49.33 | JerJer | our upstream changes their rates sometimes daily |
| 20:49.47 | JerJer | but only bothers to inform us every 2-3 weeks |
| 20:50.04 | JerJer | which was all it took |
| 20:50.09 | eKo1 | that sucks |
| 20:50.34 | JerJer | we weren't the only ones hit with that same scam, at the same time in fact |
| 20:50.58 | bhima^ | so why doesn't Verizon get hit with it too? |
| 20:51.07 | JerJer | now if you go look up the rates for the numbers that was being dialed everybody has a much higher specific rate, but that wasn't the case when this happened |
| 20:51.18 | JerJer | bhima^: i'm sure they do get hit |
| 20:51.22 | *** join/#asterisk valence (~valence@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 20:51.23 | bhima^ | (and, given their clout, force the upstream people to fix it...) |
| 20:51.54 | JerJer | VoIP, espcially open-source VoIP, simply facilitates the scam with much less costs involved |
| 20:52.44 | Beirdo | yeah :( |
| 20:53.17 | *** join/#asterisk ManxPower (~eric@ip-209-16-83-10.i-55.com) |
| 20:53.50 | JerJer | its just like the zombie modem dialer virii |
| 20:54.01 | JerJer | just more elegant |
| 20:54.06 | Beirdo | yup |
| 20:54.12 | sivana | JerJer: so now you're PayPal only and Domestic calling only? |
| 20:54.14 | JerJer | and deceitful |
| 20:54.16 | *** join/#asterisk SwK (~SwK@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 20:54.32 | Beirdo | better not be domestic only... unless Puerto Rico is considered domestic :)( |
| 20:54.34 | Beirdo | heh |
| 20:54.42 | mtgh | nufone no longer does intl calling anymore? |
| 20:54.58 | JerJer | sivana: US and Canada - we haven't cut off those older customers that we trust slightly more |
| 20:55.05 | sivana | I see, ok |
| 20:55.21 | Beirdo | can't blame you for that |
| 20:55.33 | xming | sorry to hera that jerejer |
| 20:55.33 | Beirdo | well, my calls to PR still work, so I'm happy :) |
| 20:55.48 | *** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
| 20:55.57 | JerJer | then we've unilaterally blocked those destinations that are stereotypically noticed for high fraud |
| 20:56.05 | Beirdo | good :) |
| 20:56.08 | TripleFFF2sdf | one knows if open h323 works well ? i got compile probs |
| 20:56.32 | Beirdo | you need to do what you need to do to not get repeatedly hosed. |
| 20:56.50 | TripleFFF2sdf | who is |
| 20:56.54 | TripleFFF2sdf | jerjer |
| 20:57.12 | TripleFFF2sdf | btw fruandster on voip providers |
| 20:57.17 | TripleFFF2sdf | hes a vietnam guy |
| 20:57.18 | xming | !Jerjer |
| 20:57.22 | xming | ~ |
| 20:57.25 | xming | ~Jerjer |
| 20:57.25 | jbot | well, jerjer is the guy who runs nufone |
| 20:57.47 | TripleFFF2sdf | he sells accounts to vietnam chicks and guys.. and creates them with stolen cards |
| 20:57.59 | TripleFFF2sdf | caled there |
| 20:58.19 | TripleFFF2sdf | basiclly internation assholes will make this business north american |
| 20:58.31 | *** part/#asterisk hound (tor@97c13b58ed5a9c1b.session.tor) |
| 20:58.34 | tzanger | jerjer's vietnamese? I don't think so |
| 20:58.44 | TripleFFF2sdf | i dont see why i would risk getting my bus loss for 1% of clientele thats does 995 of fraud |
| 20:58.47 | TripleFFF2sdf | no |
| 20:58.53 | TripleFFF2sdf | who said jerejr viet ? |
| 20:58.56 | TripleFFF2sdf | ah nevermind |
| 20:58.57 | *** part/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
| 20:59.03 | JerJer | di di maw |
| 20:59.20 | tzanger | "who is jerjer... he's a vietnam guy... he sells accounts to vietnam chicks and guys..." |
| 20:59.47 | Beirdo | heh |
| 21:00.03 | *** join/#asterisk pixolex (~chatzilla@85.138.107.157) |
| 21:00.08 | harryvv | high frad means no justice and law not doing anything about it. |
| 21:00.17 | harryvv | fraud |
| 21:00.34 | *** join/#asterisk Trionnis (lordkuri@12-215-249-177.client.mchsi.com) |
| 21:00.43 | Beirdo | and if you want the law to do something about it... you are asking for the FCC to control it more tightly |
| 21:00.51 | Beirdo | or CRTC in Canada |
| 21:00.53 | Trionnis | is anyone aware of authentication issues with sipmedia recently |
| 21:00.54 | Trionnis | ? |
| 21:00.57 | Beirdo | that comes with the territory |
| 21:01.11 | *** join/#asterisk dlotina (~root@200.29.14.69) |
| 21:01.24 | blitzrage | brookshire: you around? |
| 21:02.06 | Netgeeks | bah, high fraud means nothing about the law. It means the operators have not found it such an issue as to address it. |
| 21:02.28 | bhima^ | or it means that the system is too complicated to navigate properly and fix the fraud. |
| 21:03.08 | Netgeeks | The fraud that hit nufone could have been prevented if the carrier who connected ot the fraud termination immediately flagged rates outside the normal expected rates for the area and notified it's customers, who would then flag and notify thiers |
| 21:03.37 | Netgeeks | Nufone *could* have been notified withing minutes of the new route that it was a very high cost route and he could have blocked it |
| 21:03.40 | dlotina | does markster is online? |
| 21:03.46 | *** join/#asterisk T-Squared (~ted@hidden.serreyn.com) |
| 21:03.53 | Trionnis | paste is here: http://pastebin.ca/19215 I had been attempting to send a fax, it errored out a couple of times, then started doing that. No configs have been touched. |
| 21:03.59 | brookshire | mark is at cluecon |
| 21:04.21 | *** join/#asterisk T-Squared (~ted@hidden.serreyn.com) |
| 21:04.22 | Netgeeks | but it really isn't in the interest of the carriers to do any of that... since for them, there is cost associated with no profit in doing it |
| 21:04.46 | dlotina | mark told me find him on irc? he use markster as nick? |
| 21:04.57 | Trionnis | and I'll note that the fax is what had the error, not the phone. it connected ok, and started sending. |
| 21:04.59 | blitzrage | dlotina: mark <--> kram |
| 21:05.06 | dlotina | thanks blitzrage |
| 21:05.49 | riksta | are there nothing cheaper than the sangoma E1 cards available? |
| 21:05.59 | *** part/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com) |
| 21:06.11 | SwK | riksta: Digium has T1/E1 single port cards |
| 21:06.17 | SwK | and Quad Ports |
| 21:06.28 | *** join/#asterisk mago2-cn (~maxglucks@200.109.166.83) |
| 21:06.31 | SwK | or you should have gone to cluecon they just gave away 2 cards |
| 21:06.37 | SwK | and are giving away more tomorow |
| 21:06.44 | SwK | </darthclue> |
| 21:07.03 | blitzrage | SwK: lol |
| 21:07.11 | mago2-cn | Hi, I'd like your advice, can't decide whether or not to use alphanumerical values u=on sip.conf names. What are your thoughts? |
| 21:07.44 | *** join/#asterisk meppl (~mephisto@84.245.165.40) |
| 21:08.01 | riksta | SwK: the telecomes provider has said we can colo our asterisk box there, and he said this to me "we can provide the channels to you via E1 or "TCP/IP" ", now i dunno if he's used the right terminology there or he is getting confused...do you have any idea what he means by TCP/IP ? |
| 21:08.21 | JerJer | VoIP |
| 21:08.34 | Trionnis | I'd guess SIP, but possibly IAX |
| 21:08.42 | riksta | but he didnt even know what Asterisk was |
| 21:08.56 | Trionnis | there's much more than just Asterisk that does VoIP |
| 21:08.59 | eKo1 | maybe they meant TDMoIP |
| 21:09.07 | Trionnis | that's possible too :) |
| 21:09.21 | Trionnis | or h.323... |
| 21:10.05 | riksta | can asterisk accept whatever he meant (eg all forms) ? |
| 21:10.05 | SwK | riksta: well that ask him wnat TCPIP protocol |
| 21:10.08 | SwK | heh |
| 21:10.15 | riksta | SwK: i will but i have to wait till tomorrow |
| 21:10.26 | *** join/#asterisk fliplap (~rashid@63.133.150.3) |
| 21:11.26 | greg_work | mago2-cn: what's more likely to change: the extension a person is assigned, or the actual person at the company (ie ,fire/hire someone new) |
| 21:11.27 | JerJer | Its going to be SIP |
| 21:11.28 | h3x | SwK[work]: voip is usually udp :P |
| 21:11.30 | JerJer | perhaps H.323 |
| 21:11.38 | riksta | great |
| 21:11.44 | riksta | that saves me£300 on a E1 card |
| 21:11.44 | fliplap | does anyone have a suggestion on SIP gateway hardware? Right now we're using Mediatrix, who I haven't been all that happy with |
| 21:11.52 | JerJer | not great |
| 21:12.02 | JerJer | fliplap: Asterisk |
| 21:12.06 | riksta | why |
| 21:12.16 | eKo1 | sip gateway? |
| 21:12.35 | greg_work | mago2-cn: if you're more likely to reassign extensions, use the persons name as username, then its easy to reassign without messing with device config, if it's the person, use the extension and again, no messing with device configs :p |
| 21:12.43 | eKo1 | you mean pots<->sip gateway? |
| 21:12.51 | JerJer | 1U Server (your choice)+TE411P+Asterisk |
| 21:12.52 | fliplap | JerJer, we're looking for something dedicated to being a sip gateway |
| 21:12.59 | fliplap | so that we don't have to put a computer over there |
| 21:13.02 | fliplap | eKo1, nod |
| 21:13.38 | JerJer | ok 1U Server with CF drive |
| 21:13.42 | ManxPower | fliplap, all such devices with many ports are expensive. |
| 21:13.44 | eKo1 | well, all the gateways i've tried suck big donkey balls. |
| 21:13.47 | ManxPower | like EXPENSIVE |
| 21:13.48 | fliplap | ManxPower, nod |
| 21:13.49 | JerJer | no moving parts |
| 21:13.53 | Beirdo | lucky donkey, eKo1 |
| 21:14.02 | fliplap | we've got a bunch of 24 ports sitting here |
| 21:14.08 | *** join/#asterisk junbug (junya@adsl-065-013-044-139.sip.mia.bellsouth.net) |
| 21:14.25 | eKo1 | Beirdo: because they're being sucked or because they're big? |
| 21:14.26 | fliplap | i'm not one of those "money is no object" people, but for the most part, its not |
| 21:14.32 | Beirdo | both |
| 21:14.33 | Beirdo | heh |
| 21:15.03 | eKo1 | 90 % of my problems come from pots equipment. |
| 21:15.12 | fliplap | JerJer, cooling such a device becomes in issue in a hot room where most of the lines come into |
| 21:15.20 | eKo1 | I suggest you get a pri line. |
| 21:15.21 | *** part/#asterisk T-Squared (~ted@hidden.serreyn.com) |
| 21:16.03 | fliplap | i guess we can stick with mediatrix if there's not really any others out there |
| 21:16.36 | fliplap | eKo1, unfortunatly that isn't an option |
| 21:16.42 | greg_work | fliplap: depending on the load, you can get fanless mini-ITX systems that run pretty cool, and even the ones with fans aren't bad |
| 21:16.59 | fliplap | greg_work, the idea has been suggested and declined |
| 21:17.05 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 21:17.08 | fliplap | unfortunatly |
| 21:17.17 | eKo1 | why is that? |
| 21:17.40 | fliplap | eKo1, if i knew what was going on in my bosses head i'd be making a lot more money |
| 21:18.13 | greg_work | wellgate makes some gateways, not sure about them tho |
| 21:18.15 | JerJer | then buy a max TNT or 5400 |
| 21:18.18 | ManxPower | fliplap, why is PRI not an option? |
| 21:18.22 | fliplap | i'm guessing its the same reason that there aren't droves of people out to replace thier routers with linux machines |
| 21:18.23 | JerJer | but prepare to take it up the tail pipe |
| 21:18.29 | JerJer | in more ways than one |
| 21:18.32 | fliplap | ManxPower, part of it is cost |
| 21:18.44 | fliplap | and having to wait for the line |
| 21:18.48 | ManxPower | fliplap, Um, that's not usually the issue when you think about it. |
| 21:19.24 | ManxPower | We were able to double the number of users without increasing the number of trunks when we switched form analog to PRI |
| 21:19.30 | fliplap | we have several lines at out colo |
| 21:19.37 | ManxPower | Since we didn't have to dedicate specific lines to specific functions |
| 21:19.43 | fliplap | but we need the boxes at client sites |
| 21:20.20 | RomDump | Would a channel bank work? |
| 21:20.34 | fliplap | the mediatrix boxes work well, but they're kind of a pain to manage |
| 21:21.00 | fliplap | and really, a PRI line wouldn't serve the purpose anyway |
| 21:21.21 | eKo1 | pri + channel bank |
| 21:21.45 | fliplap | the idea is that all of the calls coming out of the site should be sip |
| 21:21.49 | sivana | does faxing work with a TDM400? |
| 21:22.10 | leandro_it | this is a great question, sivana! |
| 21:22.29 | sivana | I thought I remember someone saying there's issues with timing |
| 21:22.31 | ManxPower | sivana, not in my experience |
| 21:22.50 | greg_work | sivana: it apparently can, but from what I can tell it's not worth the time |
| 21:23.06 | sivana | I'm still having issues with multiple pages through a channel bank |
| 21:23.27 | fliplap | i guess dedicate sip gateway devices aren't really widely used? |
| 21:23.31 | leandro_it | me too, I try to disable echo cancel, but without success |
| 21:25.16 | leandro_it | does a solution for faxing from zap to zap interface exists? |
| 21:31.11 | *** join/#asterisk Cherebrum (jgarland@72.36.136.226) |
| 21:34.29 | *** join/#asterisk alexhopper (Alex@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 21:37.51 | leandro_it | ... |
| 21:38.08 | eKo1 | faxing is dead |
| 21:38.27 | sivana | sounds like it |
| 21:38.44 | eKo1 | and i hope pots will die soon also |
| 21:39.23 | leandro_it | You'll never get out of pots and faxes. There is always someone who need one. |
| 21:39.47 | leandro_it | ... and if there are no need for fax, there is need for analog modem connection |
| 21:39.48 | Assid | yeah.. too bad support of fax is pretty ok.. for voip based |
| 21:39.53 | kingtux | who are some good voip providers?? |
| 21:40.01 | kingtux | low price dids |
| 21:40.04 | kingtux | low rates |
| 21:40.14 | kingtux | i use telasip and voipbuster |
| 21:40.24 | kingtux | telasip is great |
| 21:40.25 | xheliox | I use Teliax. |
| 21:40.32 | Maarken | I use teliax as well. |
| 21:40.46 | kingtux | are the on iax2 protocol |
| 21:40.51 | xheliox | I can't say anythihg bad about them. |
| 21:40.52 | xheliox | Yes. |
| 21:41.03 | xheliox | They could have unlimited calling plans, but no one is perfect. |
| 21:41.22 | ManxPower | I also use Teliax |
| 21:41.43 | xheliox | ManxPower - that's good to know :) |
| 21:41.58 | ManxPower | If you want DIDs then Teliax is much better than Nufone, but if you just want a toll free number or just outgoing long distance, then they are both good. |
| 21:41.58 | kingtux | do they do sip |
| 21:42.04 | xheliox | yes |
| 21:42.05 | Maarken | yup |
| 21:42.20 | Maarken | you can pick iax or sip, specific your outbound CID, codecs, etc |
| 21:42.52 | ManxPower | (NuFone, as far as I know, only has Michigan DIDs) |
| 21:43.05 | Maarken | I mostly chose teliax because they have dids here. |
| 21:43.06 | Assid | incoming did |
| 21:43.17 | *** join/#asterisk Navman (~p_e@62.108.206.77) |
| 21:43.23 | Assid | yeah.. they do only michigan.. and toll free |
| 21:43.38 | xheliox | opposed to an OUTGOING direct INWARD dial? |
| 21:44.35 | Maarken | he's got you there. :) |
| 21:44.54 | ManxPower | There are hundreds of VoIP companies that offer outbound calling and toll free numbers |
| 21:45.04 | ManxPower | Not a lot of them provide DIDs. |
| 21:45.11 | *** join/#asterisk jpmcallister (~jpmcallis@kapla.escelsa.com.br) |
| 21:45.12 | hardwire | damnit |
| 21:45.17 | PhreeStyle | Can asterisk be used behind a firewall? |
| 21:45.18 | hardwire | I just ended up with a bunch of includes |
| 21:45.27 | Assid | PhreeStyle: sure |
| 21:45.51 | xheliox | It really depends on how restrictive of a firewall and what you're using Asterisk for. |
| 21:47.34 | hardwire | http://pastebin.ca/19218 |
| 21:47.45 | hardwire | does this method of laying out the config make any sense to anybody else? |
| 21:48.49 | hardwire | I suppose it woudl be hard to read like that |
| 21:49.08 | *** join/#asterisk asteriskmonkey (~phil@69.158.154.80) |
| 21:49.22 | asteriskmonkey | is there any digium staff on right now? |
| 21:49.48 | *** join/#asterisk jpmcallister (~jpmcallis@kapla.escelsa.com.br) |
| 21:50.28 | *** join/#asterisk jdg (~jdg@CA03F80F.adsl.mana.pf) |
| 21:50.59 | *** join/#asterisk Romik (~romik_@1.fix.netvision.net.il) |
| 21:51.18 | asteriskmonkey | can anyone help me with some dialing stuff? |
| 21:51.29 | *** join/#asterisk fa[20] (faceoff@devel.acdbddh.eu.org) |
| 21:51.35 | kingtux | how many simultaneous calls can you get with teliax |
| 21:51.35 | fa[20] | elou |
| 21:51.36 | kingtux | ?? |
| 21:51.43 | hardwire | kingtux: I can get as many as I want |
| 21:51.50 | hardwire | I have a server colocated there as well |
| 21:52.05 | *** part/#asterisk jdg (~jdg@CA03F80F.adsl.mana.pf) |
| 21:52.13 | Maarken | kingtux: https://www.teliax.com/newaccount/?r=1&cp=default |
| 21:52.19 | Maarken | that should answer most of your questions |
| 21:54.31 | ManxPower | asteriskmonkey, Digium will be happy to provide you fee based consulting services. Call or e-mail them. |
| 21:54.42 | *** join/#asterisk z00dax (~z00dax@kbsingh.plus.com) |
| 21:54.44 | *** part/#asterisk z00dax (~z00dax@kbsingh.plus.com) |
| 21:54.45 | ManxPower | asteriskmonkey, But they are not going to do it for free. |
| 21:55.00 | fa[20] | what is that? Aug 3 23:47:26 WARNING[4347]: chan_zap.c:6241 handle_init_event: Detected alarm on channel 1: Red Alarm |
| 21:55.11 | asteriskmonkey | its theere gear :P |
| 21:55.38 | asteriskmonkey | any reason why asterisk would jump from 63meg memeroy usage to nearly 500meg? |
| 21:55.47 | *** part/#asterisk mkrufky (~mk@68.160.103.77) |
| 21:56.16 | ManxPower | asteriskmonkey, Digium only provides HARDWARE support for their cards, not support for Asterisk |
| 21:56.27 | hardwire | unles you pay them |
| 21:56.30 | ManxPower | fa[20], The line was unplugged or went down. |
| 21:56.32 | asteriskmonkey | ah ok |
| 21:58.01 | Mavvie | ManxPower: well, they support initial zaptel driver support (http://www.digium.com/index.php?menu=service_category&category=support) |
| 21:58.22 | hardwire | I think I need to turn up the volume |
| 21:58.36 | asteriskmonkey | grrr damn digium toll free cant be called from canada |
| 21:58.40 | *** join/#asterisk doughecka (~Miranda@doughecka.user) |
| 21:58.40 | fa[20] | ManxPower But later I have larm cleared on channel 1 (.. to 19) |
| 21:58.43 | asteriskmonkey | any additional numbers anyone knows of? |
| 21:58.44 | fa[20] | so they get up? |
| 22:00.58 | fa[20] | ManxPower and zap show channels show that channels good |
| 22:02.14 | asteriskmonkey | ah for fudge sakes.. damn centos keeps running away and eating memory whats the best os for asterisk guys? |
| 22:02.23 | JerJer | um Linux |
| 22:02.32 | h3x | fedora core 2 |
| 22:02.33 | asteriskmonkey | what you think centos is "P |
| 22:02.37 | asteriskmonkey | ok cool |
| 22:02.42 | phil0u | 'lo. I have spent hours searching voip-info.org, google to find solution about FXO X10[01]P not being able to detect hangups, but still no luck :( would anyone know a definitive answer on that issue ? |
| 22:02.45 | fa[20] | yup |
| 22:02.45 | h3x | yes 2 |
| 22:02.45 | JerJer | slackware |
| 22:02.46 | h3x | not 4 |
| 22:02.46 | h3x | not 3 |
| 22:02.47 | h3x | 2 |
| 22:02.59 | Darwin35 | FBSD |
| 22:02.59 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 22:03.01 | h3x | its the most brainless and painfree installation |
| 22:03.04 | JerJer | or my own rollup of a kernel+busybox+glibc |
| 22:03.12 | h3x | glibc dosent get along with asterisk |
| 22:03.22 | h3x | i had a developer try to get that shit working for 3 months |
| 22:03.25 | h3x | it dosent fucking work |
| 22:03.32 | heison | i'm sorta stuck with AGI... |
| 22:03.48 | Maarken | h3x: uh, all linux is glibc |
| 22:03.50 | h3x | are you using zaptel hardware? |
| 22:03.56 | asteriskmonkey | yes |
| 22:04.02 | h3x | Errrr sorry my bad i was thinking uclibc |
| 22:04.02 | asteriskmonkey | digium stuff |
| 22:04.05 | h3x | i saw busybox and was |
| 22:04.15 | heison | i have in extensions.conf, exten=>_350[2|4|8],agi,baby_music.agi |
| 22:04.16 | Maarken | oh. yeah. |
| 22:04.20 | h3x | blah!@ |
| 22:04.22 | Maarken | uclibc is scary. |
| 22:04.23 | h3x | i havent eaten yet today |
| 22:04.36 | JerJer | its better now |
| 22:04.40 | JerJer | uclibc |
| 22:04.48 | h3x | uclibc is harder to port shit to than like, palm |
| 22:04.51 | JerJer | but yes, I do prefer glibc in my distro |
| 22:05.22 | Maarken | my server's are all BSD. asterisk is on the openbsd box. |
| 22:05.25 | h3x | are t1 cards working on bsd now |
| 22:05.27 | Darwin35 | I run asterisk on fbsd and it all works |
| 22:05.41 | Darwin35 | festival and sphinx and |
| 22:05.46 | h3x | nice |
| 22:05.49 | Darwin35 | loads more |
| 22:06.03 | h3x | did you have to patch it or is all that stuff in cvs now |
| 22:06.14 | mog_home | but it works better in linux darwin |
| 22:06.25 | Maarken | how so? |
| 22:06.26 | Darwin35 | look at the asterisk-bsd mailing list and the FreeBSD asterisk wikik page |
| 22:07.17 | *** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au) |
| 22:07.29 | mog_home | its written for linux |
| 22:07.31 | Darwin35 | mog I would ddebate on that but not going to get into a os war |
| 22:07.37 | Darwin35 | use the os you know best |
| 22:07.40 | mog_home | drivers are better |
| 22:07.41 | h3x | hah its in ports huh |
| 22:07.43 | mog_home | for linux |
| 22:07.45 | asteriskmonkey | ah shit t1 support no pri support in freebsd :( |
| 22:08.04 | Darwin35 | the drivers came from fbsd and where ported to linux and then ported back |
| 22:08.07 | heison | baby_music.agi takes the input extension and call via SIP to the 7960 on an autoanswer extension |
| 22:08.10 | mog_home | so if you have a real t1 no bsd |
| 22:08.19 | Darwin35 | not so |
| 22:08.20 | mog_home | if you look at zaptel drivers originally |
| 22:08.27 | mog_home | and look at digium maintained ones |
| 22:08.27 | Darwin35 | go read we have drivers on fbsd |
| 22:08.28 | h3x | well shit |
| 22:08.31 | mog_home | i think you would be amazed |
| 22:08.31 | heison | this is working fine... but i can't figure out the next step |
| 22:08.53 | asteriskmonkey | damn id take freebsd over fedora core 2 anyday way more secure |
| 22:08.54 | h3x | Well, somebody said on here they had a freebsd txxxp driver |
| 22:08.57 | h3x | for t1 cards |
| 22:09.14 | Darwin35 | yes they ar ein the ports |
| 22:09.19 | Maarken | asteriskmonkey: not to mention sexier. ;) |
| 22:09.21 | eKo1 | how about e1 cards |
| 22:09.29 | Darwin35 | the zaptel drivers are in the ports tree so lis libpri |
| 22:09.35 | heison | i want to make baby_music.agi a toggle - it checks whether there is currently music playing to the 7960, if so it performs a soft hangup or hangup; only plays music when there is no active channel on that extension |
| 22:09.36 | Darwin35 | go read the wiki page |
| 22:09.44 | SkramX | Im working on a call back program, going to be leeter that your mom! |
| 22:09.52 | heison | is there a way to show channels from AGI? |
| 22:10.10 | asteriskmonkey | http://www.voip-info.org/tiki-index.php?page=FreeBSD+zaptel |
| 22:10.15 | heison | i tried exec("show channels") but 'show' isn't really an application |
| 22:10.33 | *** join/#asterisk sedwards50 (~chatzilla@adsl-67-125-150-70.dsl.irvnca.pacbell.net) |
| 22:10.33 | Darwin35 | and the Asterisk+Freebsd project page |
| 22:10.53 | Darwin35 | back in 5 |
| 22:10.58 | Darwin35 | break |
| 22:11.15 | sedwards50 | anybody using chanspy |
| 22:11.20 | sedwards50 | ? |
| 22:11.24 | asteriskmonkey | how to i get a did to ring an iax client |
| 22:11.26 | *** join/#asterisk HellAgony (~HellAgony@200.121.192.238) |
| 22:11.46 | h3x | well that sucks, the drivers dont work in SMP mode |
| 22:12.05 | asteriskmonkey | fedora core 2 dosnt do proper smp anyhow either hehehe |
| 22:12.10 | sedwards50 | How about "dial(iax2/roadie@roadie)" |
| 22:12.16 | h3x | even after a yum update? |
| 22:12.19 | h3x | i never had any problems with it |
| 22:13.03 | asteriskmonkey | sedwards50: the syntax i dont get the dial (iax2/ip of server? @ username? ) is that right? |
| 22:13.38 | asteriskmonkey | i did a yum update and had to reload my zaptel drives in (That was in centos 3.5 thogh) |
| 22:13.55 | Maarken | can't you just do dial(iax/username)? |
| 22:14.10 | *** join/#asterisk tkoehler (~OCR-IRC@port-195-158-168-21.dynamic.qsc.de) |
| 22:14.17 | *** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net) |
| 22:14.27 | loud | h3x, they do work on an smp machine |
| 22:14.48 | h3x | loud not on freebsd |
| 22:14.56 | loud | ah, thats correct. |
| 22:15.07 | *** join/#asterisk asteriskmonkey (~phil@69.158.154.80) |
| 22:15.13 | asteriskmonkey | crap fell out of channel |
| 22:15.24 | *** part/#asterisk wrmem (~monnin@monnin-win.cso.uiuc.edu) |
| 22:15.33 | focks | anyone know why7960 firmware pos3-07-0-00 does not have a .sbn file? just .sb2 and .loads? |
| 22:15.45 | asteriskmonkey | so is it dial(iax2/ server address @ username) is that correct? |
| 22:16.23 | Sedorox | its Dial(IAX2/<user>:<pass(optional)>@<host>/<extention> |
| 22:16.51 | Sedorox | in mine.. I just used mercury@mercury.. so it sends user mercury.. with all the stored stuff under the iax2 [mercury] |
| 22:17.31 | eKo1 | i don't think you need the additional mercury |
| 22:17.45 | Sedorox | no.. I tried otherwise.. doesn't work very well.. had some problems with it |
| 22:17.50 | ManxPower | When you dial by ip address or hostname none of the sections of iax.conf or sip.conf will be used. |
| 22:17.50 | asteriskmonkey | so mercury in the second instance uses the store info under [mercury] |
| 22:17.59 | ManxPower | Therefore it's generally a BAD IDEA to dial by IP. |
| 22:18.25 | Sedorox | yes... its mercury(user)@mercury(host/entry in iax2.conf) |
| 22:18.28 | ManxPower | focks, Did you check the release notes for that version of the software |
| 22:18.53 | focks | ManxPower, i don't have them :( all this came from voipsupply.com. did something switch for that version? |
| 22:19.06 | *** join/#asterisk PhreeStyle (~PhreeStyl@cpe-24-221-52-165.az.sprintbbd.net) |
| 22:19.57 | PhreeStyle | my internet sucks. |
| 22:20.44 | Maarken | I got the upgrade for mine. worked pretty good. lots of new porn. |
| 22:20.52 | Darwin35 | ok back |
| 22:20.58 | asteriskmonkey | so i specify the dial command in the acutal extensions.conf right ? so it would be something like s,1, Dial(IAX2/<user>:<pass(optional)>@<host>/<extention> ? |
| 22:21.11 | JerJer | no pass - use the peer |
| 22:21.14 | PhreeStyle | I have set up asterisk on my powerbook, but it is behind an airport - I can not seem to make outbound calls... I used the sunrise stuff, and registered with the freeworld dialup... |
| 22:21.24 | JerJer | IAX2/user@peer/exten|timeout|options |
| 22:21.54 | asteriskmonkey | anyone got an example they can post on pastebin.ca for me |
| 22:22.02 | phil0u | focks use ethereal to find out what files are to be needed through TFTP |
| 22:22.10 | Sedorox | asteriskmonkey: you only need the password if you don't have the entry in iax2.conf |
| 22:22.14 | Sedorox | otherwise its just redundant |
| 22:22.17 | Darwin35 | look at your iax.conf |
| 22:22.29 | Darwin35 | the guestlogin is fine |
| 22:22.32 | focks | phil0u, I know which file is needed. P023-07-3-00.sbn, but I don't have that file |
| 22:22.40 | Darwin35 | clone it |
| 22:22.42 | phil0u | focks it helped me quite a bit to get here to 7.5 fw |
| 22:22.51 | asteriskmonkey | i have got my iaxy set up in the aix.conf file |
| 22:23.02 | asteriskmonkey | how do i get a phone to dial to it though where do i put that :P |
| 22:23.03 | JerJer | touch P023-07-3-00.sbn |
| 22:23.07 | JerJer | now you have that file |
| 22:23.18 | loud | 7.3 is way old, use 7.5 |
| 22:23.20 | asteriskmonkey | and damn it everytime i press a button when i pick up the iaxy phone i get a bust signal |
| 22:23.21 | phil0u | JerJer , well ;) |
| 22:23.29 | focks | loud, don't you have to go in order? |
| 22:23.32 | JerJer | its a true statement :) |
| 22:24.02 | focks | I'm coming from 6 |
| 22:24.06 | phil0u | focks depend from where to try to upgrade |
| 22:24.21 | phil0u | s/where to/where you/ |
| 22:24.24 | JerJer | i'll stick with the IAX firmware |
| 22:24.37 | loud | cco is down anyways, your cco account has been deleted. |
| 22:25.16 | PhreeStyle | ok, I see that I am supposed to register something in the iax.conf - but I know it is fairly lame but none of the information on the freeworld site looks like the stuff in the config file? |
| 22:25.43 | focks | phil0u, can I go from 6 - 7.3? |
| 22:25.44 | Sedorox | asteriskmonkey: you might have early dial on (if it has it...) but anyway... for a example... I have two servers.. lets say... A and B... if I wanna route all calls for PSTN through A.. I would do something like.... exten => _9.,1,Dial(IAX2/username@A/${EXTEN})... on B... |
| 22:25.56 | *** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca) |
| 22:25.57 | phil0u | focks yes |
| 22:26.01 | focks | sweet |
| 22:26.01 | TripleFFF2sdf | Expire : 377703 |
| 22:26.01 | TripleFFF2sdf | <PROTECTED> |
| 22:26.08 | TripleFFF2sdf | can one tell me signiuf of these 2 ? |
| 22:26.13 | TripleFFF2sdf | in sip show peer |
| 22:26.16 | *** join/#asterisk vtsherwood (user@cpe-24-210-53-246.columbus.res.rr.com) |
| 22:26.18 | TripleFFF2sdf | is that seconds ? |
| 22:26.22 | vtsherwood | greets all |
| 22:26.23 | TripleFFF2sdf | Expire 3777703 ? |
| 22:26.23 | phil0u | focks even to 7.5 directly i think |
| 22:26.37 | focks | i think that solves my problem because i have 7.3 sbn files |
| 22:26.58 | phil0u | focks good then :) |
| 22:27.09 | Darwin35 | all 25 of my x401's are online and working |
| 22:27.13 | focks | oh shit, bad-app header |
| 22:27.13 | sedwards50 | asteriskmonkey -- sorry, I got distracted. |
| 22:27.16 | vtsherwood | anyone else having problems with their users not being able to save their outgoing voicemail message? It saves the file but does not play it. using Asterisk RealTime |
| 22:27.19 | Darwin35 | just lack functions |
| 22:27.51 | Darwin35 | time to work on firmware src and change some things |
| 22:28.22 | vtsherwood | anybody at all? ;) |
| 22:28.55 | phil0u | focks is it SIP FW or anothe one doesn't remember exactly the cisco nomenclature |
| 22:29.03 | focks | SIP |
| 22:29.24 | phil0u | focks ok then there is no P023-07-3-00.sbn file in it |
| 22:29.45 | focks | Hmmhesays, what should I be using in my image_version var? |
| 22:30.01 | focks | damn nick complete. hmm what should I be using in my image_version var? |
| 22:30.26 | TripleFFF2sdf | ??? |
| 22:30.39 | phil0u | focks P0S3-07-3-00 should be ok |
| 22:30.43 | *** part/#asterisk vtsherwood (user@cpe-24-210-53-246.columbus.res.rr.com) |
| 22:30.49 | *** join/#asterisk Beccara (~Tristram@222-152-13-41.jetstream.xtra.co.nz) |
| 22:31.01 | *** part/#asterisk Navman (~p_e@62.108.206.77) |
| 22:31.04 | phil0u | or try P003-07-3-00 if it doesn't work |
| 22:31.07 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 22:31.26 | *** join/#asterisk darkskiez (~mhb@host-84-9-79-99.bulldogdsl.com) |
| 22:31.43 | darkskiez | meh, cisco's 7.5 Sip firmware seems to break inband progress :/ |
| 22:32.16 | phil0u | darkskiez what do you mean by "inband progress" ? |
| 22:32.20 | focks | phil0u, that's why I was doing originally, but I saw an error in solarwinds TFTP "requesting pos3-07-0-00.sbn file does not exist" |
| 22:32.50 | phil0u | fock not pos3-07-0-00.sbn but p0s3-07-0-00.sbn |
| 22:33.21 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E868.versanet.de) |
| 22:33.22 | darkskiez | phil0u: maybe i'm using the wrong term.. But when you hear the phone networks ringing etc, and not a generated one. with 7.5 the phone overlays its own ringing over the inband one, so you hear two ringing's |
| 22:33.24 | Nebukadneza | hi guys |
| 22:33.25 | focks | either way, there is no .sbn file |
| 22:33.44 | Nebukadneza | i got more or less a emergency ...is there a way to schedule commands in asterisk (like with cron?) |
| 22:34.14 | phil0u | focks and BTW, all files should be in capital, well i mean be sure to respect the case |
| 22:34.30 | *** join/#asterisk JunK-Y (~foobar@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 22:34.36 | darkskiez | phil0u: the .sbn is lower case tho |
| 22:35.00 | phil0u | darkskiez yes true |
| 22:35.15 | focks | should there exist a file called P0S2-07-0-00.sbn? |
| 22:35.21 | darkskiez | no |
| 22:35.25 | darkskiez | you made that p |
| 22:35.35 | focks | darkskiez, then why is the phone asking for it? |
| 22:35.40 | darkskiez | its not |
| 22:35.57 | darkskiez | you typed it into a config file like that |
| 22:35.58 | phil0u | darkskiez funny, i never noticed that ring problem :) |
| 22:36.07 | darkskiez | phil0u: in 7.5 ? |
| 22:36.14 | focks | darkskiez, i put P0S3-07-0-00 into the config file |
| 22:36.20 | phil0u | darkskiez yes |
| 22:36.28 | darkskiez | phil0u: calling out a pri ? |
| 22:36.37 | darkskiez | phil0u: never noticed? have you tried ? |
| 22:36.52 | focks | and then the phone requests P0S3-07-0-00.sbn from my TFTP server which replies not today buddy |
| 22:36.56 | phil0u | darkskiez only using FXO on POTS |
| 22:37.20 | *** join/#asterisk Dalfry (~DalFry@gateway.ishisystems.com) |
| 22:37.26 | darkskiez | focks: you do need the file |
| 22:37.49 | focks | darkskiez, so i do need it. do you happen to have it? |
| 22:37.52 | Nebukadneza | no ideas how schedule a command (//call) |
| 22:37.58 | phil0u | focks do you have the "P0S3-07-0-00.sbn" file ? ( from FW 7.0 right ?) |
| 22:38.05 | Dalfry | hello, can someone help me with asterisk@home please? I have a few custom config changes to be done |
| 22:38.08 | focks | phil0u, no |
| 22:38.10 | darkskiez | Nebukadneza: what type of command ? |
| 22:38.21 | focks | just .sb2 and .loads |
| 22:38.24 | Nebukadneza | darkskiez: Dial for example |
| 22:38.33 | phil0u | focks then you need it :) Please contact your closest Cisco Reseller |
| 22:38.35 | darkskiez | Nebukadneza: if its a CLI command you can just put asterisk -rx 'sip reload' into your crontab |
| 22:38.46 | focks | phil0u, that's what I've been saying all along! |
| 22:38.49 | Nebukadneza | hm ... i dont think it is |
| 22:38.52 | Nebukadneza | i cannot dial on the cli, right? |
| 22:39.07 | darkskiez | Nebukadneza: depends if you have a chan_alsa/oss loaded |
| 22:39.28 | darkskiez | Nebukadneza: so, what are you wanting to do at a particular time ? |
| 22:39.34 | darkskiez | more specifically |
| 22:39.55 | Nebukadneza | Dial out on a isdn card and play a sound |
| 22:40.04 | Nebukadneza | just those 2 commands ... dial and background |
| 22:40.22 | darkskiez | right... |
| 22:40.33 | darkskiez | are you going to phone lots of people with annoying messages? |
| 22:40.37 | Nebukadneza | no |
| 22:40.49 | darkskiez | look up wake up call on the wiki |
| 22:40.49 | Nebukadneza | i am going to wake myself and a friend up to catch a train :P |
| 22:40.56 | Nebukadneza | hm ... okay |
| 22:40.58 | Nebukadneza | no other way? |
| 22:41.06 | Nebukadneza | (dont have time to install all that perl stuff) |
| 22:41.15 | darkskiez | you have perl installed already |
| 22:41.25 | asteriskmonkey | anyone else here with a pri? |
| 22:41.30 | Nebukadneza | i need to get it running in the next 10 minutes ... (and google got nothing else except that wakeup) |
| 22:41.31 | darkskiez | asteriskmonkey: yes |
| 22:41.40 | Nebukadneza | darkskiez: hrm ... not the agi module thing |
| 22:41.40 | Nebukadneza | y |
| 22:41.41 | Nebukadneza | okay |
| 22:41.43 | Nebukadneza | lets let it be |
| 22:41.45 | Nebukadneza | thanks anyway |
| 22:41.56 | Nebukadneza | need sleep or else no chance to get that train |
| 22:41.58 | Nebukadneza | but thanks |
| 22:41.58 | Nebukadneza | gn8 |
| 22:42.20 | asteriskmonkey | darkskiez: does it take about 10mins after a rooboot for it to come fully back online to asterisk? |
| 22:43.01 | darkskiez | ast_freak: no, its instant |
| 22:43.11 | asteriskmonkey | what os are you running? |
| 22:43.23 | darkskiez | asteriskmonkey: debian/x86 |
| 22:43.39 | phil0u | darkskiez good choice ;) |
| 22:43.54 | Maarken | are grandstream 101s any good? |
| 22:44.14 | darkskiez | any good at what? |
| 22:44.26 | asteriskmonkey | darksckiez : what version of debian should i download |
| 22:44.31 | darkskiez | asteriskmonkey: ?fully come back? |
| 22:44.59 | darkskiez | is it the OS boot up time you are having problems with ? |
| 22:45.08 | asteriskmonkey | no |
| 22:45.20 | darkskiez | the distro will have little effect on anything else |
| 22:45.20 | *** join/#asterisk RomDump (romdump@otaku.freeshell.ORG) |
| 22:45.31 | darkskiez | what do u mean by fully ocome back.. |
| 22:45.35 | asteriskmonkey | im using centos what happens is it takes about a good 5-10mins before its stops ringing busy and allowing ccalls in |
| 22:45.56 | darkskiez | anything in the logs? |
| 22:46.03 | asteriskmonkey | like i see the messages for about 5-10min about d channel going up and down |
| 22:46.07 | focks | darkskiez, it's not simply a matter of specifying P0S3-07-0-00.sb2 instead of P0S3-07-0-00 is it? i do have P0S3-07-0-00.sb2 |
| 22:46.10 | darkskiez | might be a problem with the phone network |
| 22:46.27 | darkskiez | focks: you need .sb2 .sbn and .loads files |
| 22:46.31 | asteriskmonkey | coulde it be my asterisk settings.. set it as a slave instead of master? |
| 22:46.43 | focks | darkskiez, damn, they shorted me... |
| 22:46.53 | focks | 1 friggin file |
| 22:47.20 | darkskiez | focks: .sbn is the universal application loader, .sb2 is the application image |
| 22:47.52 | focks | darkskiez, i see. and you don't have that file you can lend me do you? |
| 22:48.19 | darkskiez | asteriskmonkey: what linetype ? |
| 22:48.56 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:49.17 | darkskiez | pastebin your zapata and zaptel configs |
| 22:49.18 | asteriskmonkey | ni2 |
| 22:49.24 | asteriskmonkey | will do |
| 22:50.49 | darkskiez | i'm not familiar with ni2 |
| 22:50.56 | phil0u | darkskiez are you aware of problems about X100P/X101P FXO card not detecting hangups ? i'm currently getting mad at this and can't find any definitive answer yet |
| 22:51.28 | asteriskmonkey | darksiez: http://pastebin.ca/19226 |
| 22:51.32 | asteriskmonkey | national type 2 |
| 22:51.39 | asteriskmonkey | full pri basically |
| 22:51.40 | eKo1 | phil0u: the definitive answer is pots sucks |
| 22:51.49 | Maarken | heh |
| 22:51.51 | Maarken | so true |
| 22:51.58 | JerJer | i would bet money you are running a clone X100P |
| 22:52.32 | h3x | how would that cause his problem |
| 22:52.40 | darkskiez | phil0u: if you called them? it wont, the phoneline doesnt know they've hungup. |
| 22:52.45 | JerJer | cuz clones are simply driver compatible |
| 22:52.58 | darkskiez | asteriskmonkey: I have a euroisdn |
| 22:52.59 | h3x | so, its the same damn card |
| 22:53.05 | JerJer | no it is not |
| 22:53.11 | *** join/#asterisk anthm (~anthm@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 22:53.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:53.15 | phil0u | JerJer yes i'm using DigitNetworks X101P clone |
| 22:53.27 | JerJer | i have proven time and time again that the clone X100Ps are very inferior |
| 22:53.36 | h3x | its the same god damn card! |
| 22:53.36 | JerJer | phil0u: then call them for support |
| 22:53.41 | JerJer | no it is not |
| 22:53.52 | JerJer | it is a very general driver |
| 22:54.08 | JerJer | its not even the same chipset - so how can it be the same damn card? |
| 22:54.14 | phil0u | JerJer: So you would think it's because of this ? |
| 22:54.30 | JerJer | motorolla hasn't produced the chipset mark chose for the X100P in quite a while |
| 22:54.30 | mog_home | not only that |
| 22:54.34 | JerJer | so it is not the same card |
| 22:54.34 | h3x | well its the same as the x100p that digium used to sell |
| 22:54.39 | mog_home | digium doesnt even use it anymore |
| 22:54.39 | JerJer | no it is not |
| 22:54.41 | h3x | we weren't talking about x101 |
| 22:54.47 | JerJer | it is simply driver compatible |
| 22:54.48 | mog_home | jerjer is right |
| 22:54.54 | mog_home | and digium modified the card slightly |
| 22:54.55 | JerJer | phil0u: most certainly |
| 22:55.01 | JerJer | mog_home: no mods |
| 22:55.01 | h3x | adding resistors dosent change the board any |
| 22:55.06 | phil0u | JerJer sigh :( |
| 22:55.16 | JerJer | just selecting the absolutely proper chipset |
| 22:55.23 | JerJer | for very specific reasons |
| 22:55.25 | *** join/#asterisk |HAL9000| (hal9000@bsd.miki.eu.org) |
| 22:55.37 | SkramX | i have a perl script, it gets 2 (and later to be three) inputs [via chomp()]... now how do i make it runable from the web? |
| 22:55.40 | SkramX | oops |
| 22:55.42 | SkramX | wrong place |
| 22:55.48 | SkramX | does anyone know tho? |
| 22:55.51 | JerJer | rm -rf <perlscript> |
| 22:56.01 | phil0u | but busydetect should at least work , no ? |
| 22:56.04 | Maarken | SkramX: change how it works and use CGI. :) |
| 22:56.07 | ManxPower | JerJer, I thought that was: rm -rf / |
| 22:56.12 | JerJer | busydetect is even worse |
| 22:56.13 | SkramX | how? |
| 22:56.14 | SkramX | heh |
| 22:56.20 | Sedorox | [18:53] <phil0u> JerJer yes i'm using DigitNetworks X101P clone |
| 22:56.28 | Sedorox | that motorola chipset thats on them.. from ebay |
| 22:56.33 | Sedorox | I've found to be total shit.. |
| 22:56.39 | Sedorox | only works if you use the drivers they send |
| 22:56.44 | Sedorox | and doesn't work well at all |
| 22:56.54 | h3x | oh they are using motorola eh |
| 22:56.55 | Sedorox | someone else had a problem like that about a month ago... |
| 22:57.01 | Sedorox | yea.. but its some off the wall thing |
| 22:57.01 | ManxPower | Sedorox, is it an IA92 device or something else. |
| 22:57.10 | JerJer | because it is an inferior, driver compatible device |
| 22:57.10 | Sedorox | I'm not sure.... |
| 22:57.17 | h3x | i see, well all the ones i seen/used were ia92s |
| 22:57.31 | JerJer | which mark never tried to hide or limit |
| 22:57.36 | ManxPower | h3x, Me too. They worked OK for me. Well, as well as any analog thing does. |
| 22:57.37 | Sedorox | all I know is the other dude's would work unless he used the included drivers.. and it had all the battery stuff removed.. and it would also be no alarmed |
| 22:57.56 | Sedorox | wouldn't work* |
| 22:57.58 | phil0u | Sedorox hmm well , it's true that DigitNetworks provide drivers, but i though it would just be the same * drivers but just repackaged |
| 22:58.08 | JerJer | no |
| 22:58.14 | h3x | well i guess you should try them then |
| 22:58.17 | ManxPower | Ya know that Intel doesn't produce the original X101P cards anymore right? |
| 22:58.27 | Sedorox | they are 'the same' in the fact it is the zaptel.. but heavly modified |
| 22:58.29 | JerJer | DigitNetworks is a hack - just trying to make money because they sound like some other company |
| 22:58.37 | ManxPower | I'll bet they are getting harder and harder to find now. |
| 22:58.42 | phil0u | h3x sure i will, never could have thoughy it would/could make a difference :( |
| 22:58.56 | Sedorox | I've never heard of the X101P untill I saw the clones pop out of no where on ebay |
| 22:59.05 | h3x | those boards in any form suck anyway, get a voip fxo device |
| 22:59.09 | h3x | like a sipura or something |
| 22:59.15 | Sedorox | hehe |
| 22:59.16 | ManxPower | I have like 3 original X101P Digium cards sitting in a drawer, as well as 2 or 3 "clone" cards. |
| 22:59.30 | Sedorox | I have a X100P clone.. works just fine... |
| 22:59.35 | h3x | ive had real digium x10x boards lock up machines |
| 22:59.38 | ManxPower | Sedorox, Digium stopped selling them like 6 months ago or more. |
| 22:59.43 | h3x | after they run for a month or two |
| 22:59.45 | Sedorox | yea.. |
| 22:59.52 | h3x | the system freezes or the board just needs a hard reset |
| 23:00.37 | JerJer | i still have dozens of real X100P based systems in production |
| 23:00.53 | JerJer | and 99% of them I haven't so much as thought about in many months |
| 23:01.00 | h3x | zaptel timing? heh |
| 23:01.08 | JerJer | 911 ports |
| 23:01.13 | h3x | oh |
| 23:01.16 | JerJer | fail-over if net dies |
| 23:01.37 | phil0u | well, anyway here in France, national telco only seems to send busy tone to tell calling party has hung up |
| 23:01.40 | *** join/#asterisk Nix (~Nix@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 23:01.58 | JerJer | load the france stuff in zaptel |
| 23:01.59 | h3x | haha you didnt mention your country |
| 23:02.10 | h3x | that would help |
| 23:02.17 | phil0u | i haven't been able to find anything about disconnect supervision and France Telecom :) |
| 23:02.30 | phil0u | well sure i could have started with that :) |
| 23:02.58 | phil0u | but still, JerJer of course i provided the right settings :) in zaptel.conf |
| 23:03.21 | phil0u | even read the source :) like luke said :) insmod wcfxo opermode=1 |
| 23:04.56 | twisted[asteria] | psychobilly owns me |
| 23:07.49 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:10.05 | JerJer | all your base are belong to us |
| 23:10.24 | Maarken | wow. meme flashback |
| 23:12.59 | TripleFFF2sdf | lol |
| 23:13.02 | TripleFFF2sdf | oui c vrai |
| 23:13.26 | *** join/#asterisk PhreeStyle (~PhreeStyl@cpe-24-221-52-165.az.sprintbbd.net) |
| 23:13.36 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 23:13.41 | Damin | Hey guys.. |
| 23:13.55 | |HAL9000| | names |
| 23:14.45 | Damin | Russel? |
| 23:14.46 | Darwin35 | wich driver |
| 23:15.00 | JerJer | phil0u: sure it can be |
| 23:15.31 | phil0u | JerJer hehe well not worse regarding hangups detection :) |
| 23:15.46 | *** join/#asterisk colde (~colde@colde.active.supporter.pdpc) |
| 23:15.55 | phil0u | Darwin35 was asking me ? |
| 23:16.12 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-68-76-249-238.dsl.chcgil.ameritech.net) |
| 23:17.15 | phil0u | Darwin35 DigitNetworks provide some kind of tarball with their clones , unfortunately, it doesn't even compile , geez |
| 23:17.37 | Damin | Drumkilla: |
| 23:18.35 | phil0u | so looks like i'm out of luck for hangups detection with FXO X101P clones :( |
| 23:19.36 | darkskiez | i read that as hummus detection |
| 23:19.39 | darkskiez | mmm hummus |
| 23:19.40 | darkskiez | its late |
| 23:19.43 | darkskiez | night |
| 23:19.43 | *** part/#asterisk darkskiez (~mhb@host-84-9-79-99.bulldogdsl.com) |
| 23:19.59 | shido6 | Darwin35 are you a darwin guru? |
| 23:20.08 | Maarken | apparently he's got pretty good hummus detection |
| 23:20.11 | Netgeeks | twisted still here? |
| 23:20.28 | Darwin35 | twisted is never really here. |
| 23:20.39 | JerJer | define 'here' |
| 23:20.41 | Darwin35 | just to make sure you understand your request |
| 23:20.57 | Netgeeks | let me try again |
| 23:21.01 | shido6 | Darwin35 are you a darwin guru? |
| 23:21.19 | Netgeeks | Twisted: are you reading this screen and able to comprehend the words I am typing? |
| 23:21.25 | Darwin35 | no I meesed with it . but I got the nick from SeaQuest |
| 23:21.31 | shido6 | oh! |
| 23:21.32 | shido6 | heh |
| 23:21.34 | Darwin35 | I was a seaquest adict |
| 23:21.42 | *** part/#asterisk |HAL9000| (hal9000@bsd.miki.eu.org) |
| 23:21.47 | Darwin35 | the dolphin rocked |
| 23:21.59 | Darwin35 | his name was Darwin |
| 23:23.17 | Darwin35 | and 35 was my age last year |
| 23:23.24 | Darwin35 | I should change to darwin36 |
| 23:23.58 | Ariel_ | wow how did they every get that recorded. Ohh baby come to cluecon..... |
| 23:24.44 | Darwin35 | they paid her to |
| 23:24.51 | Darwin35 | she works for digium |
| 23:25.11 | Darwin35 | you pay she will record just about anything |
| 23:25.17 | Ariel_ | Darwin35, yes I understand that. But wow just so strange you would pay for it. |
| 23:25.45 | Ariel_ | Darwin35, she does not directly work for digium last I knew. |
| 23:25.58 | *** join/#asterisk pifiu (~myassisbi@208.205.181.170) |
| 23:26.05 | mog_home | she works with us |
| 23:26.12 | mog_home | not exactly for us |
| 23:26.21 | mog_home | she does a lot of work for other people |
| 23:26.28 | mog_home | she is the voice for ipod support at apple |
| 23:26.33 | mog_home | and other random things |
| 23:27.00 | TripleFFF2sdf | o wrapendpoint.o |
| 23:27.06 | TripleFFF2sdf | anyone else go t a prob b on this |
| 23:27.16 | TripleFFF2sdf | rapendpoint.cxx:852: no matching function for call to `H323AudioCodec:: |
| 23:28.06 | JerJer | mog_home: and like a lot of Canadian radio commercials |
| 23:28.20 | JerJer | and like airport announcements |
| 23:29.22 | Darwin35 | hell pay enough I bet JerJer will even put his voice to your prompts |
| 23:29.27 | fugitivo | apple released a mouse with 2 buttons! |
| 23:29.37 | TripleFFF2sdf | wow |
| 23:29.40 | TripleFFF2sdf | serious ? |
| 23:29.43 | denon | fugitivo: arent they afraid it'll confuse the users? |
| 23:29.43 | TripleFFF2sdf | man |
| 23:29.48 | Maarken | 4 buttons, actually |
| 23:29.54 | Maarken | and a 4-way scrollball |
| 23:29.56 | fugitivo | it's called "Mighty mouse" |
| 23:29.57 | TripleFFF2sdf | i think in the next century they will take the giant leap and add the wheel on it.. |
| 23:30.02 | mog_home | its like an ipod mouse |
| 23:30.07 | *** join/#asterisk PhreeStyle (~PhreeStyl@cpe-24-221-52-165.az.sprintbbd.net) |
| 23:30.12 | fugitivo | TripleFFF2sdf: it comes with a 360 wheel! |
| 23:30.23 | TripleFFF2sdf | wel mac come with a return policy |
| 23:30.24 | TripleFFF2sdf | ;) |
| 23:30.31 | TripleFFF2sdf | that all i need |
| 23:30.37 | TripleFFF2sdf | 10 days to give the shit back |
| 23:31.02 | brookshire | i hear the "mighty mouse" is lame |
| 23:34.14 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
| 23:34.24 | asteriskmonkey | woo! finally got routing to work on the damn iaxy clients.. |
| 23:34.51 | shido6 | this is such a pain |
| 23:35.03 | Netgeeks | what's that shido? |
| 23:35.10 | cypro-cluecon | x/w 3 |
| 23:35.23 | shido6 | can someone with a real mac tar and gz their /etc directory and dcc it to me? |
| 23:35.38 | Chereb-cluecon | cypro-cluecon : hey |
| 23:35.54 | shido6 | Netgeeks, I'm installing mac os x on my intel |
| 23:35.59 | *** join/#asterisk shadeboboo (~shadebob@212.217.71.121) |
| 23:36.00 | shadeboboo | hi |
| 23:36.01 | Netgeeks | ah |
| 23:36.04 | cypro-cluecon | :) |
| 23:36.47 | Chereb-cluecon | cypro-cluecon : are you down babysitting the keg? |
| 23:37.22 | mago2-cn | hi, would anyone be so kind and tell me how to modify the callerid number with something like setcallerid? |
| 23:37.38 | shadeboboo | I have a problem to integrate asterisk with a siemens Hicom200. Scheme is : E1 <---->Asterisk<----E1--->Hicom 200. When an outgoing call comme to asterisk a native bridge is ok with the Hicom. But when I want to initialize a call from the hicom |
| 23:37.39 | Chereb-cluecon | shido6 : I thought the OSX for x86 had some sort of DRM |
| 23:38.04 | shido6 | well we're going to find out now arent we :) |
| 23:38.12 | shadeboboo | to the asterisk CLI show : Extension '' in context 'vers_pbx' from '138' does not exist. Rejecting call on channel 0/31, span 2 |
| 23:38.12 | shido6 | first you install darwin |
| 23:38.13 | mago2-cn | found it, thx |
| 23:38.13 | Chereb-cluecon | heh heh |
| 23:38.15 | shido6 | then the upgrade |
| 23:38.21 | Chereb-cluecon | shido6: aha |
| 23:38.36 | shido6 | then you unrar three huge 1 gig files ( collectively) to the partition |
| 23:38.43 | shido6 | and reboot and pray |
| 23:38.48 | shadeboboo | it's seem that's asterisk don't understand call initializing from the Hicom... Someone can help me? |
| 23:39.03 | shido6 | i blew away 120 gig drive just to try it |
| 23:39.31 | shido6 | all i need to survive is my paypal account, my gmail and a my /etc/asterisk/*.conf 's |
| 23:39.39 | shido6 | and a iaxy |
| 23:39.46 | shido6 | +n |
| 23:39.58 | asteriskmonkey | :D iaxy's kick ass |
| 23:40.06 | Sedorox | lol |
| 23:40.06 | shido6 | my pap2 is cooler |
| 23:40.11 | shido6 | but it wont fit in my pocket |
| 23:40.20 | shido6 | so fsck it |
| 23:40.21 | Sedorox | shido6: if it works... let me know.. :p |
| 23:40.25 | shido6 | i will |
| 23:40.27 | Maarken | iaxy's are more expensive than the single FXS SIP devices. :| |
| 23:40.28 | asteriskmonkey | i like how the iaxy fits in the pocket but not the power adapter |
| 23:40.33 | Maarken | sadly, money talks. |
| 23:40.37 | hardwire | ok |
| 23:40.39 | hardwire | I have echo again |
| 23:40.48 | hardwire | thanks to the people at my telco fucking with the pri again |
| 23:40.54 | hardwire | their 50 other dms100 pri don't have echo |
| 23:40.55 | Sedorox | I would really like to run it on my laptop.. but I heard it needs SSE3 and the DRM stuff |
| 23:40.57 | hardwire | and I shouldn't be unique |
| 23:41.04 | hardwire | I call.. I get echo |
| 23:41.09 | hardwire | I used the monitor plugin |
| 23:41.18 | hardwire | and the monitor plugin does not show the echo |
| 23:41.21 | hardwire | thats pissing me off |
| 23:41.25 | hardwire | if course its not however |
| 23:41.28 | mago2-cn | Please tell me if correct: will this prefix 101 to the caller id number while preserving caller id name? |
| 23:41.30 | mago2-cn | exten => _100XXXX,1,SetCIDNum(cname[101${CALLERID}|a]) |
| 23:41.33 | hardwire | I shouldn't have used merged |
| 23:41.34 | Nugget | I run OS X on my laptop :P |
| 23:41.53 | Sedorox | Nugget: bah... not a apple laptop :p |
| 23:42.08 | JerJer | mago2-cn: wow i spoze - but why? |
| 23:42.20 | Sedorox | hehe, I would want to do that |
| 23:42.45 | mago2-cn | i'm trying to call between same extension numbers, like 1001 but in different contexts |
| 23:47.04 | mago2-cn | exten => _1011XXX,1,Goto(contextb,${EXTEN:3},1) |
| 23:47.06 | mago2-cn | works |
| 23:47.35 | mago2-cn | tandem switching |
| 23:53.02 | shido6 | its building darwin 8.1 |
| 23:53.03 | shido6 | finally |
| 23:53.55 | shido6 | http://grabberslasher.no-ip.com/macosx/ |
| 23:55.18 | Maarken | if it won't run a gui, what's the point? |
| 23:55.27 | Maarken | osx without the gui is darwin. |
| 23:55.41 | ManxPower | I thought it was freebsd 8-) |
| 23:56.03 | Sedorox | darwin is freebsd |
| 23:56.06 | Sedorox | well.. based on... |
| 23:56.06 | Nugget | no. |
| 23:56.09 | Nugget | darwin is mach. |
| 23:56.13 | Maarken | well, darwin's userspace is mostly freebsd |
| 23:56.19 | mago2-cn | how do i refer to callerid but just the number, as calleridname? |
| 23:56.19 | Maarken | the kernel is mach |
| 23:56.24 | Sedorox | eh |
| 23:56.28 | Maarken | calleridnum |
| 23:56.49 | mago2-cn | thanks! |
| 23:57.05 | Nugget | in RMS-speak, it would be FreeBSD/Darwin I suppose. :) |
| 23:57.26 | Nugget | but I think that places altogether too much importance on the userland tools (in both cases) |
| 23:57.51 | Maarken | RMS needs more valium in his diet. |
| 23:57.56 | Nugget | indeed |
| 23:57.59 | *** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net) |
| 23:58.14 | Nugget | he'd probably be well-served by having to work for a living, too. |
| 23:58.29 | Maarken | that or bathing would kill him. |
| 23:58.47 | stormfr | hello, anybody already see a problem with asterisk opening 50k inodes in one minute ? |
| 23:58.49 | bhima^ | When he was coding, he really did kick ass, from what I've read. |
| 23:58.49 | Maarken | damn stinky hippies. |
| 23:59.14 | bhima^ | Maybe I was lucky, but I didn't notice any odors. :) |
| 23:59.51 | *** join/#asterisk NormAst (~NormAst@toronto-HSE-ppp3974050.sympatico.ca) |