| 00:06.18 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 00:07.47 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 00:15.06 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 00:23.04 | *** join/#asterisk Micc (n=Micc@c-98-225-57-96.hsd1.wa.comcast.net) |
| 00:25.42 | Tech_Travis | Katty: hi back atcha |
| 00:27.39 | jblack | wazza world |
| 00:28.27 | heliosj | Howdy howdy. |
| 00:28.34 | Micc | I used to be able to transfer from my cell phone, but I don't know what changed, now I just get an error that says "features.c: Did not read data." |
| 00:29.19 | Micc | How do I know what context it is in? Is the transfer in the context of the Dial command or the context of the SIP trunk used to dial the cell phone? |
| 00:29.25 | Micc | Or something else entirely? |
| 00:30.09 | jblack | Micc: Not enough information |
| 00:30.34 | jblack | turn on debug 9, verbose 9, capture that output frm the start of a call to the failed transfer attempt, and pastebin that. |
| 00:31.02 | jblack | be ready to pastebin your extensions.conf and also a sip.conf/iax.conf with passwords replaced with XXXXXXXX |
| 00:31.39 | Micc | jblack, this should be a simple question. What context is used when features.c is transfering with the # key? |
| 00:31.56 | jblack | current, I believe |
| 00:31.57 | *** join/#asterisk doubletoker (n=doubleto@70.153.193.240) |
| 00:32.17 | Micc | jblack, I know there is also a variable that gets set BLINDXFER or something. |
| 00:32.43 | Micc | But that gets set after the call. |
| 00:33.26 | jblack | I think I've answered your question |
| 00:40.16 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:41.04 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 00:43.41 | doubletoker | what is the difference between being un/marked in meetme |
| 00:46.18 | p3nguin | How do I force a client/peer to re-register right now instead of when it has scheduled to do so, which would be at some time in the future? |
| 00:50.31 | Micc | debug and verbose 9 don't give any additional information, just features.c: Did not read data. after I press # then any other key. |
| 00:50.43 | Micc | It doesn't seem like a context issue now. |
| 00:50.57 | Micc | it says the transfer sound too. |
| 00:55.50 | jblack | maybe it's that short timeout |
| 00:56.07 | jblack | features by default has a amazingly short timeout |
| 01:00.30 | jblack | p3nguin: sip reload generally forces a reregister for all sip clients. |
| 01:01.45 | Micc | jblack, setting back to defaults in features.conf fixed it. I had # as attended transfer instead of blind xfer. |
| 01:01.51 | jblack | I don't know how to force reregistration of one acount though. Seems like something that should be there. |
| 01:02.01 | jblack | micc: Cool. =) |
| 01:02.25 | Micc | jblack, so now its looking for the extension in the context of the SIP trunk used to dial, not the context where the dial happens. |
| 01:02.41 | Micc | jblack, is there any way to temporarily set the transfer context before a call to dial? |
| 01:02.57 | jblack | offhand I don't remember. |
| 01:03.04 | jblack | you'll have to pardon me, I'm on codeine today. |
| 01:07.04 | Katty | hi |
| 01:07.08 | jblack | hi |
| 01:08.16 | Katty | still bummed. |
| 01:08.29 | jblack | yeah. life sucks |
| 01:08.43 | Katty | joins jblack in the grumpy section |
| 01:09.10 | jblack | let's wear all black and make moody "why do you give a shit" faces at everyone else in the channel. |
| 01:09.40 | Katty | do we get to wear fun hats |
| 01:09.53 | Katty | and tacky sweaters |
| 01:09.56 | jblack | no hats. only hoodys. |
| 01:10.09 | Katty | no, i think we need fun hats and tacky sweaters |
| 01:10.16 | Katty | go find one of each, and link it to me |
| 01:10.18 | Katty | googles |
| 01:10.34 | jblack | I'm going for moody gothic, not catholic church sunday. |
| 01:10.58 | Katty | http://www.foureyesjokeshop.com/ProductImages/Giant_Chef_Hat.jpg <- this will be my hat |
| 01:11.17 | Katty | jblack: this should be an effort to cheer ourselves up |
| 01:11.22 | Katty | jblack: go find a moody gothic hat |
| 01:11.53 | p3nguin | jblack: I would have thought so, too, but the SPA3102 seems to try whenever it feels, regardless of my reloading on the server side. |
| 01:11.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:12.08 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 01:12.34 | p3nguin | I let it sit for a while and it finally tried again. |
| 01:13.10 | Katty | http://dwightjohnston.files.wordpress.com/2008/11/ugly-pig-sweater1.jpg <- this will be my ugly sweater |
| 01:13.17 | doubletoker | what is marked mode for meetme |
| 01:13.50 | doubletoker | been searching the internet and no help finding any info about what marked mode is |
| 01:14.39 | Katty | jblack: http://tinyurl.com/y8wdff8 <- your hat |
| 01:14.45 | jblack | omgwtf it's a 14 year old riot |
| 01:14.47 | jblack | http://mediamemo.allthingsd.com/20091122/heres-a-first-man-arrested-for-not-using-twitter/ |
| 01:19.21 | Katty | http://i.imgur.com/NLN3Q.jpg <- twilight |
| 01:19.25 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 01:20.25 | jblack | that's what they're going crazy over? A movie? |
| 01:21.10 | Katty | i don't really know what the fuss is all about |
| 01:21.18 | Katty | the first twilight was okay, i guess.... |
| 01:21.23 | *** join/#asterisk ming_zym (n=ming_zym@121.0.29.237) |
| 01:21.43 | jblack | I didn't see either of them |
| 01:22.14 | sergey | Hi. Have queue and member=>SIP/1, (2,3) What strategy to any call first ring SIP/1 if no answ/busy SIP/2 also SIP/3 and again SIP/1 ? |
| 01:22.18 | jblack | Someone sould come out to wilkes-barre and watch the road with me |
| 01:22.21 | Katty | i've not seen the second one, tho they sure are making a fuss of it |
| 01:22.28 | Katty | the road? |
| 01:22.35 | jblack | sergey: That's so basic.... |
| 01:23.46 | jblack | exten=>s,n,Dial(SIP/1,30) \n exten=>s,n,Dial(SIP/2,30) \ |
| 01:23.49 | jblack | and so on |
| 01:24.01 | sergey | jblack: rrmemory - round robin with memory, remember where we left off last ring pass or ringall this not need for me |
| 01:24.14 | jblack | Oh, you want to loop a few times. |
| 01:24.15 | jblack | hold |
| 01:25.00 | Katty | jblack: this is very basic, i'm sure he can figure it out on his own. |
| 01:25.12 | Katty | jblack: you feed 1 pigeon, they all show up |
| 01:25.19 | sergey | jblack: yes but it need for queue because agents in different places and need one announce and time of answer |
| 01:25.44 | jblack | http://pastebin.com/m79ea95bf |
| 01:25.48 | jblack | sergey: that'll get you close |
| 01:26.13 | jblack | No... |
| 01:26.41 | jblack | every time I take the time to get you the answer, by the time I'm back with the answer, the question has changed. That's pissing me off. |
| 01:27.14 | voipmonk | heheh |
| 01:27.15 | voipmonk | :) |
| 01:28.58 | jblack | goes off to kill zombies |
| 01:29.12 | heliosj | Braaaaains. |
| 01:29.14 | sergey | jblack: i say about queue in first message :-) it need all queue functions for member but only order for ring |
| 01:30.29 | p3nguin | jblack: Looks like it takes 20 minutes before the SPA3102 gets around to registration again. |
| 01:30.39 | p3nguin | just fyi, of course. |
| 01:30.52 | jblack | adjust the timeout to 60s in sip.conf |
| 01:31.02 | jblack | I bet you had a long one, and the phone waited for that timeout |
| 01:31.12 | jblack | so it got "stuck." that happens to me |
| 01:33.20 | p3nguin | The registertimeout is set to the default of 20 (seconds) in the general section of sip.conf. |
| 01:33.40 | p3nguin | My thought is that the end device is configured for 20 minutes. |
| 01:35.16 | p3nguin | I'm okay with it being long like that, but it was a small inconvenience when changing the password on the device... I wanted to make sure it was still working, since I'm prone to typing mistakes. |
| 01:38.24 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 01:38.51 | *** join/#asterisk rdsousa (n=rds@bl5-127-9.dsl.telepac.pt) |
| 01:39.15 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 01:44.20 | *** join/#asterisk ehsjoar (n=ehsjoar@98.245.155.132) |
| 01:45.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:48.16 | *** join/#asterisk chendy (n=chatzill@119.139.17.129) |
| 01:53.33 | jblack | p3nguin: your local network bandwidth is free. set it to 60sec. :) |
| 01:55.07 | *** join/#asterisk `paul (n=kutimoy@123-242-230-55.sunnyvisiondatacenter.com) |
| 01:55.59 | `paul | i think i installed evrything and created a room in meetme.conf but i still get an invalid conference number evrytime i join a conference... i also have the ztdummy installed help pls |
| 01:56.00 | Micc | Now that I've set the transfer context, I need to set the parking lot, but nothing I do seems to work. |
| 01:57.06 | arcanine18_2000 | hi guys... |
| 01:57.52 | *** join/#asterisk diatonic (n=diatonic@71.220.183.193) |
| 01:58.07 | arcanine18_2000 | do you know where i can get g729 codec? |
| 01:58.20 | spck | is that the licensed one? |
| 01:58.35 | arcanine18_2000 | demo or free... just for testing/ |
| 01:58.37 | diatonic | Anyone have experience with Audiocodes FXS gateways and Asterisk 1.4? Having some trouble with them |
| 02:00.42 | spck | no idea, to both questions |
| 02:01.00 | p3nguin | micc: parkpos => 701-720 |
| 02:01.02 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:01.15 | p3nguin | micc: Just uncomment that default value and reload. |
| 02:01.18 | `paul | i think i installed evrything and created a room in meetme.conf but i still get an invalid conference number evrytime i join a conference... i also have the ztdummy installed help pls :( :( :( |
| 02:02.41 | diatonic | How can I do an SIP debug of a peer and write it to a file? Would "asterisk -rx 'sip debug peer x' > file.txt" work? |
| 02:03.03 | voipmonk | heh |
| 02:03.06 | voipmonk | yes |
| 02:03.11 | voipmonk | the color codes will show up tho |
| 02:03.33 | *** join/#asterisk ehsjoar (n=ehsjoar@98.245.155.132) |
| 02:03.57 | diatonic | Hmm... maybe if I tun off ansi colors in my terminal they won't |
| 02:04.05 | p3nguin | micc: It's possible that module reload res_features will be good enough and you won't have to do a full reload. |
| 02:04.20 | diatonic | So if I interrupt that command with a ctrl-c it would stop the debug I hope :D |
| 02:05.46 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 02:06.53 | Katty | does anyone work with RC /things/ at any length? |
| 02:07.08 | Katty | someone who would know the difference between 35Mhz, 72Mhz, and 2.4Ghz devices |
| 02:10.43 | carrar | RC in this channel |
| 02:10.44 | carrar | heh |
| 02:10.46 | carrar | Rate Center |
| 02:11.12 | carrar | 35Mhz is gonna have better range Katty |
| 02:11.50 | carrar | but longer antennas |
| 02:11.59 | carrar | and use more power |
| 02:12.04 | carrar | possibly |
| 02:12.12 | Katty | can you explain the logic behind that? |
| 02:12.29 | p3nguin | But you can use less power and have the same range as the higher frequency. |
| 02:12.40 | carrar | 35Mhz has a longer wave length and can travel through stuff, if thats what you need |
| 02:12.52 | carrar | travel through easier then 2.4 |
| 02:13.07 | carrar | lower wavelength means longer antenna |
| 02:13.16 | carrar | err longer |
| 02:13.24 | p3nguin | 2.4 GHz is line-of-sight, where 35 MHz probably isn't considered to be. |
| 02:13.34 | carrar | also uses more power too (but thats really a TX design issue) |
| 02:13.41 | carrar | correct |
| 02:14.05 | carrar | <- HAM |
| 02:14.07 | carrar | heh |
| 02:14.15 | p3nguin | <- also |
| 02:14.24 | Katty | that makes sense to me. |
| 02:14.36 | carrar | What is it for? |
| 02:14.56 | Katty | helicopter |
| 02:15.28 | carrar | Also you need to think about interfearence |
| 02:15.47 | carrar | don't want that crashing because someone is talking on their radio or cordless phone |
| 02:15.50 | carrar | or wifi |
| 02:15.50 | p3nguin | Is there a command from the CLI to show parking slots in use? |
| 02:16.14 | spck | CLI>help parkedcalls |
| 02:16.19 | carrar | show parkedcalls |
| 02:16.31 | spck | not in 1.6 |
| 02:16.32 | p3nguin | perfect, thanks. |
| 02:16.54 | p3nguin | help parkedcalls isn't in my 1.4, either. |
| 02:17.17 | p3nguin | The show was what I needed, anyway. |
| 02:17.35 | spck | do most people not run 1.6? |
| 02:17.52 | p3nguin | 0 parked calls. Hmm, now I have to figure out what happened to my call! |
| 02:17.57 | Katty | well we have a polycom here at the house |
| 02:18.03 | Katty | and the mobiles either operate at 900 or 1800 i believe |
| 02:18.33 | carrar | I might think 72 Mhz might be nice |
| 02:18.45 | carrar | 50 channels to choose from |
| 02:19.02 | p3nguin | Ah, found it! Damn two-line softphone had the call on hold on the other line. |
| 02:19.53 | carrar | Katty, http://en.wikipedia.org/wiki/Radio-controlled_aircraft |
| 02:20.02 | carrar | scroll down to Frequencies and sub-channels |
| 02:20.21 | *** part/#asterisk `paul (n=kutimoy@123-242-230-55.sunnyvisiondatacenter.com) |
| 02:20.25 | carrar | I'd go with 72 |
| 02:21.14 | carrar | stay away from 27Mhz & 2.4ghz |
| 02:21.34 | carrar | note that 2.4Ghz and 50/53 are also HAM Radio bands that get priority and TONS more RF power limits |
| 02:21.45 | Katty | educates herself |
| 02:21.58 | carrar | 27Mhz is CB people and their illegal radios |
| 02:22.10 | carrar | so keep far away from that |
| 02:22.27 | carrar | really leaves you with 72 & 75 |
| 02:22.47 | Katty | i don't see 35 on the list :< |
| 02:23.11 | p3nguin | hahaha illegal radios... |
| 02:23.12 | carrar | well thats legal in other countries |
| 02:23.25 | carrar | maybe not in the US, or that page isn't updated |
| 02:23.34 | p3nguin | Like amateur operators never exceed the power limits nor talk out-of-band. |
| 02:24.05 | carrar | p3nguin, true, but you will find more overmodulated over powered people running crappy linears on or around the cb bands |
| 02:24.14 | p3nguin | Oh, I know. |
| 02:24.19 | carrar | heh |
| 02:24.19 | p3nguin | Been there, done that. |
| 02:24.36 | Katty | hmmmm. |
| 02:24.40 | carrar | I've listen to Art Bell and his nutty friends on 80 meters |
| 02:24.52 | carrar | hard to tell the difference between them and CB people sometimes |
| 02:25.19 | p3nguin | People hate when I run my Kenwood on the CB channels. They think it's the worst sounding radio on there because it's not bleeding all over the place and swinging their meters like their other friends. |
| 02:25.24 | Katty | okay, so theoretically |
| 02:25.30 | Katty | the long band waves are slower |
| 02:25.34 | carrar | no |
| 02:25.54 | carrar | radio waves technically travel at the same speed |
| 02:25.59 | *** join/#asterisk jschmidt (n=jschmidt@hrbg-74-46-22-201-pppoe.dsl.hrbg.epix.net) |
| 02:26.01 | carrar | well |
| 02:26.02 | p3nguin | The energy travels... what he said. |
| 02:26.07 | carrar | yes |
| 02:26.11 | p3nguin | Roughly at the speed of light. |
| 02:26.18 | Katty | so this is all about the length of the band. |
| 02:26.26 | Katty | and the longer the band, the more trauma it can stand |
| 02:26.28 | carrar | however you have less bandwith with the lower frequencies |
| 02:26.53 | Katty | so the response tiem would be quicker on a 2.4ghz |
| 02:26.59 | carrar | the more the radio waves can penitrate the lower they are |
| 02:27.00 | spck | not much difference between the two honestly |
| 02:27.04 | spck | light and radio waves that is |
| 02:27.14 | spck | just a different spectrum of radiation |
| 02:27.22 | p3nguin | ~300M m/s, right? |
| 02:27.58 | carrar | Katty if you are sending back large amounts of DATA over the signals then higher is better |
| 02:28.02 | carrar | like images |
| 02:28.27 | carrar | but this is just ailerons and motor speeds right? |
| 02:28.34 | Katty | well not exactly |
| 02:28.36 | carrar | so it doesn't matter |
| 02:28.38 | carrar | oh |
| 02:28.51 | Katty | he is going to put a headset goggle thing on it |
| 02:28.59 | Katty | tho i'm not sure exactly how that's supposed to transmit |
| 02:29.28 | carrar | well I'm guessing you are not gonna trasmitte that on the same frequency you are controlling the aircraft with |
| 02:29.48 | Katty | i honestly have no idea |
| 02:29.51 | carrar | so for the RC part you don't need lots of BW |
| 02:30.46 | carrar | and it seems like people generally use 72Mhz for aircraft which is good |
| 02:30.53 | carrar | for controlling it |
| 02:31.19 | Katty | i would also think that 2.4 isn't that developed yet |
| 02:31.30 | carrar | maybe not for RC |
| 02:31.38 | carrar | but it is for wifi and HAM |
| 02:31.46 | carrar | which a good reason not to use it |
| 02:31.57 | Katty | ther'es not a lot of heli's out there with 2.4 |
| 02:32.19 | carrar | All it takes is about 10 seconds of bad signals and you loose your aircraft |
| 02:32.38 | Katty | 2.4ghz would help with that |
| 02:32.43 | Katty | less hopping around |
| 02:32.44 | carrar | You're not worried about other helicopters |
| 02:32.50 | Katty | no. |
| 02:33.09 | carrar | At RC sites people tag and agree typically with what channel they are on |
| 02:33.11 | *** join/#asterisk s0lid (n=s0lid@203.192.188.182) |
| 02:33.26 | jblack | huh? |
| 02:33.38 | carrar | it's if there are any wifi or ham stuff going on around the area that will really mess you up |
| 02:33.58 | Katty | well we do have wifi here |
| 02:34.02 | carrar | not even worth it in my mind to entertain that band |
| 02:34.07 | spck | gives me a great idea for a prank at a RC con |
| 02:34.11 | Katty | but i doubt it would interfere in the back yard |
| 02:34.11 | spck | lol |
| 02:34.15 | carrar | hahah spck |
| 02:34.19 | carrar | evil |
| 02:34.30 | jblack | it'll be nice when someday everything is ipv6 |
| 02:34.33 | Katty | k, 72 it is! |
| 02:34.37 | spck | just watching like 20 of them drop out of the sky at once would be worth it |
| 02:34.43 | p3nguin | We used to do that when the guys would bring their boats to the reservoir. |
| 02:34.46 | Katty | http://www.esky-heli.com/twf-sz-esky-6ch-helicopter-rtf-c-177_38.html <- help me pick one out :P |
| 02:34.47 | jblack | everything from phones to walkie talkies |
| 02:34.54 | carrar | Katty use 72Mhz for RC, then you can use 2.4 for whatever it is you are attaching to the hellicopter |
| 02:34.56 | *** join/#asterisk diatonic (n=diatonic@97.121.28.29) |
| 02:34.56 | spck | eh shit i should be careful DHS is prolly in here |
| 02:35.06 | Katty | carrar: that's a good idea, actually. |
| 02:35.08 | p3nguin | Turn on the big amplifier and just modulate a little. They wouldn't have any control. |
| 02:35.35 | carrar | attach a batter operated wifi to the helicopter with GPS gyro stationary controls |
| 02:35.39 | carrar | battery |
| 02:35.52 | carrar | then you will have the highest wifi antenna around :) |
| 02:35.54 | spck | you can actually get IP addressable radio modems |
| 02:36.31 | spck | about $350 each now |
| 02:36.31 | lanning | bring in an arc transmitter, watch everything go down... :) |
| 02:36.31 | spck | have to stick it in something rather large though |
| 02:36.31 | Katty | http://www.esky-heli.com/esky-belt-cp-v2-red-6ch-ccpm-rc-helicopter-rtf-000003-p-4339.html <- this OR |
| 02:36.31 | Katty | http://www.esky-heli.com/esky-honey-bee-king-3-bluewhite-6ch-ccpm-rc-helicopter-rtf-000015-p-4330.html <- this |
| 02:37.33 | carrar | which one has more power? |
| 02:37.35 | spck | i take it cellular wouldn't work for RC? |
| 02:37.38 | carrar | lifting power |
| 02:37.51 | Katty | i have no idea. |
| 02:37.58 | Katty | which one will be more fun for ryan to fly |
| 02:38.03 | Katty | and why :P |
| 02:38.10 | carrar | if you are gonna attach stuff you need a good powerfull motor |
| 02:38.22 | Katty | looks at power specs |
| 02:38.34 | spck | what's the controller cost? |
| 02:39.18 | spck | eh not bad, |
| 02:39.22 | spck | never was into that stuff |
| 02:39.55 | carrar | Esky Belt CP V2 has a larger rotor diameter |
| 02:40.00 | carrar | possibly lifting more? |
| 02:40.41 | Katty | spck: there's a controller included with them |
| 02:40.44 | Katty | spck: but it's not fancy |
| 02:40.49 | spck | ahh |
| 02:41.14 | spck | i would break one of those so fast |
| 02:41.18 | spck | not even funny |
| 02:41.22 | carrar | Katty also the Esky Honey Bee King 3 doesn't have a 72Mhz option |
| 02:41.27 | Katty | ooh bad. k |
| 02:41.36 | Katty | alright, that settles that |
| 02:41.41 | carrar | heh |
| 02:41.45 | Katty | i'd guess left hand throttle? |
| 02:42.16 | carrar | You should also makeup some, free anytime, anywhere backrub gift certificates for you man!! |
| 02:42.21 | carrar | heh |
| 02:42.37 | carrar | standing around all day might be tiring |
| 02:43.28 | Katty | ^_- |
| 02:43.33 | Katty | he gets those anyway |
| 02:43.54 | Katty | horay! next problem...the gift wrap :/ |
| 02:44.31 | carrar | don't wrap it, fly it in! |
| 02:44.40 | Katty | oh hells no, i'd wreck that thing |
| 02:44.45 | Katty | or the dog would nab it on the way through |
| 02:45.13 | Katty | oh! wait! LOOK! the page says it comes with a giftbox :> |
| 02:49.46 | *** join/#asterisk haryv (i=lanny@174.1.114.16) |
| 03:03.41 | *** join/#asterisk [1]jschmidt (n=jschmidt@hrbg-74-46-22-201-pppoe.dsl.hrbg.epix.net) |
| 03:08.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:09.20 | *** join/#asterisk lmsteffan (n=laurent@195.221.84.4) |
| 03:09.44 | Katty | http://atlas.web.cern.ch/Atlas/public/EVTDISPLAY/0911189_01.jpg <- girl at bottom right has lost her happy. |
| 03:09.58 | Katty | ^- if anyone runs across her happy, please let me know. |
| 03:11.01 | jblack | That's you? |
| 03:11.14 | Katty | no, i'm not a blonde |
| 03:11.23 | Katty | you know that |
| 03:12.24 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:12.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 03:12.35 | jblack | There's more than one girl on the planet that changes her hair color from time to time. :) |
| 03:12.55 | Katty | also true |
| 03:12.55 | jblack | And she looks nothing like you. |
| 03:12.59 | jblack | feels retarded |
| 03:13.20 | Katty | hehehe |
| 03:14.13 | jblack | I've been stupid all day. |
| 03:14.20 | Katty | hugs jblack |
| 03:14.27 | jblack | hugs back |
| 03:14.32 | Katty | and we love you anyway! |
| 03:21.18 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 03:21.35 | [TK]D-Fender | *b00m* |
| 03:22.53 | leifmadsen | and in the eyes of a jackal I say kaaaaaaaa-boom! |
| 03:23.10 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
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| 03:25.16 | Juggie | leifmadsen, ping |
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| 03:32.38 | *** join/#asterisk OmbreNoir (n=OmbreNoi@173.177.52.163) |
| 03:33.24 | OmbreNoir | Hi, I have installed asterisk today, but I cant access to web... I need a little help :) |
| 03:34.00 | haryv | nice little split |
| 03:34.04 | leifmadsen | o.O |
| 03:34.17 | leifmadsen | OmbreNoir: you can't access the web? |
| 03:34.28 | leifmadsen | seems odd you can get on IRC but not a website |
| 03:35.01 | OmbreNoir | asterisk have a web interface no? |
| 03:35.09 | OmbreNoir | with managed.conf and http.conf |
| 03:35.46 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 03:36.17 | leifmadsen | OmbreNoir: oh, you want #asterisk-gui then |
| 03:36.38 | OmbreNoir | oh ok thx |
| 03:41.20 | *** join/#asterisk dshap (n=dshap@ip68-4-125-68.oc.oc.cox.net) |
| 03:41.24 | dshap | ~sipnat |
| 03:41.24 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:43.37 | OmbreNoir | leifmadsen just for info asterisk work fine on 64bits server (Centos 5.4) |
| 03:53.28 | dshap | Can someone here please tell me the proper way to start my Asterisk server? |
| 03:53.49 | dshap | i've just been typing the command "asterisk &" to run it in the background |
| 03:53.55 | dshap | and then "asterisk -r" to get onto the CLI |
| 03:54.08 | jblack | I have decided global warming is cool because it means I don't have to shovel as much snow. Everyone come join me in opening up all the bottles of soda pop at the supermarket, and we'll warm the eath together! |
| 03:54.46 | jblack | dshap: It depends upon your operating system and distribution. In the case of ubuntu and many others, "service start asterisk" |
| 03:54.58 | dshap | jblack: ah i'm using CentOS 5.3 |
| 03:55.10 | dshap | jblack: service start asterisk sounds good to me |
| 03:55.20 | dshap | jblack: how is that different than "asterisk &" |
| 03:55.23 | dshap | if you don't mind me asking |
| 03:56.06 | jblack | because "asterisk &" doesn't run the init scripts that su to the right user, set environment variables, etc |
| 03:56.13 | [TK]D-Fender | dshap: "asterisk &" runs a single instance in the background. if it crashes, thats it. the init script provided with the tarball calls safe_asterisk which restarts on failure, etc |
| 03:56.31 | dshap | [TK]D-Fender: ahhh what's up my man, long time no talk |
| 03:56.32 | [TK]D-Fender | dshap: Also you can set the init scrip to start on boot <- |
| 03:56.36 | dshap | ty for the info |
| 03:56.48 | jblack | This is week 1 administration stuff. |
| 03:56.49 | dshap | [TK]D-Fender: is that hard to do? |
| 03:56.57 | jblack | You should find someone to help you with beginning system administration |
| 03:57.14 | dshap | jblack: yea i know, i basically started learning system administration only because i wanted to start doing asterisk development |
| 03:57.18 | [TK]D-Fender | dshap: "chkconfig asterisk on" |
| 03:57.37 | [TK]D-Fender | dshap: This is RH basics.... |
| 03:57.47 | dshap | "error reading information on service asterisk: No such file or directory" |
| 03:57.49 | [TK]D-Fender | dshap: You should read some beginners guides on your OS |
| 03:57.56 | [TK]D-Fender | dshap: Perhaps you didn't install them |
| 03:57.57 | jblack | make friends with someone that undersatnd system adminitration well, and buy them frequent gift certificates for being your bitch. |
| 03:58.04 | jblack | is bored and lonely |
| 03:58.08 | [TK]D-Fender | dshap: in your source folder do "make config" |
| 03:58.26 | dshap | [TK]D-Fender: i definitely have asterisk working...your'e saying it could be installed but not as a service? |
| 03:58.59 | jblack | You didn't build asterisk from source, did you dshap? |
| 03:59.05 | dshap | jblack: yea i did |
| 03:59.07 | *** join/#asterisk xpot-mobile (n=xpot-mob@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:59.08 | [TK]D-Fender | dshap: No, your * works fine, just that you didn't set up the common OS init scripts etc that are normal to your OS. |
| 03:59.10 | dshap | jblack: i did make and make install |
| 03:59.14 | jblack | Did someone tell you to do that? |
| 03:59.25 | dshap | jblack: it was a while ago so i don't really remember |
| 03:59.31 | [TK]D-Fender | dshap: Sure you can run in manually... but thats like buying a car so you can push it everywhere instead of using the engine |
| 03:59.55 | jblack | Ok. Whoever told you to build it yourself was wrong. If you remember who, please tell me so that I can scream at them for a good 45 minutes |
| 04:00.39 | dshap | [TK]D-Fender: is it possible that i no longer have the source folder? that's what i downloaded initially, right? then that creates /etc/asterisk once i compile and install it |
| 04:01.00 | dshap | jblack: what should i have done? |
| 04:01.04 | jblack | if you ran make install, you could have deleted the sources and things still have asterisk.\ |
| 04:01.26 | jblack | They should have told you to get the asterisk that comes with centos, or helped you find a third party repository |
| 04:02.09 | jblack | Unfortunately, someone was in a twit mode, and though "I like the latest asterisk, so everyone should build the latest one from scratch" without taking 2 minutes to think it through. |
| 04:02.18 | *** part/#asterisk xpot-mobile (n=xpot-mob@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:02.36 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 04:02.38 | dshap | jblack: doesn't the asterisk future of telphony ebook tell you to make install it? |
| 04:02.40 | [TK]D-Fender | dshap: /usr/src/ |
| 04:02.54 | dshap | [TK]D-Fender: thank you |
| 04:03.06 | jblack | dshap: Yes, there's a chapter on how to build asterisk in the book, as there should be. |
| 04:03.37 | dshap | jblack: well i went by the book |
| 04:03.50 | jblack | The book probably isn't as clear as it should be that there are ramifications to building from source. |
| 04:04.29 | dshap | hmm this is very strange. whenever i try to start up asterisk it takes a lot longer now than it has in the past. it sits on "SIP channel loading..." for at least a minute or 2 before being up and running with a CLI |
| 04:04.31 | dshap | any ideas? |
| 04:04.48 | dshap | (i just tried to start it up to verify the version that i'm using, as i have multiple versions in my usr/src/ folder) |
| 04:05.18 | jblack | That's one I've never heard of before. |
| 04:05.23 | dshap | i just bought a new router today |
| 04:05.32 | dshap | so im trying to get it all set up like i had it before |
| 04:05.41 | drmessano | reads from the book of Armaments, Chapters 2, Verses 9-21 |
| 04:05.45 | dshap | all the NAT settings took me like 2 weeks of pain and suffering a while ago to get set up |
| 04:05.54 | dshap | [TK]D-Fender was a huge help |
| 04:06.08 | jblack | That won't be why asterisk is starting slow. |
| 04:06.10 | drmessano | Thou shalt notest compilest thy Asterisk if thou are not worthy in his sight |
| 04:06.11 | dshap | do u think the slow startup could be tied to my new router? |
| 04:06.13 | dshap | k |
| 04:06.27 | jblack | the only thing I can think of is reverse lookups for your host may be blocking |
| 04:06.41 | dshap | i mean...once it starts up it seems to be fine |
| 04:06.52 | drmessano | lack of a localhost entry in /etc/hosts |
| 04:07.55 | dshap | drmessano: was that directed at me? |
| 04:08.11 | jblack | Yes |
| 04:08.21 | jblack | he's expanding on what I said |
| 04:08.35 | drmessano | No, I like to paste random solutions to problems in channel |
| 04:08.35 | dshap | i dont have anything at /etc/hosts |
| 04:08.38 | dshap | what should i put there? |
| 04:08.39 | drmessano | HA |
| 04:08.41 | dshap | lo |
| 04:08.42 | dshap | l |
| 04:08.55 | drmessano | No /etc/hosts? oh nice |
| 04:09.27 | dshap | drmessano: nvm, i've got it |
| 04:09.30 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:09.44 | dshap | drmessano: "127.0.0.1 localhost.localdomain localhost" |
| 04:09.45 | drmessano | ok, and? |
| 04:09.49 | drmessano | ok |
| 04:10.19 | dshap | drmessano: and ::1 localhost6.localdomain6 localhost6 |
| 04:10.21 | dshap | [TK]D-Fender: ok i did make config in the appropriate source folder and then did chkconfig asterisk on - what did that do? |
| 04:10.40 | dshap | [TK]D-Fender: asterisk will now turn on when i reboot my server? |
| 04:11.08 | drmessano | Yeah, and if you rub it's nipples |
| 04:11.14 | dshap | ? |
| 04:11.27 | drmessano | Sorry, human joke |
| 04:11.32 | jaytee | god what nightmare evening this has been |
| 04:12.00 | jblack | jaytee: I feel your pain. :| |
| 04:12.19 | dshap | jblack: alright now i've got asterisk running as a service |
| 04:12.39 | jblack | Let's get the key people together, and we'll take over a desert island. |
| 04:12.50 | jaytee | we changed from a Class B subnet to a Class A subnet internally and tonight we moved the phone system over |
| 04:13.17 | jblack | That can be painful. |
| 04:13.47 | dshap | how do i get out of the CLI without shutting down asterisk? |
| 04:13.56 | drmessano | exit |
| 04:14.01 | jblack | ctrl-c won't stop asterisk |
| 04:14.06 | dshap | oh nice |
| 04:14.12 | dshap | oh wow |
| 04:14.14 | dshap | now when i do asterisk -r |
| 04:14.15 | dshap | i have colors |
| 04:14.22 | drmessano | ZOMG |
| 04:14.30 | dshap | before when i was just running asterisk like "asterisk &", whenever i reconnected to the CLI it never had colors |
| 04:14.37 | dshap | this is great |
| 04:14.51 | drmessano | Thats what happens when you spring for the good shit |
| 04:14.54 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 04:14.55 | jblack | you're comfusing things. |
| 04:14.59 | drmessano | Purple micrdot = win |
| 04:15.04 | drmessano | Purple microdot = win |
| 04:16.15 | drmessano | "One, two, FIVE" <-- incorrect way to install Asterisk |
| 04:16.36 | drmessano | "one, two, demand help on IRC" <-- better, but not best |
| 04:17.41 | dshap | drmessano: i think i did a combination of those :-p |
| 04:18.12 | drmessano | Havent you been in here for like 2 years? |
| 04:18.33 | dshap | drmessano: in may i started learning asterisk and was in here all day every day for like 2 weeks |
| 04:18.58 | drmessano | Yeah, thats right |
| 04:19.10 | dshap | drmessano: it's for this tech startup that im gonna start once i have more time |
| 04:19.19 | drmessano | You went from installing it to starting a telco in like 12 hours |
| 04:19.23 | dshap | yea |
| 04:19.25 | drmessano | Right |
| 04:19.27 | dshap | drmessano: still gonna start it |
| 04:19.32 | dshap | drmessano: but i need more time to work on it |
| 04:19.36 | dshap | im still in school |
| 04:19.44 | drmessano | Try getting asterisk to run on boot before taking over the planet, k? |
| 04:19.45 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 04:19.48 | dshap | hah |
| 04:19.57 | [TK]D-Fender | dshap: tell me you're not a waiter, k? |
| 04:20.13 | dshap | [TK]D-Fender: i'm an electrical/computer engineering student |
| 04:20.26 | drmessano | Oh an engineer |
| 04:20.33 | drmessano | Thats worse than a network admin |
| 04:20.36 | dshap | we don't learn system administration in school |
| 04:20.44 | dshap | i haven't really learned it outside of school either |
| 04:20.45 | dshap | lol |
| 04:21.00 | drmessano | Optimist --> The cup is half full |
| 04:21.11 | drmessano | Pessimist --> The cup is half empty |
| 04:21.22 | dshap | i just had this idea for a startup that involved a bit of telephony, so i learned how to use asterisk |
| 04:21.23 | jblack | realist --> and it's evaporating as we speak |
| 04:21.34 | drmessano | Engineering student --> I am going to reinvent the cup once I get the sippy lid off it |
| 04:21.42 | dshap | lol |
| 04:22.07 | jblack | dshap: Do you have money? |
| 04:22.30 | dshap | jblack: yea, a bit actually, but im gonna apply for these seed funding programs they have in sillicon valley for next summer |
| 04:22.44 | dshap | jblack: if it falls through, i already have a job from my internship i did last summer :) |
| 04:22.53 | jblack | if you pull enough together that you can actually pay people, I'm interested |
| 04:23.27 | dshap | jblack: hah well you can be sure that if it ever scales to anything that i'll need more than my little linux box to run, i'm gonna need to bring in people that actually know what they're doing |
| 04:23.39 | dshap | jblack: i basically just have the idea, the motivation, and enough skills to build a prototype |
| 04:24.03 | dshap | i'll obviously come to this channel and start talking, if it ever comes to that :-p |
| 04:24.11 | jblack | imagine a distributed setup across multiple machines and networks. That's the sort of stuff I do. |
| 04:24.30 | dshap | jblack: that's the sort of stuff i would not want to get involved with haha |
| 04:24.48 | dshap | jblack: i'll write your IRC username down :-p |
| 04:25.24 | dshap | jblack: or if you have like a business website or something..? |
| 04:25.28 | drmessano | <PROTECTED> |
| 04:25.30 | *** join/#asterisk Tech_Travis (n=tech_tra@cpe-76-87-9-130.socal.res.rr.com) |
| 04:25.50 | jblack | drmessano: prototypes are made out of duct tape. |
| 04:25.54 | dshap | drmessano: hah, i already have a proof of concept developed my friend. asterisk has been working for me for months, i just didn't know it wasn't properly installed |
| 04:25.58 | jblack | dshap: Just find me here. |
| 04:26.02 | dshap | jblack: word. |
| 04:26.22 | drmessano | LOLLLLL |
| 04:26.36 | drmessano | Yeah, that whole "running" thing is overrated |
| 04:26.45 | drmessano | It was configged like a mofo though |
| 04:26.53 | jblack | it builds. SHIP IT! |
| 04:26.58 | drmessano | HAHAHAHH!!! |
| 04:27.41 | dshap | jblack: hey for a startup that's what you've gotta do |
| 04:28.02 | jblack | It is. |
| 04:28.04 | drmessano | Ship something that you've not proven actually works |
| 04:28.29 | drmessano | Yep, engineering student |
| 04:28.36 | drmessano | "On paper, it works" |
| 04:28.41 | jblack | a startup is 40% tapdancing, 30% bullshit, 25% outright lies, and 5% actual something |
| 04:28.52 | drmessano | Let someone else go outside into sunlight and test it.. not me |
| 04:29.12 | jblack | I feel the need to kill the undead |
| 04:29.22 | drmessano | jblack: +/- 5% on the actual |
| 04:29.34 | drmessano | Nevermind the stupid name |
| 04:29.38 | drmessano | FONR.com |
| 04:29.39 | jblack | Yup. :) |
| 04:29.41 | ChannelZ | points - congress is that way |
| 04:29.59 | dshap | lol |
| 04:30.05 | drmessano | KaulMe.com |
| 04:30.20 | drmessano | Fonnit |
| 04:30.20 | dshap | jblack: have you ever heard of youmail.com ? |
| 04:30.28 | drmessano | Oh lord |
| 04:30.41 | jblack | Have you heard of <some small obscure thing you've never heard of> ? |
| 04:30.47 | drmessano | Thats right |
| 04:31.22 | drmessano | He was gonna reinvent youmail.com |
| 04:31.22 | drmessano | I remember this guy |
| 04:31.22 | drmessano | hang on |
| 04:31.22 | jblack | You're shitting me |
| 04:31.24 | drmessano | before we get into this, lets just look at the log from MAY and pretend we talked about it tonight |
| 04:31.37 | dshap | hahaha |
| 04:31.37 | jblack | That industry is going to die of old age before it gets out of diapers. |
| 04:31.43 | dshap | well |
| 04:32.07 | drmessano | http://ibot.rikers.org/%23asterisk/20090527.html.gz |
| 04:32.09 | ChannelZ | who-what? |
| 04:32.10 | drmessano | There |
| 04:32.25 | drmessano | Original convo, no need to reinvent THAT too |
| 04:32.44 | jblack | really. The big five (verizon, at&t, sprint, t-mobile, pick one), are slowly moving to an unlimited model where all that stuff is free. |
| 04:32.54 | jblack | and built in. |
| 04:33.01 | jblack | Youmail == pony express |
| 04:33.06 | haryv | I think T-mobile is in redmond |
| 04:33.45 | haryv | Had a oprotunity to work for them the same week, thay were formed after buying out touchwireless |
| 04:33.50 | drmessano | Digium is in Huntsville |
| 04:33.56 | jblack | hell. _google_ is moving in on youmail. |
| 04:34.08 | jblack | i.e. dooooooooooooooooooooooomed search. |
| 04:35.01 | dshap | what would you guys recommend i read to get up to speed on system administration? i plan on doing a moderate amount of asterisk development in the future and i don't want to be clueless on this stuff anymore |
| 04:35.06 | *** join/#asterisk Carlos_PHX__ (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 04:35.09 | dshap | something distro-specific? or not even |
| 04:35.57 | drmessano | dshap: Read the TFOT book and learn how to config a basic working dialplan before trying to get 50 mil in venture capital |
| 04:36.05 | drmessano | ~book |
| 04:36.06 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 04:36.38 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 04:36.46 | snadge | thats why call centres all use lucent, or now cisco ;) |
| 04:36.53 | drmessano | I know, we're all morons.. Surely a week of asterisk is enough to master it's use enough to create the next Verizon.. But newsflash, you're not the first |
| 04:37.37 | leifmadsen | you make me next vonage!? |
| 04:37.47 | leifmadsen | you pay now! |
| 04:37.52 | leifmadsen | I build latah! |
| 04:37.58 | drmessano | HA |
| 04:38.16 | snadge | my flatmate works for a large pay television company.. he asked them why they don't use asterisk.. the reply.. asterisk is not for large call centres |
| 04:38.19 | snadge | interesting :P |
| 04:38.32 | dshap | drmessano: i was talking more along the lines of linux/unix stuff which i've demonstrated that i know very little about. i know there are channels for those but i just kinda wanted to get your guys' opinions from an asterisk developer's standpoint |
| 04:38.33 | leifmadsen | funny, because I've built a fairly large call center on asterisk |
| 04:38.53 | leifmadsen | dshap: find a distro you like, then buy a book on it |
| 04:38.58 | snadge | instead.. they pay cisco god knows how much money.. for their ip telephony system |
| 04:38.58 | leifmadsen | that's what I did |
| 04:39.06 | leifmadsen | snadge: how many seats? |
| 04:39.08 | drmessano | dshap: You've demonstrated you know very little about asterisk AND linux.. get books on BOTH |
| 04:39.19 | dshap | leifmadsen: i like CentOS which is what i have - but it seems like the books are only on RHEL |
| 04:39.22 | drmessano | Steal leifmadsen's book, then buy it |
| 04:39.32 | leifmadsen | dshap: CentOS == RHEL |
| 04:39.33 | dshap | drmessano: i have a PDF of the asterisk book |
| 04:39.34 | drmessano | dshap: Duh.. |
| 04:39.35 | snadge | leifmadsen: oh.. several hundred logged in at any one time |
| 04:39.40 | dshap | leifmadsen: gotcha |
| 04:39.47 | snadge | thousands of employees |
| 04:39.52 | leifmadsen | snadge: asterisk can do it, but you need an experienced asterisk person to do it |
| 04:40.09 | drmessano | Only Russell Bryant runs Asterisk on ubuntu, and if you listen to that guy, well, you know.. |
| 04:40.16 | leifmadsen | snadge: there are few of them around. They don't mean, "Asterisk isn't for large call centers", they mean, "I don't know anyone who can do this on Asterisk" |
| 04:40.32 | leifmadsen | drmessano: that guy is a nubjockey |
| 04:40.40 | snadge | or i dont know anyone who would risk doing this on asterisk |
| 04:40.46 | snadge | if the cisco stuff doesnt work.. cisco gets fined heaps of money |
| 04:40.54 | snadge | if asterisk doesnt work.. well.. too bad :p |
| 04:41.04 | drmessano | There's a guy running an ITSP on Asterisk, large number of concurrent calls.. badass shit |
| 04:41.22 | drmessano | I say that like there's just the one.. but there's a few out there |
| 04:41.36 | snadge | i'd be curious as to what the largest asterisk deployment is |
| 04:41.39 | drmessano | Just one I got a nice demo of the layout, and it's pretty awesome |
| 04:42.07 | haryv | university of huston is biggest? |
| 04:42.08 | dshap | leifmadsen: do you think a book on RHEL would cover UNIX commands or just RHEL-specific stuff? |
| 04:42.12 | snadge | as with anything open source.. you never know where its going to pop up.. but larger installations should generate more attention |
| 04:42.32 | leifmadsen | dshap: RHEL == Linux, so all the commands are going to be similar |
| 04:42.39 | drmessano | I hear Cisco runs Asterisk internally, though they lack some features with their SIP flashed 7960's |
| 04:42.55 | snadge | lol |
| 04:42.58 | drmessano | Ok, maybe not |
| 04:43.02 | drmessano | But that would rock out |
| 04:43.14 | snadge | the anti dog food principle |
| 04:43.21 | leifmadsen | snadge: the largest deployments tend to be hush-hush, but the Canadian government has about 100 large servers running Asterisk to handle the IVRs for CRA (Canada Revenue Agency) |
| 04:43.22 | snadge | run your enemies equipment.. so you know what they're up to |
| 04:43.27 | drmessano | hahahah |
| 04:43.36 | drmessano | Would that be eating your own dog shit? |
| 04:43.54 | snadge | i'll go with that yeah.. hehe |
| 04:43.56 | leifmadsen | Cisco *is* paying attention to Asterisk, along with the other big providers |
| 04:44.12 | leifmadsen | they have it in their shops I would imagine |
| 04:44.25 | snadge | cisco deserve to die.. i had a guy from nortel, tell me how theres this massive conspiracy |
| 04:44.27 | leifmadsen | I have no proof other than to say they're stupid if they don't |
| 04:44.35 | snadge | to kill of ATM networks, and replace them with ethernet |
| 04:44.38 | snadge | even though its inferior |
| 04:44.39 | drmessano | Cisco is afraid of Asterisk.. that's why they keep getting dates for the developers |
| 04:44.42 | leifmadsen | ya, nortel people are kinda crazy |
| 04:44.44 | drmessano | Trying to distract them |
| 04:45.07 | haryv | So what contributed to the demise of Nortel? |
| 04:45.15 | snadge | the reality is that ATM is better than ethernet.. but its like VHS vs betamax |
| 04:45.33 | Juggie | leifmadsen, ssshhhhh |
| 04:45.36 | snadge | ethernet.. juniper.. cisco |
| 04:45.43 | leifmadsen | Juggie: tell no one! |
| 04:46.13 | snadge | huawei |
| 04:46.28 | Juggie | leifmadsen, its less but i woudnt stake my career on a 1000 seat call center on *. |
| 04:46.40 | Juggie | letting bell handle it is fine by me. |
| 04:46.40 | drmessano | Just watch twitter: "leifmadsen: Drinking a $500 cup of tea and watching TV with my gf" at the same time in Cisco's evil underground bunker "HAHA, another week without Asterisk's little DUNDI world domination... Our plans are working!!" |
| 04:46.44 | haryv | Lief, been in the cra data center here in van. Heard rummer a few years ago, that all there PBXs were going to be junked as soon as the voice cat5 lines were going to be installed. Those PBX's were not that old. |
| 04:47.11 | drmessano | The week russellb got married, Cisco dropped a new release of CCM.. coincidence? |
| 04:47.17 | Juggie | haryv, i highly doubt that. |
| 04:47.18 | drmessano | All well planned |
| 04:47.29 | Juggie | cra is heavily centrex for general office |
| 04:47.31 | haryv | Juggie, or at least updated |
| 04:47.42 | jaytee | the demise of Nortel began shortly after Nortel changed it's advertising slogan to "Yeah, we've got a license for that!" |
| 04:47.42 | jaytee | time for me to get some sleep, nite all |
| 04:47.44 | Juggie | by heavily i mean 99.99% |
| 04:47.48 | leifmadsen | Juggie: I'd certainly try building a 1000 seat call center at this point. I've already done 100+ one |
| 04:47.49 | snadge | nortel gave away a heap of equipment for free during the .com boom.. and was relying on expensive service contracts |
| 04:47.55 | Juggie | leifmadsen, you've got balls. |
| 04:48.01 | snadge | problem is.. all these .coms went bust |
| 04:48.01 | leifmadsen | sure do! |
| 04:48.03 | leifmadsen | checks |
| 04:48.09 | Juggie | i woudnt. |
| 04:48.12 | leifmadsen | yup! still there |
| 04:48.20 | leifmadsen | Juggie: you don't even do Asterisk anymore :) |
| 04:48.24 | leifmadsen | there! I said it! |
| 04:48.28 | Juggie | :) |
| 04:48.34 | leifmadsen | 1.6.2 is pretty awesome stuff |
| 04:48.40 | haryv | so, when those .com went bust nortels income dropped off a cliff? :) |
| 04:48.44 | leifmadsen | anyways, speaking of call centers, I've got a tiny maintenance to perform |
| 04:48.53 | Juggie | leifmadsen, i play a part ;) |
| 04:48.59 | snadge | well i wont drop names.. but austar communications is one of 2 pay tv companies in australia.. they use cisco currently |
| 04:49.04 | Juggie | i'm still at the dance, just not dancing w/ the prom queen. |
| 04:49.18 | drmessano | DJ Belieny had an awesome presentation at the Atlanta Asterisk Installfest about an ITSP he had set up on 1.6 |
| 04:49.30 | snadge | but they probably could be convinced to roll out asterisk.. the I.T. people would need to be err.. persuaded somehow |
| 04:49.41 | Juggie | leifmadsen, none of the boxes would get calls w/o the 1-800 network i help manage :) |
| 04:49.53 | drmessano | Using DUNDI and realtime, load balanced boxes, fault tolerance in the form of single tasking boxes you could add/remove as needed |
| 04:50.05 | drmessano | Yet people still buy Cisco.. tsk tsk |
| 04:50.19 | Juggie | one issue we have that no one has solved |
| 04:50.22 | snadge | like the old saying.. you don't get fired for buying IBM |
| 04:50.25 | Juggie | is being able to take a box out of service |
| 04:50.28 | snadge | even though they're expensive and painful |
| 04:50.33 | drmessano | snadge: Yes you do, actually |
| 04:50.50 | snadge | they still have some enormous contracts though |
| 04:50.58 | drmessano | snadge: I fired a salesperson for buying an IBM laptop |
| 04:51.12 | snadge | well thats kind of understandable.. the thinkpads used to be good |
| 04:51.16 | drmessano | snadge: So there, mythbusted |
| 04:51.18 | snadge | now they're cheap crap made by lenovo |
| 04:51.28 | drmessano | Actually, the Lenovo shit is nice |
| 04:51.29 | leifmadsen | Juggie: I'd be interested in exploring that as I have some ways of solving that issue |
| 04:51.45 | haryv | drmessano have a issue with IBM? |
| 04:51.48 | leifmadsen | depends on topology as to which approach to apply |
| 04:51.51 | drmessano | I love that some of them are spillproof |
| 04:51.52 | Juggie | leifmadsen, i'd like to hear them |
| 04:52.00 | snadge | i used to work for big blue |
| 04:52.09 | leifmadsen | easiest way is to run everything via OpenSIPS with the distribution module I imagine |
| 04:52.21 | leifmadsen | it'll just fail over to the next server once you take it offline |
| 04:52.26 | Juggie | leifmadsen, nah, lets say you have a DS3, you know hypothetically :) |
| 04:52.34 | drmessano | OH OH |
| 04:52.38 | drmessano | Digium has that card! |
| 04:52.39 | drmessano | Wait, no |
| 04:52.40 | leifmadsen | goes back to work... like he should |
| 04:52.54 | snadge | yeah i have to reinstall a server.. sigh |
| 04:52.55 | Juggie | leifmadsen, and you have 7 * boxes w/ a quad span pri |
| 04:53.03 | leifmadsen | drmessano: classic case of marketing people getting too excited to sell something before talking to engineering |
| 04:53.53 | Juggie | leifmadsen, so the problem is how do you take a box offline |
| 04:53.53 | drmessano | I have a naked pic of the Digium DS3 card on my wall |
| 04:53.53 | leifmadsen | Juggie: very carefully |
| 04:53.53 | haryv | I did some contrct work for IBM last year. Thay crammed there scedule down our thoughts and would not let up on major upgrades of HBC. |
| 04:53.54 | haryv | Anyway, enough of them :) |
| 04:54.08 | Juggie | leifmadsen, if you use proper hunting like round robin or longest idle |
| 04:54.11 | Juggie | its almost impossible |
| 04:54.16 | Juggie | you need some way to refuse the call |
| 04:54.22 | drmessano | http://www.venturevoip.com/images/DS3_announce.gif <-- SEXY |
| 04:54.24 | Juggie | or at least push it to another channel |
| 04:54.43 | drmessano | http://www.venturevoip.com/images/DS3_announce.gif <-- Look at its perky BNCs.. *drool* |
| 04:55.25 | Juggie | leifmadsen, the problem is you need some way to take the idle channels out of service w/o interupting active calls |
| 04:55.33 | snadge | that is almost pornographic |
| 04:55.47 | Juggie | possible we might be able to do something on the adtran |
| 04:56.24 | drmessano | snadge: not saying i've looked at it naked |
| 04:56.50 | Juggie | leifmadsen, see gtalk |
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| 05:22.19 | OmbreNoir | with asterisk is possible to call to telephone normal or just to ip or Sip? |
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| 05:35.09 | ChannelZ | if you have an ATA sure |
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| 05:50.24 | TJNII | ChannelZ: I think that was an engrish variation of the ever present "How do I make free calls with Asterisk" question. |
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| 05:52.16 | bbt | is there any way to adjust the hangup time. im using a gigaset c47h and it takes a good 5 seconds from when i hang up until it actually hangs up |
| 05:52.47 | ChannelZ | TJNII: hmm probably |
| 05:52.57 | hyphenex | anyone had any experience with the OpenVox A400P? |
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| 06:02.21 | OmbreNoir | lol is not for 'free calls with Asterisk' is just for know if is possible or not |
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| 06:14.28 | error404notfound | i don't know where should i ask this, but i find this channel to be most appropriate, when i transfer a call i "407 Proxy Authentication Required" in twinkle. |
| 06:26.01 | error404notfound | bump! |
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| 06:42.09 | [gnubie] | waves |
| 06:43.11 | [gnubie] | tzafrir: how can i disable support for h323 when rebuilding asterisk to .deb package? |
| 06:43.33 | Buklov | modules.conf |
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| 07:00.37 | [gnubie] | Buklov: no, not preventing to load the module. building the asterisk in .deb format depends on h323. but since i don't need it, it is better to disable it or remove it as a dependency |
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| 08:19.07 | agx | hi, if i use Busy() in Dialplan on an incoming call from analog line (Digium TDM400p card) i keep hear it ringing instead of Busy; do i need to Answer() ? |
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| 08:19.55 | krion | hi |
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| 08:35.46 | ChannelZ | agx: You should Answer() first if it's the start of a dialplan |
| 08:36.14 | agx | ChannelZ: but this will result into a billable call for the caller |
| 08:37.53 | kaldemar | agx: is there a reason not to use Hangup() ? |
| 08:38.04 | ChannelZ | Well then don't answer the call at all |
| 08:38.27 | kaldemar | the tone is different, yes, but then it won't be billable and the user gets an indication. |
| 08:39.07 | ChannelZ | No I mean why answer just to indicate busy |
| 08:39.20 | ChannelZ | If you're worried about the other person's billing just let it ring |
| 08:39.57 | agx | i need to return BUSY as in ISDN :) |
| 08:40.14 | agx | if i just Hangup they think we're not answering them on purpose :-P |
| 08:41.10 | ChannelZ | But if you have incoming lines open that it's not busy on the telco's side, you're actually NOT answering them on purpose... ? |
| 08:43.10 | angryuser | I am searching someone willing to backport one patch from 1.4 to 1.2 the patch is simple but i dont know much C, willing to pay if this works |
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| 09:16.48 | madduck | after enabling tcp with a 1.6.2.0~rc2-0.7616 asterisk version, I can connect clients just fine, but when a call is instantiated, they cannot hear each other, despite RTP flowing as expected. |
| 09:16.57 | madduck | I do not allow reinvites, so it's all via the asterisk |
| 09:17.03 | madduck | any idea what could be going on? |
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| 09:22.59 | madduck | <PROTECTED> |
| 09:23.03 | madduck | <PROTECTED> |
| 09:23.13 | madduck | that is the server telling about port unreachable |
| 09:27.59 | madduck | it works fine with SIP/udp |
| 09:29.00 | kaldemar | is it a SIP packet that causes the port unreachable message? |
| 09:29.12 | madduck | nah, it seems one for every rtp packet |
| 09:29.16 | tzafrir | madduck, there's asterisk_1.6.2.0~rc6-0.7810 |
| 09:29.18 | madduck | 10:29:04.341503 IP 84.74.106.21.16384 > 77.109.139.86.15814: UDP, length 172 |
| 09:29.18 | madduck | 10:29:04.362444 IP 77.109.139.86 > 84.74.106.21: ICMP 77.109.139.86 udp port 15814 unreachable, length 208 |
| 09:29.27 | madduck | tzafrir: really? apt does not suggest it⦠|
| 09:30.13 | madduck | hm |
| 09:30.17 | madduck | something nat it seems |
| 09:30.18 | tzafrir | Location has changed a bit. See http://updates.xorcom.com/pkg-voip/ |
| 09:30.21 | madduck | Peer audio RTP is at port 192.168.14.98:16384 |
| 09:30.31 | madduck | Audio is at 77.109.139.86 port 15814 |
| 09:31.17 | madduck | new connection: |
| 09:31.19 | madduck | 10:30:50.797646 IP 84.74.106.21.16384 > 77.109.139.86.11182: UDP, length 172 |
| 09:31.26 | madduck | but asterisk sees |
| 09:31.26 | madduck | Peer audio RTP is at port 192.168.14.98:16384 |
| 09:31.30 | madduck | so port is right, IP wrong |
| 09:32.29 | agx | ChannelZ: sorry was on the phone; the only way to have BUSY on a ZAP analog port is to engage the line itself? |
| 09:32.59 | Faustov | hi, any idea why -U username command line option prevents asterisk from starting? Without otions or with any other options it starts and forks into bckground no problems, but with -U it dies right after startup without any errors |
| 09:33.02 | agx | ChannelZ: perhaps instead of Busy() i can do Dial(Zap/1/) to keep it up and busy? |
| 09:33.12 | Faustov | (1.6.1.10) |
| 09:33.15 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 09:38.49 | *** join/#asterisk mr_ian_ (n=mr_ian@S0106001b63f49383.du.shawcable.net) |
| 09:44.11 | *** part/#asterisk alecdavis (n=alec@202-78-149-14.cable.telstraclear.net) |
| 09:44.15 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 09:44.32 | arossouw | has anyone tried speex codecs on asterisk? |
| 09:47.12 | Faustov | I have, but with some bandwidth margin I stayed with 711 |
| 09:47.34 | *** join/#asterisk jojo^ (i=jol@ph4.se) |
| 09:47.55 | arossouw | is the quality of audio good? |
| 09:48.10 | arossouw | or does that depend on the phone model? |
| 09:48.20 | jojo^ | Doing Dial() on a SIP channel, is there some way to get the callid of the call after Dial has returned? |
| 09:48.26 | Faustov | IIRC it was comparable to gsm |
| 09:50.13 | *** join/#asterisk Omorika (n=omorika@89.201.165.226) |
| 09:50.23 | Omorika | hey guys |
| 09:50.28 | Omorika | i have a question |
| 09:51.33 | *** join/#asterisk DrCron (i=rszasz@saxonco.com) |
| 09:53.21 | jojo^ | For the active channel, the SIP callid is stored in the variable SIPCALLID. Is there some similiar way to get the SIP callid from a Dial()-call that just ended? |
| 09:53.43 | jojo^ | Omorika, Go ahead and ask |
| 09:55.35 | *** join/#asterisk thenthenio (n=mauro@217.221.242.43) |
| 09:55.46 | thenthenio | Hello! |
| 09:55.55 | thenthenio | I just set up an asterisknow PC... |
| 09:56.08 | thenthenio | Now I just want to try connecting 2 softphones and test a call... |
| 09:56.25 | thenthenio | I set up 2 internal extensions... |
| 09:56.46 | thenthenio | Now I'm trying to guess how to configure wengophone to run it with Asterisk in my local net... |
| 09:57.07 | Omorika | guys |
| 09:57.09 | Omorika | chan_dahdi.c: == Primary D-Channel on span 2 down |
| 09:57.09 | Omorika | chan_dahdi.c: No D-channels available! Using Primary channel 47 as D-channel anyway! |
| 09:57.09 | Omorika | chan_dahdi.c: -- Hungup 'DAHDI/55-1' |
| 09:57.10 | Omorika | chan_dahdi.c: == Primary D-Channel on span 2 up |
| 09:57.13 | thenthenio | What parameters I need to configure on wengophone? |
| 09:57.16 | Omorika | this is what's happening |
| 09:57.27 | Omorika | i'm getting D-channel restarted |
| 09:57.32 | Omorika | and all calls hung up |
| 09:58.09 | Omorika | libpri version is 1.4.10.2 |
| 09:58.17 | Omorika | and asterisk is 1.6.2.0-rc1 |
| 09:58.33 | *** join/#asterisk sgimeno (n=sgimeno@226.Red-80-33-64.staticIP.rima-tde.net) |
| 10:00.10 | n0cturnal | Omorika: have you spoken to your provider? D-Channel dropping out is usually a telco issue |
| 10:01.05 | Omorika | the problem is |
| 10:01.15 | Omorika | that it's happening on 2 spans |
| 10:01.25 | Omorika | each one a different provider |
| 10:01.43 | n0cturnal | ahh |
| 10:01.50 | Omorika | hard chance both providers have that problem |
| 10:02.00 | n0cturnal | i dont suppose you have a different card to try? :P |
| 10:02.26 | kaldemar | that can happen when there's a physical problem with the connection |
| 10:02.48 | Omorika | this one is Digium Wildcard TE410P |
| 10:04.29 | Omorika | i'm also seeing lots of: |
| 10:04.34 | Omorika | chan_dahdi.c: -- Moving call from channel 94 to channel 100 |
| 10:04.47 | Omorika | when exactly dahdi moves a channel? |
| 10:05.04 | Omorika | chan_dahdi.c: Call specified, but not found? |
| 10:05.18 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:85a2:a8ac:4274:12ba) |
| 10:09.46 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 10:14.18 | *** join/#asterisk contrabanda (n=c0ntr@79.99.248.8) |
| 10:15.51 | Omorika | guys |
| 10:15.56 | Omorika | i'm seeing these messages |
| 10:15.59 | Omorika | [Nov 25 11:11:38] VERBOSE[31024] chan_dahdi.c: q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED |
| 10:15.59 | Omorika | [Nov 25 11:11:38] VERBOSE[31024] chan_dahdi.c: q921.c:805 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED |
| 10:16.26 | thenthenio | Where can I find help on how to configure just 2 softphones and try an internal call between the 2? |
| 10:17.22 | *** join/#asterisk KermitTheFragger (n=KermitTh@217.149.197.118) |
| 10:26.46 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 10:33.08 | tzafrir | thenthenio, start with configuring a single soft-phone an call between asterisk and it |
| 10:33.51 | tzafrir | use the sample dialplan (extensions.conf) - the demo context provides some useful stuff for such testing |
| 10:34.21 | *** part/#asterisk contrabanda (n=c0ntr@79.99.248.8) |
| 10:34.56 | Omorika | guys |
| 10:34.59 | Omorika | question |
| 10:35.02 | creativx | denied |
| 10:35.53 | Omorika | can the lack of "switchtype" parameter in chan_dahdi cause my issues? |
| 10:36.14 | Omorika | it supposed to be set to "euroisdn" |
| 10:36.15 | tzafrir | the default value oif "switchtype" is "national" |
| 10:36.25 | kaldemar | it may cause issues for sure |
| 10:36.26 | Omorika | but, how is it working at all? |
| 10:36.37 | tzafrir | Sadly that default was set by silly Americans |
| 10:36.41 | Omorika | :) |
| 10:36.46 | Omorika | lol |
| 10:37.06 | *** join/#asterisk lmsteffan (n=laurent@114.69.191.131) |
| 10:37.11 | Omorika | ok guys.. so.. i can originate some calls, but after some time they are dropped |
| 10:37.29 | Omorika | i don't know much about ISDN |
| 10:37.42 | tzafrir | 'pri debug' (or 'pri set debug' ) is your friend |
| 10:38.03 | tzafrir | pri set debug span 1 |
| 10:38.11 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 10:38.26 | Omorika | its already set |
| 10:38.27 | thenthenio | tzafrir: I don't know what how to configure the wengophone parameters... |
| 10:38.38 | Omorika | "Protocol Discriminator: Q.931 (8)" |
| 10:39.00 | kaldemar | pastebin is your next best friend: |
| 10:39.02 | kaldemar | ~pb |
| 10:39.03 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 10:39.07 | tzafrir | thenthenio, isn't it called "cutecom" now? (Wengo no longer develops it) |
| 10:39.20 | Omorika | q921.c:444 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED |
| 10:39.27 | thenthenio | Yes that one! |
| 10:40.46 | thenthenio | tzafrir: I set up 2 extensions and I would like to test a connection between the 2 using softphones... |
| 10:41.00 | *** join/#asterisk kron4eg (n=kron@unaffiliated/fixer/x-387521) |
| 10:41.21 | *** part/#asterisk kron4eg (n=kron@unaffiliated/fixer/x-387521) |
| 10:41.59 | tzafrir | thenthenio, can you call from your phone to an IVR, voicemail or whatever on Asterisk? |
| 10:42.08 | *** join/#asterisk nighty^ (n=nighty@122.249.122.91) |
| 10:43.00 | thenthenio | tzafrir: Dunno what you mean |
| 10:43.24 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:44.09 | thenthenio | Wengophone is asking me some configs: account name, ID, password, SIP domain, Server... |
| 10:45.06 | Chainsaw | Anyone else had 1.6.1.10 hit the deck with a segfault in res_musiconhold.so? (Will enable debugging so I can properly troubleshoot it next time) |
| 10:45.53 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 10:46.47 | *** join/#asterisk romb_work (i=romb@89.28.249.41) |
| 10:46.55 | romb_work | hello all |
| 10:47.05 | *** part/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 10:47.33 | romb_work | i cant get/set callerid from calling peer in asterisk 1.6 |
| 10:47.53 | romb_work | i'm trying to use ${CALLERID(num)} |
| 10:48.20 | *** join/#asterisk Moc (n=mochouin@64.235.217.36) |
| 10:48.25 | romb_work | but it's empty |
| 10:48.29 | romb_work | any help? |
| 10:49.11 | *** join/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 10:50.40 | romb_work | i'm using chan_sip if this is matter |
| 10:51.20 | fcois93 | how to do a DIAL in the CLI (asterisk 1.6.1)? |
| 10:51.55 | kaldemar | fcois93: help console dial |
| 10:52.23 | fcois93 | kaldemar: No such command 'console dial' |
| 10:52.45 | kaldemar | then you're lacking the needed module |
| 10:53.05 | fcois93 | kaldemar: how to add it? |
| 10:53.15 | kaldemar | how did you install asterisk? |
| 10:53.19 | fcois93 | kaldemar: which module ? |
| 10:53.28 | fcois93 | kaldemar: I compiled it :) |
| 10:53.33 | romb_work | modules.conf |
| 10:53.50 | romb_work | load => app_dial |
| 10:53.56 | kaldemar | romb_work: no |
| 10:54.24 | romb_work | yes =) |
| 10:54.31 | romb_work | or autoload=yes in other way |
| 10:54.46 | kaldemar | fcois93: enable chan_console and either chan_oss or chan_alsa in make menuselect and recompile |
| 10:54.56 | kaldemar | romb_work: no, that's not the module. |
| 10:55.25 | romb_work | ok |
| 10:55.42 | romb_work | kaldemar maybe you can help me |
| 10:55.45 | romb_work | ? |
| 10:55.52 | kaldemar | depends on your issue |
| 10:55.56 | romb_work | i cant get/set callerid from calling peer in asterisk 1.6 |
| 10:56.00 | romb_work | i'm trying to use ${CALLERID(num)} |
| 10:56.04 | romb_work | but it's empty |
| 10:56.15 | Chainsaw | fcois93: You will have to use the originate command instead. |
| 10:56.23 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 10:56.23 | Chainsaw | fcois93: The dial command has been removed in Asterisk 1.6 |
| 10:56.59 | kaldemar | Chainsaw: not in my 1.6.0, 1.6.1 or 1.6.2 :P |
| 10:57.45 | *** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 11:02.01 | fcois93 | ok |
| 11:03.07 | *** join/#asterisk chendy (n=chatzill@119.139.171.37) |
| 11:05.10 | romb_work | what variable is used in asterisk 1.6.1 for caller id ? |
| 11:05.50 | Faustov | Any idea why -U username command line option prevents asterisk (1.6.1.10) from starting? Without otions or with any other options it starts and forks into bckground no problems, but with -U it dies right after startup without any errors |
| 11:06.14 | kaldemar | romb_work: func CALLERID, just like you assumed |
| 11:06.29 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 11:06.48 | romb_work | hmmm.... something wrong |
| 11:07.50 | kaldemar | show your config and CLI output if you want another pair of eyes |
| 11:09.54 | romb_work | Faustov check your file privileges |
| 11:10.30 | *** part/#asterisk Moc (n=mochouin@64.235.217.36) |
| 11:10.35 | romb_work | kaldemar you mean full config ? |
| 11:10.50 | kaldemar | the relevant dialplan config |
| 11:11.28 | Faustov | romb_work: that's correct tho :( |
| 11:11.39 | romb_work | '_8914XXXXXXX' => 1. Set(CALLERID(name)=Asterisk PBX) [pbx_config] |
| 11:11.40 | romb_work | <PROTECTED> |
| 11:11.55 | romb_work | but name does nto changes |
| 11:11.59 | romb_work | also |
| 11:12.32 | kaldemar | use a pastebin |
| 11:14.22 | romb_work | this only one lines more |
| 11:14.24 | romb_work | 2. VoicemailMain(${CALLERID(num)}@default) [pbx_config] |
| 11:15.34 | romb_work | returns only @default |
| 11:15.52 | romb_work | but if i call from this phone i see caller id |
| 11:17.27 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 11:17.37 | kaldemar | romb_work: you haven't shown anything that would set CALLERID(num). |
| 11:22.23 | romb_work | _8914XXXXXXX' => 1. Set(CALLERID(name)=Asterisk PBX) |
| 11:22.26 | romb_work | Set |
| 11:26.02 | *** join/#asterisk DJ_HaMsTa (n=werwer@c-69-136-253-186.hsd1.nj.comcast.net) |
| 11:44.03 | kaldemar | romb_work: you're setting name but trying to use num |
| 11:44.28 | kaldemar | romb_work: they are different fields |
| 11:49.34 | romb_work | i know |
| 11:49.57 | romb_work | i'm seting both name and num by callerid= in sip config |
| 11:54.12 | kaldemar | show it and show a CLI output |
| 11:54.27 | kaldemar | CLI output of a call. and use a pastebin. |
| 11:54.29 | kaldemar | ~pb |
| 11:54.30 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 12:12.01 | *** join/#asterisk Omorika (n=omorika@89.201.165.226) |
| 12:18.28 | romb_work | kaldemar my problem is module func_callerid not loaded. solved |
| 12:30.20 | *** join/#asterisk wonderworld (n=ww@62.143.22.226) |
| 12:31.20 | *** join/#asterisk k4mi1 (n=qmpelkam@xdsl-1545.wroclaw.dialog.net.pl) |
| 12:31.28 | k4mi1 | hello |
| 12:32.54 | k4mi1 | I have short question: how to shutdown / no shutdown a PSTN TDM card? |
| 12:34.37 | k4mi1 | I have a TDM400P card |
| 12:36.43 | kaldemar | restart dahdi/zaptel |
| 12:37.02 | k4mi1 | so simple :) |
| 12:49.51 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 12:51.36 | *** join/#asterisk Malkor (n=marco@hlle-d9ba1867.pool.mediaWays.net) |
| 12:51.46 | arossouw | is it better to use Zap/g1/${EXTEN} or Zap/r1/${EXTEN} in dialplan? |
| 12:52.27 | kaldemar | use whatever suits you |
| 12:54.31 | arossouw | if you have a channel with group 1 for example and you have 3 isdn connections, will it roundrobin the isdn channels? |
| 12:54.43 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 12:55.06 | arossouw | for example channel => 7-8,10-11 |
| 12:55.34 | arossouw | does it try all the lines? |
| 12:56.10 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
| 12:56.44 | kaldemar | g will take first available channel, starting from first one, r will round-robin from first to last. |
| 12:58.18 | kaldemar | arossouw: see "Dialing a Group" on this page: http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels |
| 12:58.55 | kaldemar | the same applies for DAHDI also |
| 13:01.54 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 13:01.55 | arossouw | thanks |
| 13:02.23 | *** join/#asterisk bala (i=dce39c91@gateway/web/freenode/x-iunzddoiorgliccs) |
| 13:02.25 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 13:04.00 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:04.20 | *** join/#asterisk Iamnacho (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 13:05.29 | garymc | Hi all, anytime i call a number that is out of order or doesnt exist anymore i get a generic asterisk voice message "all circuits are busy now" instead of my actual BT error message. This can become confusing as i get the same message if my dial plan is wrong. Is there anyway to stop the asterisk message and let my Telco providor message come through instead? |
| 13:07.43 | [TK]D-Fender | garymc: You have a PRI... BT doesn't put audio on the line... PRI = call progress. I'd say "make your own message" to match the result code... but we both know what the answer to that is... |
| 13:08.31 | garymc | [TK]D-Fender : Is the answer errr fcuk it? |
| 13:10.19 | *** part/#asterisk agx (n=Antonio@88.34.216.63) |
| 13:15.59 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 13:16.32 | doolittlework | hi there anyone worked with b410p usinf misdn.conf? |
| 13:18.01 | *** join/#asterisk Wired_Life (n=Chatzill@mgdb-4db8d9c9.pool.mediaWays.net) |
| 13:22.38 | *** join/#asterisk Iamnacho (n=Iamnacho@98.186.180.143) |
| 13:30.14 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:30.34 | [TK]D-Fender | garymc: Answer is "Don't like how FreePBX works? Ask someone who CARES :p" |
| 13:32.30 | *** join/#asterisk telnettech (n=telnette@71.5.32.196) |
| 13:34.56 | doolittlework | hi TK |
| 13:35.46 | beek | mornin' [TK]D-Fender |
| 13:36.15 | *** part/#asterisk Omorika (n=omorika@89.201.165.226) |
| 13:37.30 | *** join/#asterisk qdk (n=qdk@193.169.208.32) |
| 13:38.12 | *** join/#asterisk TommyBotten (n=noosjent@fcnoos-fw03.freecode.no) |
| 13:40.38 | garymc | get on this link guys! http://www.youtube.com/watch?v=Yj2oXMdZ4sk I used to work for BT in the same call centre and actually know the guy who made this call. The guy he phones expresses how i feel about BT at present. Watch / l isten to it its great. |
| 13:40.57 | garymc | The guy making the call is called peter brookes. I used to play football with him about 12 years ago |
| 13:45.09 | *** join/#asterisk oej (n=olle@eduroam-193-157-113-208.uio.no) |
| 13:45.46 | garymc | anyone listen to it? |
| 13:46.50 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:47.44 | beek | garymc: yep |
| 13:47.55 | garymc | good? |
| 13:48.12 | beek | There is obviously some history there between the callee and BT |
| 13:50.06 | *** join/#asterisk plundra (i=404@article.se) |
| 13:50.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 13:55.47 | garymc | yep.... |
| 13:55.51 | garymc | cold callers suck |
| 13:59.53 | plundra | Uhm, how can I get information about queues via the AMI for example, leaving me with the complete names of the queues? |
| 14:00.14 | plundra | Doing Action: Queues gives me the output of "queues show" in the CLI pretty mutch, where queue-names are cut off. |
| 14:01.29 | plundra | Or to put it in another way; what would be the best way to interact with Asterisk, when I want to 1) list queues 2) list their members, 3) add/pause/remove members? |
| 14:03.16 | plundra | Oh, QueueStatus was better :) Great. |
| 14:04.29 | sun28 | AMI |
| 14:04.37 | sun28 | perl/expect |
| 14:05.37 | plundra | Yeah I'll stick with AMI :-P Just needed to find the right command. |
| 14:06.07 | plundra | (And I'll stick with python, but except seems like a nice way to interacet :)) |
| 14:06.24 | *** join/#asterisk mort_gib (n=mjensen@83.36.63.16) |
| 14:12.38 | *** part/#asterisk romb_work (i=romb@89.28.249.41) |
| 14:17.53 | *** join/#asterisk jkroon (n=jkroon@41.244.47.84) |
| 14:17.54 | Katty | peeks in |
| 14:17.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:18.08 | *** join/#asterisk jkroon (n=jkroon@196.35.70.235) |
| 14:18.33 | jkroon | hi guys, is there anybody here that would be able to assist me with the sip auth checking procedure? |
| 14:18.54 | Katty | infobot: ask? |
| 14:18.55 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:19.04 | jkroon | it seems the issue from yesterday might be a legit asterisk bug not setting stale=true. |
| 14:19.46 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 14:19.50 | sehh | hey people |
| 14:20.43 | Katty | hi |
| 14:20.47 | jkroon | the sequence seems to be c:REGISTER, s:SIP401, c:REGISTER (with auth this time), s:SIP/200 OK. After this the client sends an INVITE (same nonce+auth as second REGISTER), which elicits another SIP/401 Unauthorized, without the state=true flag. |
| 14:20.57 | sehh | q: i've got a problem with DTMF detection. My two asterisk PBX's are both using ISDN lines (mISDN with Beronet cards), the problem is that they can't detect each others DTMF, while they can detect DTMF from other phones. Any ideas? |
| 14:21.18 | jkroon | they are argueing that this second SIP/401 should have stale=true set. |
| 14:22.19 | ManxPower-work | jkroon: have you checked out the pedantic= option? I dunno if that will help, but you should look at it. |
| 14:22.51 | jkroon | ManxPower-work, no i haven't. can i set it for a single client only? |
| 14:23.04 | Katty | jkroon: why don't you google and find out! |
| 14:23.20 | jkroon | already busy but often the reply is faster from the # :) |
| 14:23.20 | ManxPower-work | jkroon: All I know is it has something to do with being more strict in SIP processing. |
| 14:23.31 | jkroon | reads |
| 14:23.58 | jkroon | also Katty - the info google finds on asterisk is usually relatively outdated. |
| 14:23.58 | ManxPower-work | I've never had to use it, I just know about it because I read the .sample config files in every major release of Asterisk. |
| 14:24.23 | jkroon | Enable slow, pedantic checking of Call-ID:s, multiline SIP headers and URI-encoded headers. |
| 14:24.33 | jkroon | in other words, back to the source to see what it really does. |
| 14:24.42 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:24.42 | ManxPower-work | sehh: if you were using standard zaptel/dahdi I'd say set the toneduration=500 in /etc/asterisk/zapata.conf or /etc/zaptel.conf |
| 14:24.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:25.07 | sehh | ManxPower-work, not using those, just mISDN |
| 14:25.07 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 14:28.56 | madduck | why would a client connected to asterisk with nat=yes properly send the external IP4 in SIP packets, e.g. |
| 14:28.59 | madduck | Via: SIP/2.0/UDP 84.74.106.21:1030;branch=z9hG4bK2skbvhl122tlounr94k8jqb;rport |
| 14:29.04 | madduck | and |
| 14:29.04 | madduck | o=Nokia-SIPUA 63427418573281625 63427418573281625 IN IP4 84.74.106.21 |
| 14:29.22 | madduck | but if the same client connects via TCP, then the internal address appears |
| 14:29.58 | madduck | Via: SIP/2.0/TCP 192.168.14.98:5060;branch=z9hG4bK48qr2a5mamq4sklpth3tlj3;rport |
| 14:30.05 | madduck | o=Nokia-SIPUA 63427418777654375 63427418777654375 IN IP4 192.168.14.98 |
| 14:30.24 | madduck | is linux conntrack not capable of SIP/tcp? |
| 14:30.53 | madduck | or does conntrack not actually change the SIP packets? (I think it must) |
| 14:31.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 14:31.26 | *** join/#asterisk [gnubie] (n=bintut@119.56.59.38) |
| 14:31.40 | [gnubie] | waves |
| 14:32.06 | [gnubie] | wonders if tzafrir is around |
| 14:32.26 | tzafrir | is around |
| 14:32.49 | ManxPower-work | madduck: you don't want nat=yes with any client side NAT support. |
| 14:33.07 | ManxPower-work | It frequently breaks stuff. In your case, this might be an exception |
| 14:33.09 | *** join/#asterisk af_ (n=getsmart@88.149.240.33) |
| 14:33.36 | [gnubie] | nice! ;) |
| 14:33.46 | madduck | ManxPower-work: i still don't quite understand what nat=yes does, but the firewall-side SIP support is necessary to let packets pass |
| 14:33.55 | madduck | i don't actually know if the SIP packets are rewritten |
| 14:34.02 | [gnubie] | tzafrir: please help. i'm having an error in building asterisk. kindly check => http://paste.debian.net/52380/ |
| 14:34.05 | madduck | i don't think they are |
| 14:34.59 | ManxPower-work | madduck: nat=yes mostly just tells Asterisk to ignore the address/port info in the SIP packet and just use the info from the packet header. I'm sure it does more, but that's the most important thing. |
| 14:35.41 | jkroon | madduck, no, i don't think the linux kernel module handles SIP/TCP. |
| 14:36.03 | madduck | jkroon: so for SIP/udp, the kernel modules actually change the SIP packets? |
| 14:36.09 | jkroon | yes. |
| 14:36.29 | jkroon | based on what I recall from my perusal of that particular code about a year back. |
| 14:36.35 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:36.41 | ManxPower-work | the SIP client could use STUN to find their external address as well |
| 14:36.58 | madduck | that would explain the behaviour i am seeing: with nat=yes and nat=no, the UDP client can do everything, the TCP client sees no audio |
| 14:37.14 | snadge | on an unrelated note.. i just went to an information evening at the freemasons grand lodge.. and it was excellent.. im petitioning to join tomorrow :P |
| 14:37.34 | ManxPower-work | I consider Asterisk's TCP support to be experimental |
| 14:37.41 | madduck | what confuses me is the RTP stream according to tcpdump: |
| 14:37.47 | madduck | 15:37:21.794678 IP 77.109.139.86 > 84.74.106.21: ICMP 77.109.139.86 udp port 11074 unreachable, length 208 |
| 14:37.47 | tzafrir | hmm... |
| 14:37.59 | madduck | why does the 77 host, the server, tell the client about an unreachable port? |
| 14:38.27 | madduck | ManxPower-work: sure, same, but there are more and more android clients, and sipdroid is so fucking broken that it can only do TCP |
| 14:38.34 | voipmonk | so it can pick another random port unless you have specified one |
| 14:38.45 | madduck | voipmonk: huh? |
| 14:38.53 | voipmonk | good lord... |
| 14:39.01 | ManxPower-work | madduck: Ah. In my world All Softphones Suck. I never touch them. |
| 14:39.03 | voipmonk | hides back under his rock.... its much too early for this |
| 14:39.12 | Katty | hugs voipmonk |
| 14:39.13 | snadge | call me a noob.. but i installed sipdroid on my phone and it "just worked" when talking to my asterisk server? |
| 14:39.24 | Katty | puts a DND sign in front of voipmonk's rock |
| 14:39.31 | madduck | snadge: can you receive incoming calls? |
| 14:39.34 | voipmonk | chuckles in his sleep |
| 14:39.40 | madduck | snadge: i bet you cannot, because sipdroid does not listen on 5060/udp |
| 14:39.47 | madduck | everything else works though, outgoing is fine |
| 14:39.48 | snadge | madduck: oh, i only tried to make outgoing calls |
| 14:39.51 | ManxPower-work | Katty: it's not a rock, it's an alternative housing design |
| 14:39.51 | Katty | voipmonk: having a rough morning myself. |
| 14:39.54 | madduck | exactly |
| 14:39.55 | angryuser | I am searching a gateway with a stable T.38 / FEC /ECM support (carrier grade) any ideas what should i choose ? |
| 14:39.57 | snadge | thats a real bummer! |
| 14:40.14 | madduck | snadge: http://code.google.com/p/sipdroid/issues/detail?id=181 |
| 14:40.18 | Katty | ManxPower-work: he can call it whatever he wants. |
| 14:40.21 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 14:40.22 | ManxPower-work | angryuser: you and thousands of other disapointed people. |
| 14:40.50 | snadge | madduck: without reading the bug detail.. can the issue be resolved? |
| 14:40.56 | [gnubie] | tzafrir: any comments? |
| 14:41.06 | angryuser | ManxPower-work, any useful ideas ? |
| 14:41.26 | Katty | http://www.bobevans.com/downloads/BlackFriday10_FreePumpkinPie.pdf <- free pumpkin pie at Bob Evans |
| 14:41.34 | madduck | snadge: theoretically, sipdroid is open-source. |
| 14:41.57 | jojo^ | For the active channel, the SIP callid is stored in the variable SIPCALLID. Is there some similiar way to get the SIP callid from a Dial()-call that just ended? |
| 14:42.07 | snadge | yeah but sipdroid is presumably written by people who know a lot more about how it works.. and how to fix it |
| 14:42.23 | *** join/#asterisk ajohnson (n=aaron@65.122.4.130) |
| 14:42.41 | snadge | than people who would have to invest considerably more effort and time.. just to get to that level to be able to look at the problem and solve it ;) |
| 14:43.02 | madduck | snadge: i am sure if you waved money into their faces, someone might implement it. ;) |
| 14:43.04 | ManxPower-work | hugs his Polycom 550 |
| 14:43.09 | madduck | it should not be hard at all |
| 14:43.17 | madduck | udp client: |
| 14:43.17 | madduck | <PROTECTED> |
| 14:43.17 | madduck | <PROTECTED> |
| 14:43.22 | madduck | tcp client: |
| 14:43.23 | madduck | <PROTECTED> |
| 14:43.23 | madduck | <PROTECTED> |
| 14:43.37 | ManxPower-work | DO NOT FLOOD! |
| 14:43.43 | voipmonk | jojo^ u can store all the info in a new set of variables or u can have it write them into a mysql db .... but once the call is gone you're not going to get info from asterisk on it anymore :) |
| 14:43.45 | madduck | ManxPower-work: that was not flooding. |
| 14:43.46 | madduck | so two things: first, linux conntrack cannot handle tcp it seems |
| 14:43.47 | angryuser | madduck snadge , fring is out for android too (not gpl) but the only soft with skype support |
| 14:44.01 | madduck | angryuser: fring requires all calls to be routed through the US |
| 14:44.10 | snadge | yeah but sipdroid can talk to skype, via asterisk |
| 14:44.19 | voipmonk | madduck: pastebin.ca is your friend, or u may just get kicked |
| 14:44.20 | angryuser | madduck, nope you got sip account for any provider there |
| 14:44.24 | madduck | and second, asterisk has no way to reliably deal with NATted clients without firewall support, at least for TCP it seems. |
| 14:44.36 | madduck | voipmonk: in all channels I am used to flooding starts with 6 lines |
| 14:44.48 | voipmonk | its a lil different in here |
| 14:44.58 | madduck | where are the channel guidelines? |
| 14:45.01 | kaldemar | ~pb |
| 14:45.02 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:45.04 | voipmonk | since the debug is so very long , and we dont want the n00bs following your footsteps |
| 14:45.20 | jojo^ | voipmonk, So I somehow need to send the dial into a part of the dialplan and do some getting/setting there, instead of just letting a plain Dial() bridge the new channel with the current channel? |
| 14:45.28 | voipmonk | 6 leads to 8 and 8 leads to 400 |
| 14:45.44 | madduck | well sorry, but hardly a reason to bite my head off now. ;) |
| 14:46.02 | voipmonk | now you're asking a question I can work with, jojo^ |
| 14:46.29 | snadge | so why does sipdroid use tcp in the first place? |
| 14:46.34 | voipmonk | jojo^ , bridging a current channel? Tell me, what are you trying to accomplish |
| 14:47.14 | tzafrir | [gnubie], It'll take some work for me to adjust those patches. I'll try to take a look at it later |
| 14:47.47 | snadge | using sipdroid over my 3g connection probably violates my carriers terms of service :P |
| 14:48.17 | snadge | but i choose to remain ignorant of this fact |
| 14:48.26 | jojo^ | voipmonk, Well, I've got an inbound call, which I "forward" to a new channel by doing a plain "Dial" and, in the case of an answer, letting asterisk automatically "bridge" the channel who caused the Dial with the newly created channel (the answered one). |
| 14:48.34 | voipmonk | until they google irc chatrooms for your nick :) |
| 14:48.53 | snadge | well i said probably.. i would have to read it to find out ;) |
| 14:49.38 | jojo^ | voipmonk, So just an extension with a Dial. Now I want to do some postprocessing after the Dial()-ed channel hangs up. An one of those depends on me knowing the SIP callid of the channel created with Dial() |
| 14:50.05 | voipmonk | so u need h, |
| 14:50.27 | voipmonk | checkout the wondrous "h" |
| 14:50.41 | kaldemar | or option g for Dial |
| 14:50.51 | madduck | so my explanation for http://slexy.org/view/s2NnkN4C3u, which repeats ad infinitum and there are never packets in the other direction is that since this is an Echo() call, remote does not have any audio to send and so does not. |
| 14:51.00 | ManxPower-work | wouldn't all those things give you the CALLING channel info, not the CALLED channel info? |
| 14:51.21 | madduck | the audio never arrives because port 11074/udp is filtered by the packet filter since nf_conntrack_sip did not extract the port from the SIP/tcp packets |
| 14:51.26 | [gnubie] | tzafrir: what do you suggest now for me? |
| 14:51.49 | tzafrir | try adjusting the patch yourself :-( |
| 14:52.17 | [gnubie] | tzafrir: which part? |
| 14:53.19 | [gnubie] | tzafrir: or, can i just disable this h323 from building since i don't have a need for it? |
| 14:53.37 | tzafrir | to see where exactly the patch fails: patch -p1 <debian/pathes/h323-no-deps-on-asterisk --dry-run |
| 14:53.47 | tzafrir | don't forget the --dry-run |
| 14:54.05 | voipmonk | very good [gnubie] |
| 14:54.22 | jkroon | ok, looks like i'm cooking a patch for asterisk's SIP authentication tonight. |
| 14:54.23 | voipmonk | remove unnessentials to simplify your asterisk deployment, what a wonderful idea. |
| 14:54.30 | voipmonk | .... |
| 14:54.34 | tzafrir | The breakage is in the makefile, so it is likely a matter of adjustment |
| 14:54.34 | jojo^ | voipmonk, Well, the trouble is not doing something upon hangup, but what to run in lets say 'h' to get the SIP callid. |
| 14:54.37 | jkroon | may solve my SNOM problem too ... |
| 14:55.17 | tzafrir | [gnubie], http://svn.debian.org/viewsvn/pkg-voip/asterisk/trunk/debian/patches/h323-no-deps-on-asterisk?view=log |
| 14:55.20 | jojo^ | kaldemar, Thank you. So g stays in the context after dial. Interesting :) |
| 14:57.50 | [TK]D-Fender | jojo^: Only if they are the ones who hung up |
| 14:59.35 | *** join/#asterisk TheDavidFactor-H (n=chatzill@c-68-34-116-180.hsd1.md.comcast.net) |
| 14:59.37 | [gnubie] | tzafrir: anyway to just disable h323 from "debuild -us -uc"? |
| 15:00.09 | jojo^ | ManxPower-work, Do you have an idea of what I could do to get the CALLED channel info? |
| 15:00.35 | *** join/#asterisk bowenp (n=root@75.138.246.167) |
| 15:01.16 | tzafrir | You can remove that patch regardless. With it you get pointless dependency on openh323 even if you don't install the separate openh323 module |
| 15:01.32 | ManxPower-work | jojo^: no idea whatsoever. I don't even know if it's possible. |
| 15:01.47 | tzafrir | if you don't have disk-space issues, just install it |
| 15:01.52 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 15:01.53 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195) |
| 15:02.09 | ManxPower-work | jojo^: Asterisk isn't a SIP proxy, it's a B2BUA. You may need a proxy to get the info you need (OpenSER, etc) |
| 15:02.11 | bowenp | Hi, need advise on hardware, I'm starting from scratch for my new company. |
| 15:02.19 | [gnubie] | tzafrir: but the problem here is i cannot build asterisk because of that h323 |
| 15:02.26 | tzafrir | Alternatively, tweak the package to disable oepnh323 (--without-pwlib, and several places that refer to chan_h323.so) |
| 15:02.43 | ManxPower-work | bowenp: We laugh at people that ask that question |
| 15:02.46 | tzafrir | the patch can be disabled in debian/patches/series |
| 15:03.43 | [gnubie] | tzafrir: you mean, i just comment out all the h323 lines on debian/patches/series file? |
| 15:03.59 | bowenp | hay, I'm a newbie....looking to go analog with a 4 port card, what is out there besides an Digium that is compatable? |
| 15:04.06 | madduck | happens to know the nf_conntrack_sip maintainer and sent him an email |
| 15:04.08 | madduck | \o/ |
| 15:04.47 | jojo^ | ManxPower-work, Thanks for the realistic answer. We've got a kamailio in front, I'll see if we can figure out some other way to accomplish what we are trying. |
| 15:05.08 | bowenp | The other issue is fax out capablity |
| 15:05.29 | kaii | mh .. got a strange problem .. on one of our boxes (1.4.27) VoiceMails() are not written to disk .. there is no email address defined, and in the debug log i do not see any "sendmail" calls. i see in verbose that the voice is recorded to a temporary file ... but after the call, INBOX is empty. |
| 15:06.01 | kaii | any idea where i should pay attention? i'm stuck, configuration seems okay (after triple checking) |
| 15:06.29 | ManxPower-work | kaii: check available disk space. |
| 15:06.42 | kaii | 4% used |
| 15:06.59 | TheDavidFactor-H | bowenp, several, sangoma is one; I don't know anything about the fax side |
| 15:07.26 | kaii | ManxPower-work: if disk was full, i had other problems than voicemail :) |
| 15:07.28 | ManxPower-work | the most reliable faxing is done over a POTS line direct to the telco. |
| 15:08.01 | ManxPower-work | I work for a CLEC that does mostly VoIP and even we put in POTS lines for customer faxing. |
| 15:08.43 | kaii | agreed, faxing via voip is messy |
| 15:09.53 | pif | tzafrir: hi? |
| 15:10.02 | bowenp | yes manxpower, pots is the way to go on that side, I was hoping to have a way to call in, and request document numbers to be faxed back if internet connective is not available on site. |
| 15:10.23 | bowenp | and fax is at the client site. |
| 15:10.27 | tzafrir | pif, hello |
| 15:10.49 | pif | tzafrir: hello, you know if 4bri's are supported in latest dahdi? |
| 15:10.54 | fcois93 | hello all |
| 15:10.55 | tzafrir | yes |
| 15:10.58 | tzafrir | it is |
| 15:11.01 | fcois93 | need help for diversion problem |
| 15:11.07 | pif | tzafrir: even for 1.4.x ? |
| 15:11.39 | fcois93 | if I look at the draft-levy-sip-diversion-08.txt, asterisk dont generate diversion at good time |
| 15:12.15 | [gnubie] | tzafrir: do i still need the zaptel-source if i already installed dahdi-source, dahdi-linux and dahdi-tools ? |
| 15:12.26 | tzafrir | well, supported at the dahdi level. You need to have a proper ISDN stack. With asterisk 1.4 it's either bristuffed asterisk or a patched asterisk 1.4 |
| 15:12.28 | pif | [gnubie]: nope |
| 15:12.36 | Faustov | Ok i found out why asterisk would not start with -U, could not enter /var/log/asterisk dir - maybe an error message would be in place? |
| 15:12.52 | tzafrir | [gnubie], you can use either zaptel or dahdi. it's a build-time option |
| 15:13.05 | [gnubie] | pif: i see.. |
| 15:13.47 | madduck | i seem to have a channel that is unused which I cannot hangup or remove |
| 15:13.47 | madduck | SIP/c450ip-0293dad8 13@phones:50011 Up Dial(SIP/piper,"30","TtKkXxU(c |
| 15:13.51 | madduck | i already did |
| 15:13.54 | madduck | channel request hangup |
| 15:13.57 | madduck | but without success |
| 15:14.01 | madduck | how can I get rid of this? |
| 15:14.09 | pif | tzafrir: ok, so plain vanilla non-patched support is in 1.6 only |
| 15:14.32 | tzafrir | right |
| 15:14.37 | [gnubie] | tzafrir: but when tried to 'debuild -us -uc' the asterisk-1.4.21.2~dfsg , it's looking for zaptel-source.. that's also because zaptel-source is defined in the debian/control file |
| 15:14.45 | *** part/#asterisk bowenp (n=root@75.138.246.167) |
| 15:15.42 | tzafrir | technically that's because it's written in the build-depends field in debian/control |
| 15:16.16 | pif | tzafrir: and do your 1.6 debian packages have that support, including dahdi-source? |
| 15:16.32 | pif | at the lenny level, or only sid? |
| 15:17.58 | *** join/#asterisk jpcansa (n=jpbenavi@200.91.99.148) |
| 15:18.03 | Amorsen | Weird: If I do *8 to catch a call which has t option set, my call inherits the t option and I can transfer the call by DTMF |
| 15:18.31 | Amorsen | If the call doesn't have the t option set, I can't transfer the call by DTMF |
| 15:20.33 | ManxPower-work | Amorsen: correct |
| 15:21.40 | pif | tzafrir: current dahdi-source doesn't build with 2.6.31 (which has a big security fix), any pending issue? |
| 15:21.44 | tzafrir | pif, it's a lenny backport |
| 15:21.46 | *** join/#asterisk Tagor (n=none@85.146.140.109) |
| 15:22.01 | Tagor | How can I route an incoming call to another extensions than default? |
| 15:22.12 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 15:22.20 | madduck | one way to fix my problem is by using a SIP proxy, which lets clients connect with TCP, but makes asterisk believe that they are speaking UDP |
| 15:22.23 | madduck | i wonder if this could work |
| 15:22.36 | tzafrir | the 2.2 branch builds. 2.2.1-rc1 was already tagged. Once it will be released I'll build packages from it |
| 15:22.58 | tzafrir | should fix all the newer kernel build issues |
| 15:23.12 | madduck | tzafrir: xorcom.com DNS seems broken |
| 15:23.17 | tzafrir | though IIRC I also backported those fixes to the debian packages |
| 15:23.21 | pif | tzafrir: ok, thanks for all the info |
| 15:23.46 | madduck | ns{1,2}.panelboxmanager.com do not answer |
| 15:23.47 | arossouw | i think tcp is better than udp, with udp if a packet gets dropped you have issues |
| 15:23.59 | TommyBotten | eh? |
| 15:24.11 | drmessano | lol |
| 15:24.32 | madduck | why would panelboxmanager.com put both their DNS into the same frickin' subnet? |
| 15:24.35 | madduck | idiots |
| 15:24.36 | pif | arossouw: and it's a good thing |
| 15:24.45 | [TK]D-Fender | Tagor: there is no such thing as "default" Calls go where you tell them to |
| 15:24.48 | TommyBotten | Each protocol has it's uses. It's hard to say that one is better than the other |
| 15:24.57 | *** join/#asterisk dandre (n=daniel@82.236.48.30) |
| 15:25.08 | madduck | arossouw: depends on what you are trying to do. with audio, it's better to lose a second of speech than to incur delays |
| 15:25.10 | TommyBotten | [TK]D-Fender: Sure there's a default: _. ;) |
| 15:25.20 | drmessano | TCP2 is gonna make TCP look like UDP and UDP look like NetBIOS, you know? |
| 15:25.38 | madduck | TCP2 only works with IPv9 though. |
| 15:25.49 | *** join/#asterisk muiro (n=muiro@173.89.177.15) |
| 15:25.54 | TommyBotten | TCP2 is not web2.0 compliant yet |
| 15:25.57 | arossouw | madduck: ok, that makes sense |
| 15:25.57 | pif | barelah |
| 15:25.57 | madduck | and will actually be a session layer on top of SPDY |
| 15:25.59 | drmessano | I can't run IPv9 on Windows 3.11 |
| 15:26.00 | [TK]D-Fender | TommyBotten: No, there isn't. |
| 15:26.13 | madduck | drmessano: you can't run any IP on windows 3.11 ;) |
| 15:26.15 | madduck | you need wfw ;) |
| 15:26.16 | *** join/#asterisk moy (n=moy@74.12.130.190) |
| 15:26.24 | pif | drmessano: not without Trumpet Winsock :) |
| 15:26.39 | pif | madduck: 3.11 is wfw |
| 15:26.48 | madduck | oh, true |
| 15:26.53 | drmessano | Yeah |
| 15:26.56 | madduck | shows what i know |
| 15:27.11 | drmessano | There was no "Windows 3.11" 3.11 was Workgroups |
| 15:27.20 | Tagor | [TK]D-Fender: in sip.conf I added 'context=test' but when I call to the number it goes to the context 'default'. Is there a way to have it go to context=test? |
| 15:27.29 | arossouw | really disliking grandstream now |
| 15:27.36 | tzafrir | madduck, now it resolves :-( |
| 15:27.40 | madduck | under what circumstances would nat=no not work? |
| 15:27.49 | pif | threw sold all his granstreams to a greek on ebay :) |
| 15:27.51 | drmessano | You wouldnt want no |
| 15:27.57 | madduck | tzafrir: true. consider getting a different backup DNS ;) |
| 15:27.57 | drmessano | no is not the same as never |
| 15:28.02 | drmessano | never is the real no |
| 15:28.08 | drmessano | no is sorta no |
| 15:28.15 | arossouw | wishes he could convince manager that grandstream is useless |
| 15:28.17 | madduck | drmessano: no means rfc something and that sounded like the best option |
| 15:28.21 | muiro | drmessano: remind me never to let you near my sister |
| 15:28.28 | pif | arossouw: they are crap |
| 15:28.50 | drmessano | lol |
| 15:28.51 | arossouw | pif: i know, and ive said so a hundred times, but i'm not going to argue |
| 15:29.01 | [TK]D-Fender | Tagor: * is clearly not matching the section enrty in sip.conf that the call is coming from otherwise it'd go where you told it |
| 15:29.02 | arossouw | pif: with my manager |
| 15:29.13 | [TK]D-Fender | Tagor: Good odds your auth it broken |
| 15:29.19 | drmessano | No has to do with some implementations shitty use of the RFC |
| 15:29.35 | drmessano | Unlike Asterisks shitty use of the RFC |
| 15:30.14 | tzafrir | pif, are you sure it doesn't build? checking.... |
| 15:30.17 | arossouw | i think a company needs to spend more on digital voip, buying cheap stuff results in alot of issues |
| 15:30.29 | madduck | tzafrir: when will you finally sign that repo? ;) |
| 15:30.33 | madduck | it's not /that/ much effort⦠|
| 15:30.47 | pif | tzafrir: I have 1:2.2.0.2~dfsg-1 |
| 15:30.56 | tzafrir | I didn't test those packages. I don't trust them |
| 15:30.59 | *** join/#asterisk UQlev (n=yuriy@87.228.199.125) |
| 15:31.14 | pif | oh, they're not yours? |
| 15:31.18 | tzafrir | I sign them when I test them |
| 15:31.34 | madduck | oh lovely |
| 15:31.36 | madduck | i just upgraded ;) |
| 15:31.40 | pif | tzafrir: should I use your private repo ? |
| 15:31.41 | madduck | no, i mean signing the release files |
| 15:31.53 | tzafrir | pif, on Lenny? |
| 15:32.09 | pif | both sid and lenny |
| 15:32.23 | arossouw | just curious, how does digium's asterisk software release cycle work, ..or is that confidential |
| 15:33.32 | *** join/#asterisk thenthenio (n=mauro@217.221.242.43) |
| 15:34.55 | *** join/#asterisk TimToady_ (n=moi@79.103.63.13.dsl.dyn.forthnet.gr) |
| 15:35.13 | thenthenio | Hello! |
| 15:35.48 | thenthenio | I'm not able to register my X-lite softphone to asterisk |
| 15:36.10 | arossouw | whats the error messagE? |
| 15:36.37 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:36.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:37.04 | Tagor | [TK]D-Fender: I get this error: Call from 'othernumber654321' to extension '1234567' rejected because extension not found. .... could you have a look at my config? http://www.nopaste.nl/?paste=1539 I don't understand what's wrong |
| 15:37.10 | thenthenio | I see X-lite says Error 408 Request timeout |
| 15:38.03 | [TK]D-Fender | Tagor: Just like it says, you don't have an EXTENSION to match 1234567 |
| 15:39.18 | arossouw | thenthenio: check the port and hostname |
| 15:39.30 | Wired_Life | noanswer playback dosent work with my fritz isdn card... any one an idea? |
| 15:39.40 | muiro | thenthenio: are you getting any messages on Asterisk about the register? You may need to turn sip debugging on. |
| 15:40.03 | Tagor | [TK]D-Fender: but the extension '[1234567]' in extensions.conf IS the extension, isn't it? |
| 15:40.21 | [TK]D-Fender | Tagor: no, that is a SIP DEVICE |
| 15:40.31 | thenthenio | I saw (wireshark) the client sent a SIP packet and asterisk answers with an ICMP packet!!! |
| 15:40.35 | [TK]D-Fender | Tagor: extension = EXTENSIONS.CONF |
| 15:41.46 | thenthenio | muiro: ICMP destination unreachable (port unreachable)!!! |
| 15:42.43 | pif | tzafrir: /usr/src/modules/dahdi/drivers/dahdi/wctc4xxp/base.c:776: error: 'struct net_device' has no member named 'set_multicast_list' |
| 15:43.11 | arossouw | thenthenio: you dont maybe have a firewall enabled? |
| 15:44.14 | *** join/#asterisk kudos4421 (n=kudos442@users.misterart.com) |
| 15:44.20 | thenthenio | I assume on server part no, given that it is a AsteriskNOW fresh setup, on the client I disabled the Windoz firewall! |
| 15:44.20 | arossouw | thenthenio: try nmap -sU -p 5060 destinationip , assuming the sip peer is listening on udp port 5060 |
| 15:44.21 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:44.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:45.18 | thenthenio | arossouw: form which host? |
| 15:45.29 | *** join/#asterisk yoshx (n=yoshx@67.46.119-80.rev.gaoland.net) |
| 15:45.31 | arossouw | thenthenio: the X-lite client |
| 15:45.56 | thenthenio | X-lite client is WIndoz... |
| 15:46.07 | *** join/#asterisk |Cybex| (n=John@212.178.82.20) |
| 15:47.12 | arossouw | not sure how asteriskNow works, but login to asteriskNow and issue asterisk -rx "sip show peers" |
| 15:47.27 | [gnubie] | gtg now.. thanks tzafrir.. ;) |
| 15:47.27 | arossouw | is your sip.conf set correctly? |
| 15:47.30 | [gnubie] | waves |
| 15:47.32 | *** part/#asterisk [gnubie] (n=bintut@119.56.59.38) |
| 15:48.05 | Tagor | [TK]D-Fender: I changed it, now I have this: http://www.nopaste.nl/?paste=1540 but I get the same error |
| 15:48.31 | thenthenio | arossouw: 5060/udp closed sip |
| 15:48.36 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-116-094.pools.arcor-ip.net) |
| 15:48.56 | arossouw | thenthenio: i guess you havent set your sip.conf or you havent reloaded asterisk |
| 15:49.06 | [TK]D-Fender | Tagor: pastebin a call with SIP DEBUG enabled |
| 15:50.03 | Blackvel | is there nowadays a standard for voip numbers in the US? e.g to contact big software vendors? what services do they use? is there a central numbering service (e.g for hidden direct calls) or is everyone still pinned to individual providers and non-standard direct ip addresses? |
| 15:50.29 | thenthenio | I just started AsteriskNOW and interact with FreePBX I don't know what to set in sip.conf... |
| 15:50.57 | thenthenio | I supposed FreePBX GUI did it for me! |
| 15:50.59 | yoshx | hello |
| 15:51.11 | thenthenio | arossouw: ^ ^ |
| 15:51.18 | arossouw | thenthenio: refer to http://www.voip-info.org/wiki/view/sip.conf |
| 15:51.22 | kaldemar | thenthenio: then #freepbx and #asterisknow are the channels for you |
| 15:52.02 | arossouw | or try switchvox :-) |
| 15:53.55 | madduck | tzafrir: musiconhold only works for some select clients now |
| 15:54.23 | madduck | if the e71 holds, twinkle hears moh, but if twinkle holds, the e71 doesn't. |
| 15:56.33 | tzafrir | hmmm... building the package with 2.6.32-rc8 fails: /home/tzafrir/Proj/Alioth/pkg-voip/build-area/usr_src/modules/dahdi/drivers/dahdi/dahdi-base.c:839: error: âTASK_INTERRUPTIBLEâ undeclared (first use in this function) |
| 15:56.43 | tzafrir | I'll look into it later |
| 15:57.54 | arossouw | is it possible to configure asterisk in way, that after business hours (certain timeframe) asterisk leaves a message to inbound calls? |
| 15:58.21 | madduck | arossouw: of course, look into advanced dialplans on voip-info.org |
| 15:58.30 | arossouw | ok, great |
| 15:59.09 | arossouw | these people are calling after hours, i'll have to figure out how to stop them |
| 15:59.35 | arossouw | theyre probably calling boyfriends |
| 15:59.59 | arossouw | damm , its difficult to control phone abuse |
| 16:00.05 | thenthenio | what is the relationship between Asterisk and FreePBX in AsteriskNOW? |
| 16:00.08 | madduck | use mixmonitor to record all talks and automatically send them to all@yourcompany |
| 16:00.19 | arossouw | lol, thats an idea |
| 16:01.03 | kaldemar | thenthenio: asterisknow is an operating system with asterisk and freepbx among other software installed, and freepbx is a GUI for asterisk. |
| 16:01.14 | arossouw | can compannies legally , prevent employees from calling after hour? |
| 16:01.31 | drmessano | Create a wav file of you way back in the background telling someone "come on baby, I want you again" like you're urging someone to get off the phone, and play it on the channel after 2 minutes when the time is after a specific hour |
| 16:01.47 | arossouw | roflol |
| 16:01.57 | drmessano | arossouw: Of course, you can shut the damn phones off too |
| 16:02.17 | ChannelZ | it's your phone sysem, you can prevent them from doing pretty much anything. Only thing you'd probably need exception for is emergency services |
| 16:02.20 | thenthenio | kaldemar: Thanks! And is FreePBX GUI able to setup sip.conf config file? |
| 16:02.36 | arossouw | i've implemented VMAuthenticate, but they can probably give away phone access codes aswell |
| 16:03.58 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:04.13 | arossouw | anyone Snom phones are losing their passwords, *sigh |
| 16:04.21 | arossouw | s/anyone/anoying |
| 16:04.55 | kaldemar | thenthenio: a subset of the settings, yes |
| 16:05.28 | arossouw | ChannelZ: i'm just the engineer, the senior ppl need to make that call, but i think they need to ammend the employment contract, and say phone abuse will not be tollerated |
| 16:05.50 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:05.58 | thenthenio | kaldemar: seems that fresh install od AsteriskNOW has port 5060 closed (nmap says) |
| 16:06.04 | ChannelZ | are they making long distance calls or something? |
| 16:06.41 | arossouw | not long distance, but cellular and in south africa , the telco industries like to rip consumers off |
| 16:06.42 | thenthenio | kaldemar: how to open 5060 port from FPBX GUI? |
| 16:06.51 | arossouw | the costs are higher, than what they should be |
| 16:07.26 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 16:07.38 | [TK]D-Fender | thenthenio: GUI's are not supported in this channel. Please visit theirs |
| 16:07.42 | [TK]D-Fender | ~freepbx |
| 16:07.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:08.22 | arossouw | ah, its fine, if it was my money i'd be bothered, but fortunately its not |
| 16:08.33 | thenthenio | infobot: OK! |
| 16:08.34 | infobot | fine |
| 16:10.10 | *** join/#asterisk kisom (n=x@c-19dde155.648-1-64736c11.cust.bredbandsbolaget.se) |
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| 16:10.30 | arossouw | has anyone tried connecting asterisk with skype? |
| 16:10.38 | arossouw | just curious how it works |
| 16:11.34 | drmessano | There is a Skype channel driver for Asterisk |
| 16:11.45 | ChannelZ | I've got it |
| 16:11.48 | *** part/#asterisk thenthenio (n=mauro@217.221.242.43) |
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| 16:11.57 | drmessano | I've got 2 licenses |
| 16:12.02 | drmessano | I'm better |
| 16:12.04 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 16:12.15 | ChannelZ | But I had an extra $66 for a hooker |
| 16:12.46 | drmessano | I have your moms home number? |
| 16:13.03 | ChannelZ | your loss :) |
| 16:13.08 | drmessano | lol |
| 16:14.06 | p3nguin | arossouw: exten => _NXXZXXXXXX,n,GotoIfTime(18:02-7:58,mon-fri,*,*?weeknights,${EXTEN},1) |
| 16:14.35 | p3nguin | arossouw: Of course you'll need more than that for a complete dialplan, but that gives you something to go off of. |
| 16:14.38 | *** join/#asterisk razu (n=razu@razu.data.ee) |
| 16:15.25 | p3nguin | arossouw: I also have another line for sat-sun. |
| 16:16.48 | *** join/#asterisk waa (n=waa@201.94.132.6) |
| 16:17.31 | arossouw | p3nguin: thank you very much, i'll figure the rest out |
| 16:19.36 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:19.36 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:21.42 | madduck | if i drill a 10000-20000/udp hole into the firewalls, RTP initiated with SIP/tcp suddenly works |
| 16:22.06 | p3nguin | amazing |
| 16:22.27 | lmadsen | which makes perfect sense |
| 16:22.29 | madduck | assuming the client chooses port 12345/udp for RTP, how does the firewall know that it should redirect RTP there? |
| 16:22.31 | lmadsen | modify rtp.conf |
| 16:22.33 | madduck | lmadsen: yes |
| 16:22.41 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 16:22.52 | lmadsen | make the range smaller |
| 16:23.07 | madduck | lmadsen: nah, i'd rather fix nf_conntrack_sip. ;) |
| 16:23.13 | madduck | assuming the client chooses port 12345/udp for RTP, how does the firewall know that it should redirect RTP there? |
| 16:23.31 | madduck | or does incoming audio only work once an outgoing RTP packet has crossed the firewall? |
| 16:23.35 | madduck | that would explain it. |
| 16:23.48 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:51c1:589a:68b1:93e1) |
| 16:23.49 | Faustov | lmadsen: doesn't the SIP module in iptables act like the one for FTP, that is, open the ports for a current SIP session itself, without having to "drill a hole"? |
| 16:24.07 | madduck | Faustov: it does not track tcp, only udp |
| 16:24.23 | Faustov | I see |
| 16:25.17 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 16:25.23 | madduck | Faustov: hence 25 17:23 < madduck> lmadsen: nah, i'd rather fix nf_conntrack_sip. ;) |
| 16:26.09 | Faustov | madduck: you'll definitely get a lot of cookies if you fix that :P |
| 16:26.26 | madduck | i have already started talking to the maintainer |
| 16:26.31 | madduck | problem is that i am really not a coder |
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| 17:24.38 | Kobaz | how would i set a standard channel variable on a channel, ie: not using SHARED |
| 17:25.00 | Kobaz | since SHARED has it's own namespace and has to be access explicitly |
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| 17:31.47 | Katty | scowls |
| 17:32.02 | Katty | one of our newly hired employees reported a problem to me today she wanted me to work on |
| 17:32.09 | Katty | rather than her manager. |
| 17:32.10 | [TK]D-Fender | Kobaz: ? |
| 17:32.15 | Katty | and her issue is that she cannot get onto facebook |
| 17:32.28 | Qwell | Fix it, stat! |
| 17:32.35 | Kobaz | [TK]D-Fender: in dialplan... clobber a channel variable on another channel |
| 17:32.58 | [TK]D-Fender | Kobaz: Yuo want to have 1 channel change vars on another? |
| 17:33.01 | Kobaz | yeap |
| 17:33.08 | Nivex | Katty: presumably that is by design |
| 17:33.12 | [TK]D-Fender | Kobaz: call a script tha uses AMI |
| 17:33.17 | Kobaz | SHARED is cool, but doesn't do what i need |
| 17:33.23 | Kobaz | [TK]D-Fender: hah, yeah... don't want to do that :P |
| 17:33.29 | Katty | Nivex: i don't know. i don't care. |
| 17:33.38 | Kobaz | i couldn't find anything built in, in dialplan to do that |
| 17:33.44 | Kobaz | so i'm writing an EXPORT function |
| 17:33.45 | Katty | Nivex: as i told her, i have better things to do that figure out why her facebooik isn't working at her place of employement |
| 17:34.20 | voipmonk | nice |
| 17:34.23 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 17:34.34 | [TK]D-Fender | Kobaz: and I don't want to have to do all sorts of things. Welcome to reality. |
| 17:34.39 | Kobaz | heh |
| 17:34.44 | Katty | i should have asked her what part of her job requires she has facebook acess |
| 17:34.56 | Kobaz | [TK]D-Fender: well, after i finish my function, i'll be able to write channel variables to other channels in dialplan |
| 17:35.42 | Kobaz | allrightey |
| 17:35.45 | Kobaz | er |
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| 18:10.12 | Deeewayne | ~silly |
| 18:10.13 | infobot | methinks silly is 'a good question, the answer to which is "yes, i am"' |
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| 18:17.41 | Qwell | Deeewayne: silly ulawian |
| 18:18.22 | Deeewayne | that's insulting! |
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| 19:14.17 | haryv | Anyone know of a free reverse lookup site? |
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| 19:19.50 | p3nguin | haryv: Whitepages.com, 411.com, probably several others that I don't know about |
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| 19:27.33 | hardwire | tweet.tweet.tweet. |
| 19:27.42 | hardwire | that should be this channels version of soandso: ping |
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| 19:53.46 | Naikrovek | haryv: whois.arin.net i believe |
| 19:54.07 | Naikrovek | ws.arin.net |
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| 20:05.54 | Kobaz | yay |
| 20:05.57 | Kobaz | my module works |
| 20:06.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 20:06.12 | Kobaz | [TK]D-Fender: now i can set channel variables on other channels in dialplan |
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| 20:21.16 | dlynes | Is there a reason why on t.38 passthrough, my end point would be getting reinvites from both the far end and from the passthrough box? |
| 20:21.51 | dlynes | i.e. it gets a ulaw reinvite from the far end, and then gets a t.38 reinvite from the passthrough box |
| 20:22.03 | dlynes | This is on asterisk 1.6.1.8 |
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| 20:35.12 | pjm | evening all, does anyone know how to make the x100p fxo card read caller ID from UK pstn, i'm using asterisk-1.6.1.10 and dahdi-linux-complete-2.2.0.2+2.2.0 |
| 20:38.06 | *** join/#asterisk garymc (n=garymc@host86-164-38-69.range86-164.btcentralplus.com) |
| 20:38.16 | garymc | Anyone know if i can make asterisk pass calls to my employees a call at a time. As they would be on commision from home, so it could be fair. People que to take calls. Not the first to answer it? Anyone under stand what i mean? |
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| 20:48.54 | Naikrovek | garymc: so if 10 calls come in, and you have 5 people at home, it'll pass 2 to each? |
| 20:49.07 | garymc | ok |
| 20:49.18 | garymc | so say i have five people and 2 calls come in |
| 20:49.19 | Naikrovek | i'm asking you if that's what you mean |
| 20:49.23 | Naikrovek | oh |
| 20:49.24 | garymc | it is |
| 20:49.25 | Naikrovek | go ahead |
| 20:49.42 | garymc | the first two extension get the calls? 201 and 202 |
| 20:49.46 | Naikrovek | you want to balance it out so no one person gets a lot more calls than someone else |
| 20:49.56 | garymc | then another call comes in 10 mins later so 203 gets that call? |
| 20:50.00 | garymc | yes |
| 20:50.27 | Naikrovek | well i'm sure it's possible but it would involve keeping track of that in astdb or something and writing logic for it in the dialplan |
| 20:55.25 | jblack | just use standard queues? |
| 20:55.34 | jblack | One person each, as calls end, new ones are routed |
| 20:59.34 | raden_work | can 1 phone have multiple VM's are still get notified properly ? |
| 21:00.50 | [TK]D-Fender | raden_work: Yes |
| 21:01.07 | raden_work | I will have to futz with it then |
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| 21:06.21 | p3nguin | Doesn't Queue() handle calls the way that guy was asking by default? strategy=rrmemory should do just fine. |
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| 21:07.13 | p3nguin | Of course he didn't bother to stick around, so additional discussion on the matter won't really be beneficial to him. |
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| 21:17.38 | [TK]D-Fender | p3nguin: Of course not. As soon as Naikrovek whote the words "dialplan" and " astdb" the crushing reality of "fucking god-damned FreePBX" rears its head :) |
| 21:17.51 | hardwire | heh |
| 21:17.56 | p3nguin | hmm |
| 21:18.01 | hardwire | pokes p3nguin |
| 21:18.09 | p3nguin | falls over |
| 21:18.18 | hardwire | can't afford a turkey this year.. but p3nguin sounds good. |
| 21:19.04 | [TK]D-Fender | checkout time. Later all |
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| 21:41.20 | joeflyde | Goodbye |
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| 22:14.43 | chackal_sjc | guys |
| 22:15.08 | chackal_sjc | does any one here recomends any book for IAX development? (softphone) |
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| 22:29.13 | Katty | :> |
| 22:29.48 | Katty | HELLO THAR |
| 22:30.25 | Wired_Life | know anyone wich hardware supports playback with noanswer |
| 22:30.50 | Katty | :< |
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| 22:32.37 | Katty | geeze where is everyone |
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| 22:39.00 | Corydon76-dig | Katty: already gone for the holiday |
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| 22:43.38 | *** join/#asterisk Ge0RgeR^ (n=georger@213-47-10.netrunf.cytanet.com.cy) |
| 22:43.48 | Ge0RgeR^ | hello |
| 22:44.40 | Ge0RgeR^ | i need some help |
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| 22:52.59 | chuckf | with? |
| 22:54.27 | drmessano | direct dial dysfunction? |
| 22:54.34 | drmessano | It affects 1 in 10 men |
| 22:55.37 | Wired_Life | can anyone help me i want early media function |
| 22:57.38 | Godfather_ | can somebody explain me that? i call a script with agi and the cli tell me that the file doesnt exist..? http://pastebin.com/m76ce6cc5 |
| 23:00.04 | dlynes | Godfather_: does it have execution privileges? does it have permission for the user that asterisk is running as? |
| 23:00.25 | Godfather_ | dlynes, see the ls -l |
| 23:00.27 | Godfather_ | i pasted it |
| 23:00.43 | dlynes | Godfather_: I have no idea what user your asterisk is running as |
| 23:01.00 | dlynes | Godfather_: is it running as the asterisk user? |
| 23:01.23 | Godfather_ | asterisk 1797 2.1 2.1 30972 10672 ? Ssl 18:23 0:48 /usr/sbin/asterisk -p -U asterisk |
| 23:01.24 | dlynes | Godfather_: also, you shouldn't be using 0777 permissions in practice, either |
| 23:01.43 | Godfather_ | dlynes, i know.. just trying.. |
| 23:01.49 | dlynes | Godfather_: what's the first line of your 'd.php' file? |
| 23:01.49 | Sargun | Does anyone here have recommendations on SIP < -- > SS7 gateways |
| 23:02.07 | Godfather_ | debian:/usr/share/asterisk/agi-bin# cat d.php |
| 23:02.07 | Godfather_ | #!/usr/bin/php -q |
| 23:03.21 | Godfather_ | dlynes, very strange, isnt? |
| 23:03.40 | dlynes | Godfather_: Pastebin the entire contents of the agi script |
| 23:03.46 | dlynes | Godfather_: not yet |
| 23:04.06 | dlynes | Godfather_: maybe if I don't see something I expect in your php file it is, though |
| 23:04.17 | dlynes | Godfather_: but I suspect I'll see the answer in your php file |
| 23:04.46 | Godfather_ | dlynes, http://pastebin.com/m14a9a807 |
| 23:04.57 | Godfather_ | (its my first AGI script, just c&p) |
| 23:06.36 | Godfather_ | the original call was exten => 03,n,AGI(d.php,numeros.txt) (i modified that to verifiy) |
| 23:07.06 | dlynes | Godfather_: and the original call worked? |
| 23:07.18 | Godfather_ | No |
| 23:07.37 | Godfather_ | Getting the same, i tried that because the example i follow uses | isntead of , |
| 23:08.15 | Godfather_ | And i was not 100% to use "," passing arguments to an agi script, but this is not the problem i think |
| 23:09.10 | dlynes | Godfather_: have you tried running outside of an agi environment, commenting out the agi-specific calls? |
| 23:10.20 | Godfather_ | dlynes, commeting the agi calls in d.php ? |
| 23:10.40 | Godfather_ | actually i dont understand your last line, what you are meaning? |
| 23:11.12 | Godfather_ | *what are you |
| 23:13.23 | dlynes | Godfather_: actually, nvm...just try running it outside of the agi environment, to see if you get an error |
| 23:13.53 | Godfather_ | dlynes, can you pass me another agi() script to verify that the error is in d.php? |
| 23:15.10 | Godfather_ | I restarted asterisk also, to verify that * actually known that there is an agi script inside ../agi-bin |
| 23:18.41 | Godfather_ | dlynes, any ideas? |
| 23:18.48 | dlynes | Godfather_: AGI(agi-test.agi) |
| 23:19.06 | dlynes | Godfather_: That's usually in your agi-bin directory if you did a source install |
| 23:19.07 | *** part/#asterisk Malkor (n=marco@hlle-d9ba1867.pool.mediaWays.net) |
| 23:19.09 | Godfather_ | dlynes, i dont have this under /agi-bin |
| 23:19.24 | Godfather_ | /usr/share/doc/asterisk-config/examples/agi-test.agi |
| 23:19.27 | Godfather_ | i got it here |
| 23:19.44 | Godfather_ | i mv it to /agi-bin and test it again |
| 23:20.03 | dlynes | Godfather_: ok |
| 23:20.42 | dlynes | Godfather_: also try running the following command at the command prompt: 'which php' |
| 23:20.50 | dlynes | Godfather_: what do you get for the result? |
| 23:21.53 | Godfather_ | dlynes, with agi-test.agi works! |
| 23:22.03 | dlynes | Godfather_: what do you get from 'which php'? |
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| 23:22.27 | Godfather_ | dlynes, nothing. |
| 23:22.44 | dlynes | Godfather_: you don't have php command line binaries installed, then |
| 23:22.56 | dlynes | Godfather_: and that's why your agi won't work |
| 23:23.20 | Godfather_ | dlynes, i've installed php5-common http://pastebin.com/m3b15a915 |
| 23:23.34 | dlynes | Godfather_: agi-test.agi worked because it uses perl, not php |
| 23:23.35 | Godfather_ | hum... then, what should i install? |
| 23:23.39 | Godfather_ | i see |
| 23:23.42 | dlynes | Godfather_: which distro are you using? |
| 23:23.45 | Godfather_ | debian |
| 23:23.59 | dlynes | Godfather_: ubuntu, or lenny/etch? |
| 23:24.05 | Godfather_ | lenny |
| 23:25.03 | dlynes | Godfather_: apt-get install php5 php5-common libapache2-mod-php5 |
| 23:25.18 | dlynes | Godfather_: the last one is only needed if you want to use php5 on apache on that machine |
| 23:25.28 | dlynes | Godfather_: oops...forgot one more |
| 23:25.34 | dlynes | Godfather_: apt-get install php5-cli |
| 23:26.17 | Godfather_ | dlynes, that last is the only pack i havent |
| 23:26.22 | Godfather_ | i try d.php now |
| 23:27.39 | Godfather_ | dlynes, yes, you were right, "php5-cli", now works with d.php :D |
| 23:27.59 | Godfather_ | ty |
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| 23:28.22 | dlynes | np |
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| 23:45.39 | p3nguin | dlynes: Good job. |
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