| 00:29.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:31.14 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 00:31.43 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 00:33.28 | grabes222 | Well whatever causes me to dump core in >=1.6.0.14 and 1.6.1.X does NOT happen in 1.6.2.0 rc8 :) |
| 00:34.33 | *** join/#asterisk psykon (n=joel@221-116.96-97.tampabay.res.rr.com) |
| 00:36.43 | TommyBotten | grabes222: So 1.6.2.0 works? |
| 00:36.58 | TommyBotten | TommyBotten: I'm using a AES(AEX) 800 card |
| 00:37.00 | TommyBotten | eh |
| 00:37.05 | TommyBotten | grabes222: I'm using a AES(AEX) 800 card |
| 00:37.46 | grabes222 | 1.6.2 is not crashing on faxing for me... I have sent 20+ and unable to reproduce |
| 00:37.56 | grabes222 | I am using a Sangoma A104d |
| 00:38.24 | TommyBotten | Cool. I'll test it on my end |
| 00:38.41 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 00:38.58 | TommyBotten | My issue is, wether or not this fixes it, I'm a bit afraid of pushing 1.6.2.0-rc4 into a prod environment |
| 00:40.15 | grabes222 | well its rc8, and I agree. I mainly use this machine to proxy calls from the PSTN to other servers, so its functionality is limited |
| 00:41.16 | grabes222 | TommyBotten : I am having MUCH better faxing results, so I am keeping my fingers crossed for now. Around 8:30EST this machine will hit around 60 channels so I will use that as a guidline |
| 00:42.49 | TommyBotten | Ah. Please tell me how it works out for you |
| 00:43.23 | TommyBotten | In my setup, this node does all of the traffic handling, and fax is only a small part of it |
| 00:43.35 | TommyBotten | And much better fax results - are there any other changes? |
| 00:45.09 | grabes222 | TommyBotten : I haven't done any config changes for 1.6.2 so its a drop in replacement as of now |
| 00:46.24 | TommyBotten | Ok |
| 00:54.54 | *** join/#asterisk p3nguin (i=BuhPjX1J@9xf6gbZ.a2infotech.com) |
| 00:56.51 | bcrisp | howdy |
| 01:02.32 | TommyBotten | Hi |
| 01:08.06 | bcrisp | when i dial into our IVR the first portion of the audio is cut off... should i be adding some silence before the IVR? |
| 01:08.24 | ChannelZ | yeah |
| 01:08.32 | bcrisp | whats good, like 2 seconds? |
| 01:08.34 | ChannelZ | you can put a little delay in the Answer() |
| 01:09.10 | bcrisp | im loving life after discovering the flexibility of local channels |
| 01:09.12 | ChannelZ | I do fine with a second on analog DAHDI channels |
| 01:09.28 | bcrisp | the calls are piped in through a sip prvoider |
| 01:09.38 | bcrisp | i think 1 - 2 seconds should do the trick |
| 01:10.12 | ChannelZ | try a second and go up from there if needed. |
| 01:10.16 | ChannelZ | I'm not sure why this stuff gets cut off |
| 01:10.29 | ChannelZ | I had to do 1.5sec on a Skype channel |
| 01:11.39 | *** part/#asterisk Davedan (n=me@DSL217-132-63-74.bb.netvision.net.il) |
| 01:12.25 | bcrisp | its working good at 1400ms |
| 01:12.37 | bcrisp | one thing is kinda strange.. it doesnt ring |
| 01:12.50 | bcrisp | i mean.. the caller doesn't hear any ringing while its in the process of connecting in |
| 01:14.28 | ChannelZ | well it delays after picking up before continuing |
| 01:14.46 | ChannelZ | so it should pick up immediately. If you want the caller to hear a ring, then do a Wait() before the Answer() instead I guess |
| 01:15.35 | p3nguin | bcrisp: Answer(), then Wait(2) or Playback(silence/2). |
| 01:15.59 | ChannelZ | Answer() and Wait(2) is the same as Answer(2000) |
| 01:17.35 | p3nguin | If you really wanted ringing, you could always play it. I don't see why you would need any ringing since your IVR will be beginning playback in just a second or two anyway. |
| 01:17.49 | dlynes | ChannelZ: typically, you want to do Wait(2), and then Answer() though, so that you get the caller id |
| 01:18.13 | ChannelZ | depends on your hardware and how lazy your phone company is :) |
| 01:18.46 | bcrisp | p3nguin thanks |
| 01:19.15 | dlynes | ChannelZ: well, if you're answering an analog call |
| 01:19.16 | file | did I really just watch a commercial with an LG phone that has a projector attachment? I think so. |
| 01:19.37 | bcrisp | file: soon you will see 3d projection in smart phones |
| 01:19.38 | dlynes | file: you've been drinking too much of that newfie screech |
| 01:20.04 | file | shakes his head |
| 01:20.20 | ChannelZ | Thankfully I haven't had any problem with my TDM card; I run an AGI almost right off the bat that looks up the caller ID in a database and I've never missed one that I've seen |
| 01:21.49 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:22.35 | ChannelZ | at home though my SPA3102 I have to delay a couple seconds or it doesn't get it |
| 01:25.12 | bcrisp | man i love perl |
| 01:25.51 | ChannelZ | oh no, don't start |
| 01:25.57 | TommyBotten | hehe |
| 01:27.01 | bcrisp | :/ |
| 01:27.20 | bcrisp | the fact that it has die as a keyword... gotta love it |
| 01:27.23 | *** join/#asterisk dkirker (n=dkirker@adsl-75-15-141-6.dsl.snlo01.sbcglobal.net) |
| 01:27.35 | TommyBotten | hehe |
| 01:27.59 | TommyBotten | Perl is great when you need something done. But I never spend any time at all in my spare time. |
| 01:32.43 | grabes222 | TommyBotten : FYI 1.6.2 rc8 seems pretty stable I have passed a little over 1500 calls through an no issues |
| 01:33.56 | *** join/#asterisk heliosj (n=jeff@i216-58-41-253.cybersurf.com) |
| 01:34.08 | bcrisp | TommyBotten: im just learning it now, but like its simplicity |
| 01:40.58 | TommyBotten | grabes222: Nice. thanks for the input |
| 01:41.12 | TommyBotten | bcrisp: Well, there is a backside to it |
| 01:41.45 | TommyBotten | When dealing with 'hints'. Is there a specific application/module that needs to be loaded? |
| 01:41.58 | bcrisp | ~roulette |
| 01:41.59 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 01:42.43 | TJNII | ~roulette |
| 01:42.44 | infobot | ACTION watches tjnii pull the trigger: Click! |
| 01:43.01 | TommyBotten | ~roulette |
| 01:43.02 | infobot | ACTION watches tommybotten pull the trigger: Click! |
| 01:43.06 | TommyBotten | *phew* |
| 01:44.21 | bcrisp | no bullets... |
| 01:44.27 | bcrisp | (recession) |
| 01:45.00 | TJNII | At least infobot isn't one of those kids who tries to play Russian Roulette with a semi-auto.... |
| 01:45.44 | TommyBotten | Well, a semi auto has usually a higher capacity... so it should be less of a chance, right? |
| 01:46.09 | bcrisp | you guys seen Intacto? |
| 01:46.13 | TommyBotten | Nope |
| 01:46.20 | bcrisp | that movie is awesome |
| 01:46.29 | bcrisp | its "like OMG" |
| 01:46.47 | TommyBotten | Does it have an idiot playing russian roulette with a semi automatic? |
| 01:55.10 | bcrisp | kind of |
| 01:55.19 | bcrisp | it has blindfolded people sprinting to a heavily wooded forest |
| 01:55.25 | bcrisp | s/to/through |
| 01:55.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 01:55.45 | bcrisp | s\to\through |
| 01:55.46 | bcrisp | hmm |
| 02:01.28 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 02:02.29 | p3nguin | bcrisp: s/to/through/ |
| 02:02.39 | p3nguin | You have to terminate the expression. |
| 02:03.43 | bcrisp | ah |
| 02:03.44 | bcrisp | ty |
| 02:06.17 | bcrisp | s/ty/thank you/ |
| 02:07.29 | bcrisp | time to work on the xmas tree, peace out |
| 02:22.40 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 02:26.54 | *** join/#asterisk TommyBotten (n=noosjent@217-14-12-26-dhcp-osl.bbse.no) |
| 02:30.32 | *** join/#asterisk chendy (n=chatzill@58.251.103.213) |
| 02:46.00 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 02:46.03 | Katty | hello. |
| 02:46.16 | ChannelZ | hola |
| 02:46.43 | TommyBotten | Hiya Katty |
| 02:50.34 | Katty | anything going on |
| 02:50.55 | TJNII | jblack was looking for you a couple hours ago. |
| 02:51.00 | Katty | hmm |
| 02:51.12 | TJNII | And getting quite annoyed with the ~seen suggestions. |
| 02:51.29 | Katty | 'seen suggestions'? |
| 02:51.41 | TJNII | The ~seen bot command |
| 02:51.45 | Katty | he was probably wondering how sammy is doing |
| 02:51.59 | Katty | sammy being a ferret, who tangled with Riddick last night |
| 02:52.12 | TJNII | Riddick? |
| 02:52.17 | Katty | a 90lb german shepherd |
| 02:52.22 | TJNII | Aah |
| 02:53.44 | TJNII | My folks have a german shepard about that size. The 15lbs house cat is afraid of him, whereas the 5lbs stray cat my mom took in doesn't take any crap. |
| 02:54.07 | Katty | nods |
| 02:54.24 | Katty | riddick was 7 weeks when i brought him home. the ferrets were already over a year old then. |
| 02:54.24 | TJNII | I assume Sammy survied? |
| 02:54.38 | Katty | they were raised together, climbed all over riddick, dig in his food bowl... |
| 02:54.48 | Katty | riddick has never shown any signs of aggression. not once. |
| 02:55.01 | Katty | sammy's front left leg is red... |
| 02:55.07 | Katty | probably broken blood vessels |
| 02:55.19 | Katty | but sammy's eating fine, playing fine, walking fine... |
| 02:55.27 | TJNII | That's good. |
| 02:55.33 | Katty | nods |
| 02:55.43 | Katty | i know sammy has occasionally tried to nip at riddick's paws |
| 02:56.02 | Katty | i gave riddick a triscuit last night...and i think sammy just maybe poked his buttons at the wrong time or decided he wanted it |
| 02:56.32 | Katty | not sure, to be honest...one minute all was well, the next growling and squealing...but the time i'd flown out of my chair sammy was out of riddick's mouth |
| 02:56.33 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 02:56.38 | jblack | katty: Ok |
| 02:58.52 | Katty | http://www.furniturestoreblog.com/image/2009/12/Star%20Wars%20Furniture%20Han%20Solo%20Office%20Desk.jpg |
| 02:59.27 | jblack | Thanks for the update. That's what I was aking for earlier. |
| 02:59.35 | *** join/#asterisk pirulo (n=andres@70.56.223.76) |
| 02:59.39 | Katty | (= |
| 02:59.43 | TJNII | Katty: Did you see the picture on reddit of the squirrel that got a hold of a paintball? |
| 02:59.48 | Katty | yes i did :> |
| 02:59.51 | Katty | it was /adorable/ |
| 03:00.35 | jblack | The IM client on my phone seems broken. It must not be properly marking me away. |
| 03:00.47 | Katty | it shows you were away earlier |
| 03:00.52 | Katty | on your gtalk account |
| 03:01.02 | jblack | Oh, so you ignored away. |
| 03:01.05 | jblack | Ok, then it works. |
| 03:01.19 | Katty | i always ignore it, for that reason |
| 03:02.45 | jblack | so.. what if I had been sleeping? |
| 03:02.58 | Katty | ^_- |
| 03:03.03 | Katty | who takes their im client to bed |
| 03:03.11 | jblack | I always keep my phone with me. |
| 03:03.19 | Katty | you're weird |
| 03:03.30 | Katty | does it make noise on IM? |
| 03:03.32 | Katty | or just ringing |
| 03:03.48 | jblack | It makes noise. It says "text message, text message". |
| 03:04.35 | Katty | why do you take it to bed with you anyway? |
| 03:04.59 | jblack | why do I keep my phone with me when I sleep? In case there's an emergency, someone can reach me. |
| 03:05.03 | jblack | Like last night. |
| 03:05.25 | Katty | what happened last night? |
| 03:05.59 | Katty | or were you talking about sammy? i know you weren't asleep then :P |
| 03:06.21 | jblack | I'm talking about sammy. That's an example of why I keep my phone with me everywhere I go |
| 03:07.02 | Katty | ahh i see |
| 03:08.09 | jblack | although things have gotten bad enough with some people that I had to set my phone up to silence itself at 11:30. |
| 03:08.15 | Katty | hopefully that wasn't me. |
| 03:08.15 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 03:08.23 | Katty | usually i'm zonked out and dead to the world by then |
| 03:08.25 | Katty | hi crispy |
| 03:08.26 | bcrisp | i just can't stay away from here |
| 03:08.30 | bcrisp | hiya |
| 03:08.34 | Katty | it's okay. |
| 03:08.38 | jblack | Hi bcrisp. :) |
| 03:08.39 | TJNII | We understand. |
| 03:08.46 | bcrisp | *anon? |
| 03:09.13 | Katty | is it legal to have a staircase in a house with no railing on one side? |
| 03:09.37 | jblack | that would be housing code, and that varies from state to state |
| 03:09.38 | bcrisp | hmm i would think not |
| 03:09.50 | TJNII | The house I grew up in had one like that. |
| 03:09.52 | bcrisp | although ive seen em |
| 03:10.03 | Katty | http://208.106.250.72/_media/imgs/articles2/a96905_5-Dangerous.jpg |
| 03:10.34 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
| 03:10.39 | TJNII | That's a bit much. |
| 03:10.40 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 03:10.44 | Katty | that seems like a bad call. |
| 03:10.56 | bcrisp | yes.. and those stairs look slippery! |
| 03:11.04 | bcrisp | imagine walking down those in a pair of socks |
| 03:11.06 | jblack | That's freaky. :) |
| 03:11.20 | Katty | the only way i'd think it was okay is if it was for a cat |
| 03:11.28 | Katty | but it's obvious that's not for a cat |
| 03:11.44 | jblack | That must be stainless steel |
| 03:11.44 | bcrisp | that would actually be cute for a cat.. have mini stairs like that |
| 03:11.49 | bcrisp | maybe 12" wide |
| 03:11.53 | TJNII | I'd think there has to be something on the end. Those steps look thin enough that even stell would flex too much at the ends. |
| 03:12.09 | TJNII | s/stell/steel |
| 03:12.19 | jblack | Heh. That server is running IIS |
| 03:12.21 | bcrisp | actually ... they should make kitty slides for cats |
| 03:12.27 | Katty | bcrisp: i know several people with various shelves mounted on the wall for their cats |
| 03:12.50 | bcrisp | heh |
| 03:13.20 | Katty | i know one person who got a cube shoe thing, and carpeted it for their cat |
| 03:13.33 | *** join/#asterisk joobie (n=joobz@mx01.anric.com.au) |
| 03:13.39 | jblack | what's a cube shoe thing? |
| 03:13.55 | Katty | http://www.shopgetorganized.com/images/products/P16591B.jpg |
| 03:14.14 | jblack | Oh, huh |
| 03:14.57 | jblack | I wonder if the cat rotates cubbys. |
| 03:15.23 | Katty | http://www.furniturestoreblog.com/images/wall%20mounted%20cat%20furniture%20by%20cat%20interiors-thumb-350x444.jpg <- cute |
| 03:16.04 | jblack | This is more cute: http://img.photobucket.com/albums/0903/Natalka/FlickCATS/CatPot2.jpg |
| 03:17.09 | bcrisp | http://2.bp.blogspot.com/_qcrhDVVzhs4/SduccU43wqI/AAAAAAAABgk/WsTSc50pcjA/s1600-h/suicidemodel.jpg |
| 03:17.14 | Katty | there's an animal sactuary about 1.5hrs north of here that is very elaborate with their cat housing |
| 03:17.54 | bcrisp | jblack: cat stew? yummm |
| 03:18.21 | jblack | is now in the mood for chinese food |
| 03:19.23 | ChannelZ | had tacos |
| 03:19.46 | jblack | There's place not far from here.. pancheros I think it's called. _INCREDIBLE_ mexican food |
| 03:19.47 | Katty | i had breakfast |
| 03:19.58 | Katty | jblack: oh i do like me some good mexican food. |
| 03:20.05 | bcrisp | jblack where are u? |
| 03:20.05 | Katty | jblack: perhaps it's my favorite thing in the whole wide world |
| 03:20.07 | jblack | Like "I wish I was mexican" good. |
| 03:20.26 | jblack | bcrisp: The United States. |
| 03:20.41 | bcrisp | heh |
| 03:20.45 | Katty | we have a place out here which has vegetarian nachos |
| 03:20.59 | bcrisp | mexican food all over here |
| 03:21.00 | ChannelZ | not vegan I hope |
| 03:21.08 | Katty | lots and lots of veggies, with good variety, served in a white cheese sauce with chips |
| 03:21.20 | Katty | ChannelZ: no, but it would be considered vegetarian. |
| 03:21.21 | jblack | aren't normal nachos vegetarian |
| 03:21.27 | Katty | no |
| 03:21.34 | ChannelZ | depends on what you put on them eh? |
| 03:21.43 | Katty | the typical nachos here have beef, chicken, steak, or shrip |
| 03:21.48 | jblack | I think of nachos as corn chips and cheese. |
| 03:21.58 | ChannelZ | That's 7-11 nachos |
| 03:21.58 | Katty | far from it |
| 03:22.04 | Katty | yeah like theator nachos |
| 03:22.22 | ChannelZ | I typically do refried beans and jalapenos |
| 03:22.25 | ChannelZ | sour cream if I have it |
| 03:22.29 | jblack | that's where I see nachos. 7-11 and the movie theater. |
| 03:22.47 | psykon | Katty, I thought you liked Ruby Tuesdays the best |
| 03:22.52 | Katty | when i make my own, i use blue corn chips, beef or bison, cheese, tomato, onion, salsa, sour cream, guacamole... |
| 03:22.54 | ChannelZ | well get some nachos at your 'I wish I was mexican' restaraunt |
| 03:23.39 | jblack | They make their tortillas on the fly. |
| 03:23.59 | Katty | great. now i'm hungry. :< |
| 03:24.10 | bcrisp | there is a place i go nearby that serves jalapeno cream cheese covered chimichangas yum |
| 03:24.13 | jblack | they have this big.. squasher thing, where they put in dough, and they steam each tortilla as they need it. And the veggies there... It's all just so... I mean, wow |
| 03:24.19 | Katty | the parents made breakfast for dinner. their chickens have been laying a lot of eggs lately. 6 or 7 a day |
| 03:24.38 | ChannelZ | tortilla press |
| 03:24.38 | Katty | so that's why they had breakfast for dinner when i was visiting |
| 03:24.58 | bcrisp | ok stop it you're all making me starving |
| 03:25.04 | jblack | I was definitely tortilla impressed. |
| 03:25.20 | Katty | biscuits, gravy, sausage, fried potatoes, and eggs. |
| 03:25.33 | bcrisp | i want some gyudon |
| 03:25.39 | Katty | what's gyudon? |
| 03:25.43 | bcrisp | or some okonomiyaki |
| 03:25.48 | Katty | sounds orieental |
| 03:25.53 | bcrisp | yeah japanese |
| 03:25.59 | Katty | seafood? |
| 03:26.15 | bcrisp | gyudon - ultra thinly sliced beef over rice in a shoyuu sauce |
| 03:26.24 | Katty | shoyuu? |
| 03:26.37 | ChannelZ | lemme shoyuu sumthin! |
| 03:26.38 | Katty | fish sauce? |
| 03:26.38 | bcrisp | well pseudo shoyuu .. |
| 03:26.57 | jblack | tries to remember if he's left the house since friday |
| 03:27.23 | bcrisp | kind of katty.. but not required |
| 03:27.24 | Katty | bcrisp: images.google.com shows it is similiar to soy sauce |
| 03:27.36 | bcrisp | dashi broth / mirin / sugar / soy sauce |
| 03:27.41 | Katty | prefers the sweeter tamari to soy sauce |
| 03:27.45 | bcrisp | its a common base for japanese dishes |
| 03:27.50 | Katty | nods |
| 03:27.56 | bcrisp | very tasty and easy to make |
| 03:28.07 | Katty | a little goes a long way with me |
| 03:28.11 | Katty | but ryan pours it on |
| 03:28.18 | ChannelZ | uhhh |
| 03:28.20 | ChannelZ | stupid spammers |
| 03:28.24 | Katty | he takes sushimi and drowns it in soy sauce |
| 03:28.24 | ChannelZ | <PROTECTED> |
| 03:28.35 | bcrisp | i just use the broth to simmer the beef.. i normally dont pour it in like soup |
| 03:28.42 | Katty | nods |
| 03:28.47 | Katty | ChannelZ: that doesn't look good. |
| 03:29.03 | Katty | bcrisp: ryan also takes ramen, with shrimp, and soy sauce |
| 03:29.12 | bcrisp | hehe |
| 03:29.18 | bcrisp | ramen can actually be great if done right |
| 03:29.23 | Katty | he also likes ramen with chili and cheese and hot dogs too |
| 03:29.27 | Katty | so there's no accounting for him |
| 03:29.36 | bcrisp | in japan there are a lot of ramen shops |
| 03:29.44 | Katty | yeah but ramen there isn't waht ramen is /here/ |
| 03:29.51 | bcrisp | yeah |
| 03:29.57 | bcrisp | but at 10c a pack.. i mean:) |
| 03:30.07 | Katty | you know what i love? |
| 03:30.08 | Katty | rice noodles |
| 03:30.12 | Katty | omg, they are awesome. |
| 03:30.20 | bcrisp | i wanna make some tempura |
| 03:30.21 | Katty | my coworkers stare when i eat it. they have no idea what it is. |
| 03:30.36 | Katty | hard to come by tho... |
| 03:30.39 | bcrisp | they probably think its octopus tentacles or something |
| 03:30.46 | Katty | i can only seem to get it at an import store up in st. louis |
| 03:31.22 | bcrisp | japanese sweet curry and rice.. ok its settled |
| 03:31.26 | bcrisp | off to cook |
| 03:31.27 | *** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 03:31.30 | Katty | bye (= |
| 03:31.34 | bcrisp | cya |
| 03:33.35 | p3nguin | World Market? |
| 03:34.51 | Katty | it's next to walgreens |
| 03:34.58 | Katty | if you're familiar with kirkwood |
| 03:35.52 | *** join/#asterisk m0e (n=eX@41.196.209.68) |
| 03:37.18 | Katty | it's global foods market |
| 03:38.38 | *** join/#asterisk mog (n=mog@c-71-228-185-24.hsd1.al.comcast.net) |
| 03:38.38 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:38.51 | Katty | http://tinyurl.com/yc75qwv |
| 03:41.19 | p3nguin | Hadn't heard of that one before. |
| 03:41.55 | Xetrov` | i dont see a world market |
| 03:42.58 | p3nguin | http://www.worldmarket.com/ I guess |
| 03:43.25 | Katty | http://www.globalfoodsmarket.com/ |
| 03:43.44 | Katty | they have lots of photos |
| 03:44.42 | Katty | they have special clerks for people who have trouble with english (= |
| 03:44.57 | Xetrov` | sounds like the place my friend took me to in atlanta |
| 03:45.30 | Katty | best of all, they have an entire aisle dedicated to chocolate :> |
| 03:45.42 | Katty | lebanese is my favorite. |
| 03:46.51 | p3nguin | Nice site for a food store. |
| 03:46.58 | Katty | it is,isn't it? |
| 03:47.02 | p3nguin | quite |
| 03:47.08 | Katty | they get a ton of business |
| 03:47.17 | Katty | anywho it's bedtime |
| 03:47.23 | Katty | cya in the morning (= |
| 04:14.08 | *** join/#asterisk bcrisp (i=bcrisp@ip72-222-167-229.ph.ph.cox.net) |
| 04:28.25 | bcrisp | im using mixmonitor for recording queue calls.. is there a way to customize the filenames it writes to? |
| 04:30.23 | bcrisp | id like to use the interfacename (minus invalid characters) of the queue member as part of the name |
| 04:31.01 | bcrisp | hold your answers.. ill read some more |
| 04:32.08 | jblack | I had a piece of tooth break off. that's scary |
| 04:33.28 | bcrisp | ouch |
| 04:33.55 | bcrisp | i stepped on a sewing needle and it went all the way through my foot |
| 04:34.16 | bcrisp | had to use pliars to get it out |
| 04:36.00 | b14ck | Hey all, I'm experimenting with this asterisk build here. I've hit a point where I apparently Asterisk won't actually load extensions.conf at all. Is there a module that needs to be enabled in order for extensions.conf to load? I can't find documentation for it anywhere :x |
| 04:36.43 | bcrisp | b14ck what happens when you |
| 04:36.50 | bcrisp | are in the CLI and type dialplan reload |
| 04:37.04 | b14ck | no such command |
| 04:37.12 | b14ck | i can do dialplan set, show, or debug |
| 04:37.20 | bcrisp | what version? |
| 04:37.25 | b14ck | this is 1.6.1.11 |
| 04:37.34 | bcrisp | um |
| 04:37.39 | b14ck | I've got modules.conf set to not do any autoloading, and I've whitelisted only certain modules. |
| 04:37.48 | b14ck | I think that may be the issue, but not sure which one is missing |
| 04:37.49 | b14ck | =p |
| 04:37.49 | bcrisp | hm |
| 04:37.56 | bcrisp | one sec |
| 04:38.43 | b14ck | http://pastie.org/private/yxeg4mit9zzsqyw47glaw here's my modules.conf if you're interested |
| 04:38.55 | b14ck | sorry for the formatting, this pastebin doesnt handle my tabs well |
| 04:41.38 | bcrisp | ya im kinda new.. not sure what loads the dialplan commands |
| 04:42.17 | bcrisp | it thought it was core |
| 04:43.30 | bcrisp | b14ck - if you turn back on module auto load |
| 04:43.37 | bcrisp | does dialplan reload become available? |
| 04:44.01 | b14ck | no, actually. i didnt even test that before :x |
| 04:44.05 | b14ck | crap |
| 04:44.19 | bcrisp | did you build it from src? |
| 04:44.24 | b14ck | yep |
| 04:44.26 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:44.47 | bcrisp | do a dialplan show |
| 04:45.33 | bcrisp | ah.. |
| 04:45.41 | bcrisp | do you have pbx_config set to load in modules.conf? |
| 04:46.08 | b14ck | nope |
| 04:46.25 | bcrisp | load it |
| 04:46.27 | bcrisp | and restart |
| 04:46.44 | b14ck | hrm, i dont even have that modujle |
| 04:46.50 | b14ck | i guess when i tried to slim down the install |
| 04:46.51 | b14ck | i removed it |
| 04:46.52 | b14ck | heh |
| 04:46.55 | b14ck | that would explain =p |
| 04:46.58 | bcrisp | lol |
| 04:47.40 | bcrisp | pbx_config.so => (Text Extension Configuration |
| 04:47.53 | b14ck | heh |
| 04:47.54 | b14ck | yep |
| 04:47.56 | b14ck | i removed it |
| 04:47.57 | b14ck | lol |
| 04:47.59 | bcrisp | rofl |
| 04:48.02 | bcrisp | u can use ael? |
| 04:48.28 | bcrisp | b14ck it might be easier to allow auto load of modules |
| 04:48.32 | bcrisp | then use noload lines |
| 04:48.49 | b14ck | i never use ael stuff for this box |
| 04:48.50 | x86 | i want to call a number and play a wav file, is this possible? |
| 04:48.53 | b14ck | this is just for playing |
| 04:48.56 | bcrisp | x86 yes |
| 04:48.58 | b14ck | x86, ya, Playback() |
| 04:49.30 | x86 | exten => s,1,Dial(DAHDI/5) ; exten => s,2,Playback(foo) ? |
| 04:49.48 | bcrisp | ya if thats what u want |
| 04:49.54 | x86 | that won't try the playback after the call is hungup (and Dial returns control back to the dialplan)? |
| 04:50.03 | bcrisp | u can do this |
| 04:50.30 | bcrisp | exten => s,n,Goto(s-${DIALSTATUS},1) |
| 04:50.51 | x86 | eh? |
| 04:50.53 | bcrisp | then have exten => s-NOANSWER,1, .... |
| 04:50.58 | bcrisp | after you dial |
| 04:51.05 | [TK]D-Fender | x86: What do you want to be the trigger for the call-out? |
| 04:51.07 | bcrisp | you can check the dial status (i.e. they arent there) |
| 04:51.13 | x86 | [TK]D-Fender: the answer of the call |
| 04:51.15 | *** join/#asterisk ticoit (n=ticoit@201.191.190.126) |
| 04:51.25 | [TK]D-Fender | x86: No, you said * calling OUT. |
| 04:51.27 | x86 | [TK]D-Fender: as soon as the channel answers and the call is bridged, want to play the file |
| 04:51.39 | bcrisp | ah |
| 04:51.52 | [TK]D-Fender | x86: Why * is calling out? What triggers this? |
| 04:51.53 | x86 | [TK]D-Fender: actually no, i didn't specify ;) |
| 04:52.02 | x86 | [TK]D-Fender: inbound call from PSTN |
| 04:52.22 | x86 | [TK]D-Fender: or actually, inbound call from local extension too... either way |
| 04:52.35 | [TK]D-Fender | x86: Ok, call comes in. What makes it call out again? Describe the complete desired call flow |
| 04:52.58 | [TK]D-Fender | x86: Because the broken bits your dropping could mean anything so far |
| 04:53.17 | bcrisp | kinda like chinese food drive thru |
| 04:53.23 | ChannelZ | I think he's writing an automated phone prank system |
| 04:53.24 | x86 | [TK]D-Fender: from dialplan, want to issue a Dial, when the Dial answers, want to play wav |
| 04:53.42 | bcrisp | like a rejection hotline |
| 04:53.49 | x86 | ChannelZ: no i want to make a fax playback system |
| 04:54.04 | [TK]D-Fender | x86: And then? |
| 04:54.25 | bcrisp | remembering parts of "dude, where's my car" |
| 04:54.26 | [TK]D-Fender | x86: And You can't just play a wav to relay a fax |
| 04:54.27 | x86 | [TK]D-Fender: that's it... hang up |
| 04:54.28 | ChannelZ | wtf is a fax playback system? |
| 04:54.38 | x86 | [TK]D-Fender: why not? |
| 04:54.41 | bcrisp | ChannelZ, it plays back the modem noise |
| 04:54.50 | p3nguin | bcrisp: MixMonitor(whatever-you-feel-like-calling-the-file.gsm) |
| 04:54.50 | ChannelZ | Uh, that will totally not work |
| 04:54.51 | x86 | [TK]D-Fender: inbound faxes are recorded with MixMonitor |
| 04:54.51 | [TK]D-Fender | x86: because the 2 sides have to negotiate. |
| 04:55.15 | x86 | [TK]D-Fender: hmm ok |
| 04:55.15 | bcrisp | p3nguin, recording queue calls.. i was just using the mixmonitor option |
| 04:55.16 | p3nguin | bcrisp: I use MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.gsm,aW(1)) |
| 04:56.02 | bcrisp | p3nguin, nice |
| 04:56.58 | p3nguin | For the queue only, take a look at Set(MONITOR_FILENAME=foo) in queues.conf to see how to use it. |
| 04:57.34 | x86 | [TK]D-Fender: so what i want to do is impossible? |
| 04:57.38 | Pan3D | lol |
| 04:58.42 | bcrisp | p3nguin, thankya |
| 04:58.53 | bcrisp | x86 nothing is impossible |
| 05:01.25 | [TK]D-Fender | x86: FFS receive the damn fax and RESEND IT. You know... with FAX SOFTWARE. The kind that has existed for far longer than a decade for desktop PC use. |
| 05:01.42 | mchou | bah |
| 05:01.51 | [TK]D-Fender | pummels x86 with a boxed copy of WinFax Pro |
| 05:01.55 | bcrisp | [TK]D-Fender, whats a fax? |
| 05:01.56 | mchou | use a real fax machine |
| 05:02.22 | ChannelZ | faxes are dead |
| 05:02.26 | mchou | faxing on pc is more trouble than it's worth |
| 05:02.28 | [TK]D-Fender | x86: Or how about app_fax? |
| 05:02.34 | [TK]D-Fender | x86: Or Hylafax? |
| 05:02.42 | bcrisp | UberFax |
| 05:06.53 | joobie | buro |
| 05:06.56 | joobie | burp |
| 05:08.28 | bcrisp | blurb |
| 05:10.45 | mchou | [TK]D-Fender: help |
| 05:11.25 | joobie | anyone got some ideas on how to track call lengths for users entering a queue? currently as people dial into the queue, an intro plays and then they are dropped into a queue.. by the time the queuemember answers the call, the CDR log has already got a good 1 minute of time build up in the call length field..... |
| 05:11.47 | joobie | trying to track seperately the amount of time the user is sitting in the queue and seperate to that the itme the call takes form when the agent picks up |
| 05:12.32 | bcrisp | hold time |
| 05:12.46 | bcrisp | hm |
| 05:13.22 | [TK]D-Fender | mchou: Sorry... not touching that one... |
| 05:14.16 | [TK]D-Fender | joobie: Ever heard of this thing call the QUEUE LOG? It like... TELLS you stuff about how long they waited, talked, and like STUFF! |
| 05:14.24 | [TK]D-Fender | \o/ |
| 05:14.31 | joobie | wow |
| 05:14.36 | joobie | it sounds.. so.. perfect |
| 05:14.47 | joobie | thanks TK |
| 05:14.50 | joobie | you smartass |
| 05:14.55 | joobie | ;P |
| 05:15.50 | bcrisp | [TK]D-Fender, like omg |
| 05:16.45 | [TK]D-Fender | An then it was like.... |
| 05:16.48 | [TK]D-Fender | ~whee ! |
| 05:16.53 | bcrisp | ~roulette |
| 05:16.54 | infobot | ACTION watches bcrisp pull the trigger: BANG! |
| 05:16.55 | [TK]D-Fender | ~whee |
| 05:16.56 | infobot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 05:17.06 | Pan3D | hehe |
| 05:17.15 | bcrisp | ~blackhole |
| 05:17.16 | infobot | A very black hole, indeed, where light and matter cannot escape from. |
| 05:18.11 | bcrisp | does infobot know about * settings and such? |
| 05:18.18 | bcrisp | ~queues.conf |
| 05:18.42 | bcrisp | ~darth vader |
| 05:18.43 | infobot | You cannot imagine the power of the dark side. |
| 05:18.44 | [TK]D-Fender | bcrisp: * comes with docs on this stuff already, including the LOG FORMATS. |
| 05:18.51 | bcrisp | i know |
| 05:18.55 | bcrisp | was just curious |
| 05:19.07 | bcrisp | it'd be kinda cool to have a tip and trick macro in infobot |
| 05:19.15 | [TK]D-Fender | bcrisp: We need to batter-fry it and then put sprinkles to attract people to it :) |
| 05:19.25 | bcrisp | [TK]D-Fender, hells yes |
| 05:19.47 | bcrisp | instead of *, lets just refer to it as ( * Y * ) |
| 05:19.59 | bcrisp | that otta attract some nerds |
| 05:20.33 | bcrisp | infobot: punish bcrisp |
| 05:20.33 | infobot | ACTION forces bcrisp to install and use Windows |
| 05:22.50 | mchou | ~fried chicken |
| 05:22.57 | mchou | bah |
| 05:23.11 | mchou | what good is infobot |
| 05:23.29 | mchou | fried chicken bcrisp |
| 05:23.41 | bcrisp | ~donuts |
| 05:23.48 | mchou | pardon the pun |
| 05:24.04 | bcrisp | are you chinese? |
| 05:24.17 | mchou | lol |
| 05:24.26 | mchou | I'm french |
| 05:24.34 | bcrisp | same dif |
| 05:24.35 | ChannelZ | that explains a lot |
| 05:26.00 | bcrisp | channelz do you use the followme app? |
| 05:26.22 | ChannelZ | no |
| 05:26.28 | bcrisp | mchou, actually i was born in kentucky if that helps your joke |
| 05:26.53 | bcrisp | channelz, i was thinking it may have simplified what i was doing with the queues thing earlier |
| 05:26.53 | mchou | bcrisp: I'm partial to church |
| 05:27.10 | bcrisp | mchou, that does not compute |
| 05:27.14 | mchou | bcrisp: KFC is not so great |
| 05:27.34 | mchou | bcrisp: which part doent compute? |
| 05:27.36 | bcrisp | mchou, ya Paris Fried Chicken is the greatest! :) |
| 05:28.03 | mchou | bcrisp: seriously, Church fired chicken way better |
| 05:28.04 | bcrisp | mchou, church's ok i get it |
| 05:28.12 | bcrisp | Church's |
| 05:28.39 | mchou | popeye's isnt bad either |
| 05:28.48 | bcrisp | i dont like chicken |
| 05:28.52 | mchou | lol |
| 05:44.25 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) |
| 05:45.16 | *** join/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 05:45.58 | *** part/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 05:46.29 | bcrisp | b14ck, just couldnt stay away could ya |
| 05:47.32 | jblack | omg. |
| 05:47.49 | jblack | Godaddy has an advertisement that says "The internet is for everybody! Even black people!" |
| 05:48.19 | ChannelZ | Godaddy is the worst |
| 05:49.16 | jblack | "And we need a gay person and a jew too!" |
| 05:49.18 | p3nguin | bcrisp: Did you have a question about FollowMe()? |
| 05:50.06 | bcrisp | p3nguin: not a question, i was reading up on it... earlier i was trying to define behavior that would try to ring a particular person (queue member) on dif devices - i.e. softphone first, then cell etc |
| 05:50.28 | bcrisp | i ended up using member= > local channel instance |
| 05:50.41 | bcrisp | but i suppose i could have done the same thing by using followme |
| 05:51.22 | bcrisp | if that makes sense |
| 06:00.16 | p3nguin | I still haven't figured out what you didn't configure agents with callback, so that any agent logs in from the phone where he should take calls. |
| 06:00.30 | bcrisp | i very well might do that at some point |
| 06:00.51 | bcrisp | part of the issue is that they dont have the choice |
| 06:01.03 | p3nguin | Choice of what? |
| 06:01.09 | bcrisp | logging in |
| 06:01.38 | p3nguin | You're not making sense... you're writing the dialplan, so you can write it to do logins if you want. |
| 06:01.51 | ChannelZ | p3nguin: his agents are too stupid/lazy to do this |
| 06:01.59 | p3nguin | I see. |
| 06:02.01 | m0e | quick question, is there something like a phonebook module or something for asterisk? |
| 06:02.06 | bcrisp | they are walking around with cell phones |
| 06:02.25 | bcrisp | they aren't gonna dial anything to say 'hey im on a cell phone, queue me from here :)" |
| 06:02.31 | p3nguin | Do they ever answer calls on a phone other than those cell phones? |
| 06:02.40 | bcrisp | soft phone |
| 06:02.59 | bcrisp | they are required to answer their work cells |
| 06:03.12 | p3nguin | If they have a cell phone, why do they need a soft phone? |
| 06:03.26 | bcrisp | well in the office cell phone is a pain |
| 06:03.37 | bcrisp | hard to guarantee signal yatta yatta |
| 06:04.32 | p3nguin | Okay, then what circumstance would they be taking queue calls on the mobile phones? |
| 06:05.07 | bcrisp | when they arent in the office |
| 06:05.29 | p3nguin | How will your queue know if they are or are not in the office? |
| 06:05.44 | bcrisp | if they dont answer their softphone when available, we assume they are out of the office |
| 06:05.47 | bcrisp | and call their cell |
| 06:05.59 | bcrisp | (just hope they dont answer while in the restroom) |
| 06:06.40 | bcrisp | its a ringall scenario so.. the odds are is that someone will be available most of the time in the office |
| 06:07.05 | p3nguin | So they will just get up and walk away from their desk (soft) phone without marking their phones as dnd or anything? |
| 06:07.13 | bcrisp | yes |
| 06:07.19 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 06:07.29 | bcrisp | they are extremely busy |
| 06:07.51 | bcrisp | most are consultants |
| 06:07.54 | bcrisp | in and out all the time |
| 06:08.02 | p3nguin | Queue member penalty should only work when the lower priority is busy, not unavailable, so that won't work. I guess you probably already figured that out, though. |
| 06:08.16 | bcrisp | right |
| 06:08.17 | bcrisp | :) |
| 06:08.33 | bcrisp | instead i used member => Local/bla that a dialplan logic to check busy etc |
| 06:09.00 | p3nguin | So you've already completed this task? |
| 06:09.01 | bcrisp | aplauds himself for remarkable grammar |
| 06:09.12 | bcrisp | s/aplauds/applauds |
| 06:09.14 | bcrisp | s/aplauds/applauds/ |
| 06:09.21 | bcrisp | rofl |
| 06:09.22 | p3nguin | lol |
| 06:09.47 | bcrisp | ~roulette |
| 06:09.48 | infobot | ACTION watches bcrisp pull the trigger: Click! |
| 06:09.54 | p3nguin | s^lol^LOL^ |
| 06:09.58 | bcrisp | ya p3nguin, got it kind of working |
| 06:10.09 | p3nguin | I guess it doesn't know sed as well as I thought. |
| 06:10.49 | bcrisp | had a strange issue when dialing out of the softphone into * (via our sip provider) .. got channel unavails |
| 06:11.13 | bcrisp | ignore that |
| 06:11.25 | p3nguin | You dial from a soft phone to asterisk, via an internet-based provider... |
| 06:11.35 | bcrisp | nah im tired |
| 06:11.53 | bcrisp | used softphone to dial our did, routed into * via didww |
| 06:12.21 | bcrisp | hmmmm lol |
| 06:12.32 | bcrisp | ok, let me try again |
| 06:13.00 | p3nguin | waits patiently |
| 06:13.01 | bcrisp | softphone registered as a sip device in the employees context. employees context has pattern matching to dial out via callcentric |
| 06:13.15 | bcrisp | the # called was our didww did |
| 06:13.22 | bcrisp | which enters our incoming context |
| 06:13.44 | bcrisp | from there it was queued and first tried ringing the queue member SIP/user |
| 06:13.54 | bcrisp | that timed out and when it went to dial the cell it said channel unavail |
| 06:14.33 | bcrisp | loopty loop |
| 06:14.36 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 06:15.08 | bcrisp | ~tada |
| 06:18.29 | bcrisp | p3nguin, did you have to take some pepto after reading that? |
| 06:19.23 | p3nguin | I think I understand what you're saying, but I don't know how to put into words what I'm thinking. |
| 06:20.02 | bcrisp | lol |
| 06:20.12 | bcrisp | welcome to the club |
| 06:20.49 | bcrisp | i believe it was DIALSTATUS that was something similar to CHANNELUNAVAIL |
| 06:21.03 | bcrisp | i didn't have a _s.,1,blabla handler |
| 06:21.15 | bcrisp | so it was showing up as a warning or notice in CLI |
| 06:21.31 | [TK]D-Fender | .... |
| 06:21.47 | [TK]D-Fender | bcrisp>i didn't have a _s.,1,blabla handler <--- huh? |
| 06:21.51 | bcrisp | haha |
| 06:22.10 | bcrisp | forget it |
| 06:22.24 | bcrisp | im just saying that i was looking at DIALSTATUS in a macro |
| 06:22.32 | bcrisp | and had handlers for BUSY and NOANSWER |
| 06:23.02 | bcrisp | apparently the dialstatus was something else, and had no priority set up to handle it |
| 06:23.25 | bcrisp | CHANUNAVAIL ... when i plan to look into it again i would of course pastebin everything |
| 06:23.28 | *** join/#asterisk froud (n=sean@dsl-243-229-79.telkomadsl.co.za) |
| 06:24.51 | froud | I am looking for voice prompts. I was told of something that sounds like "celepstal" < spelling??? but I obviously have it wrong. Can somebody give me the correct name or URL. Thanks |
| 06:26.43 | [TK]D-Fender | froud: cepstral |
| 06:27.14 | [TK]D-Fender | froud: And that is a TTS engine, not a premade set of real voice prompts |
| 06:27.53 | p3nguin | bcrisp: Okay, I have an idea how to do what you were wanting to do... if you didn't already devise an operable plan. |
| 06:28.26 | bcrisp | p3nguin: what am i wanting to do? |
| 06:28.41 | [TK]D-Fender | module load res_psychic.so |
| 06:28.43 | p3nguin | bcrisp: Set the timeout on the queue's rining to be about twice that of the extensions' Dial() timeout, and also use a followme() in it. |
| 06:28.48 | bcrisp | wonders what AT&T 411 uses.. |
| 06:28.59 | bcrisp | p3nguin: oh cool |
| 06:29.16 | bcrisp | p3nguin: ya the timeout thing was a bitch to figure out, dlynes worked with me on it |
| 06:29.42 | bcrisp | p3nguin, but ya, ill give that a shot some time soon |
| 06:31.19 | p3nguin | So your queue member will be something like Local/2001@internal,1 and [internal]'s 2001 will do the following: Dial(SIP/jack,30), then check dialstatus, and then FollowMe(2001,s) as appropriate. |
| 06:31.37 | p3nguin | Make the timeout on the ringing of the queue member to be about 60-70 seconds. |
| 06:32.31 | bcrisp | yeah, its pretty similar to how it is now |
| 06:32.43 | bcrisp | except i just have the cell#s hardcoded |
| 06:32.57 | bcrisp | but followme will help out there |
| 06:32.58 | bcrisp | ty |
| 06:33.29 | bcrisp | now that my home is being vandalized, robbed, and shot at .. i find the alaramreceived (receivealarm?) application might be handy |
| 06:33.31 | p3nguin | So it will ring the softphone for 30 seconds, run a GotoIf to check that the DIALSTATUS is NOANSWER, and if it is, FollowMe() to whatever you have configured for that user. |
| 06:34.02 | bcrisp | p3nguin, ya .. discovering local channels has opened things up for me |
| 06:35.48 | p3nguin | Having the cell number hard-coded could provide for a more seamless queue, so that's probably better. Replace the FollowMe() with another Dial() for the person's mobile number. |
| 06:35.57 | [TK]D-Fender | checkout time, later all |
| 06:35.59 | bcrisp | ya thats what i have |
| 06:36.43 | bcrisp | by middle of this week i should have most of the required functionality in place.. then its off to vamp up the helpdesk |
| 06:38.03 | p3nguin | Hmm, now I'm wondering if you leave off FollowMe's s option, will it still say "The person you are calling is not at their desk." |
| 06:38.22 | p3nguin | I should test that. |
| 06:38.47 | *** join/#asterisk pids (n=pids@64.62.233.194) |
| 06:38.49 | bcrisp | next i gotta work on meetme and chanspy(or extenspy?) some more |
| 06:39.17 | p3nguin | Ah, without s, it only says "please hold while I try to locate the person you are calling. |
| 06:39.31 | bcrisp | ya.. in a queue that would not be good |
| 06:39.43 | bcrisp | "you are first in line" followed by "let me find the person..." lol |
| 06:40.03 | p3nguin | yep, so FollowMe() is out and hard-coded mobile numbers after a GotoIf to check DIALSTATUS is in. |
| 06:40.14 | bcrisp | "uh, ben... why are the abandons skyrocketing?" |
| 06:40.36 | bcrisp | im not using a gotoif tho |
| 06:42.18 | bcrisp | using something i learned from a voicemail macro: |
| 06:42.18 | bcrisp | http://pastebin.ca/1714105 |
| 06:43.07 | p3nguin | What if the called soft phone turns up BUSY rather than something that would warrant sending the call to the mobile phone? |
| 06:43.14 | bcrisp | i hangup |
| 06:43.39 | bcrisp | so they are back waiting in the queue i believe |
| 06:43.53 | bcrisp | if all the softphones are busy i dont want to dial cell #s |
| 06:44.05 | bcrisp | only if they are unavail or reject the call |
| 06:44.46 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 06:44.49 | p3nguin | I may try replacing my GotoIf with your logic. It kind of seems reasonable. |
| 06:45.08 | bcrisp | i picked it up from the voicemailmacro in the future of telephony book |
| 06:45.53 | p3nguin | If the DIALSTATUS turns out to not be one of NOANSWER or BUSY, then what happens to the call? |
| 06:46.09 | bcrisp | noanswer allows cell phone to dial |
| 06:46.13 | bcrisp | busy returns them to the queue |
| 06:46.24 | p3nguin | But that wasn't what I wanted to know. |
| 06:46.25 | bcrisp | oh |
| 06:46.26 | bcrisp | not be |
| 06:46.53 | bcrisp | yes.. that is where i need the: exten => _353-., 1, .......... |
| 06:47.29 | p3nguin | . will match letters, too? |
| 06:47.34 | p3nguin | I thought it was for numbers. |
| 06:47.41 | bcrisp | hmm.. maybe im wrong there |
| 06:47.49 | p3nguin | try it. |
| 06:48.00 | p3nguin | That's often the best way to find out. |
| 06:48.42 | bcrisp | exten => _353.,1,Goto(353-NOANSWER,1) |
| 06:48.49 | bcrisp | i think thats how ill handle it |
| 06:49.14 | bcrisp | . (period) |
| 06:49.14 | bcrisp | Wildcard match; matches one or more characters, no matter what they are. |
| 06:51.15 | bcrisp | oops, _353-.,1,Goto(353-NOANSWER,1) |
| 06:55.10 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 06:55.19 | bcrisp | i like playing with this stuff and watching CLI go nuts |
| 07:00.47 | bcrisp | time for me to turn in, nite |
| 07:00.51 | bcrisp | and thanks again |
| 07:14.02 | *** join/#asterisk Tim_Toady (n=moi@77.49.232.13.dsl.dyn.forthnet.gr) |
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| 07:41.42 | TJNII | SWEET! The PIC sitting on the workbench behind me responds to ping! |
| 08:05.12 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
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| 08:24.53 | bbt | do i need an fxo or fxs device to connect a fax to an asterisk box? |
| 08:25.44 | bbt | looking at faxing over voip |
| 08:28.05 | ChannelZ | well those two statements are at odds.. |
| 08:28.45 | kaldemar | bbt: fxs |
| 08:29.20 | bbt | are there an recommended fxs devices for asterisk > |
| 08:29.23 | bbt | ? |
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| 09:14.25 | radovan | hi, there's dependency problem with asterisk yum repo. someone who could fix this? |
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| 09:22.35 | sun28 | moin |
| 09:22.46 | sun28 | anyone use asterisk-java ? |
| 09:24.49 | sun28 | interested which framework used, have some experience with GWT |
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| 09:46.40 | guyvdb_ | If an extension is ringing, is it possiable to answer it from a different extention? Ie grab it |
| 09:47.26 | ChannelZ | Yes, in a couple ways |
| 09:47.47 | guyvdb_ | can you point me to any info? A starting point |
| 09:48.07 | ChannelZ | You can use call groups and pickup groups, and then activate the feature in features.conf for pickup, or you can write something custom into your dialplan to pickup specific extensions with the Pickup() app |
| 09:48.46 | guyvdb_ | Is call groups the easiest? I will read up on them and features.conf |
| 09:49.48 | ChannelZ | well it's more flexible I suppose; You can group phones by physicality for instance, and allow only certain extensions to pick up certain other ringing extensions without having to build in all that logic manually |
| 09:50.15 | guyvdb_ | and will that work across SIP and DAHDI? |
| 09:50.38 | ChannelZ | But basically in sip.conf (or whatever channels you're using) you can place devices into specific call groups; Then you can give them a pickup group, which means that extension can pick up a ringing phone which is a member of that call group |
| 09:51.05 | guyvdb_ | ok.. think i get it... I will go try :) thx |
| 09:51.14 | ChannelZ | yeah - see their respective configs (sip.conf, chan_dahdi.conf) |
| 09:51.40 | ChannelZ | you'll see callgroup and pickupgroup in the sample configs |
| 09:51.41 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 09:52.29 | guyvdb_ | one other quick question... if i get an incoming DADHI channel, can I make it ring on multiple SIP channels simultaneously |
| 09:54.14 | ChannelZ | yeah |
| 09:54.31 | ChannelZ | just join them together with & |
| 09:54.41 | ChannelZ | like Dial(SIP/foo&SIP/bar) |
| 09:54.55 | guyvdb_ | ok thx |
| 09:55.26 | ChannelZ | sure |
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| 09:59.00 | *** join/#asterisk gaetronik (n=gaetan@LPuteaux-156-15-5-233.w80-14.abo.wanadoo.fr) |
| 09:59.11 | gaetronik | Hi! |
| 09:59.59 | ChannelZ | HAI! |
| 10:01.18 | gaetronik | i've a question about Gosub behaviour in asterisk 1.6 vs 1.2 |
| 10:02.06 | ChannelZ | hmm I never ran 1.2 |
| 10:02.11 | gaetronik | i used to make a gosub(subroutine,exten,1) and catch invalid exten in subroutine with i exten in 1.2 |
| 10:02.25 | gaetronik | in 1.6 this does not work anymore |
| 10:02.34 | gaetronik | ChannelZ, it's why qe're using 1.6 today |
| 10:03.13 | ChannelZ | I don't get it, you specifically jumped to an invalid extension? |
| 10:03.49 | gaetronik | yes |
| 10:03.57 | gaetronik | it's user entry |
| 10:04.06 | gaetronik | if it exists it's good |
| 10:04.19 | gaetronik | if not say wrong entry |
| 10:06.28 | zoap | ok, zaptel is driving me nuts! i have an astribank device, compiled zaptel and try to get the device up and running... but lsusb | grep e4e4 gives me 1151 all the time... |
| 10:07.14 | ChannelZ | gaetronik: no idea. Gosub's "importance" got increased in 1.6 with the depreciation of Macro so maybe a bug was introduced |
| 10:07.45 | zoap | while /usr/share/zaptel/xpp_fxloader usb it gets loaded alright. any tips? |
| 10:08.31 | zoap | it's concerning asterisk 1.4 by the way |
| 10:08.39 | gaetronik | ChannelZ, i don't know if it's expected or not |
| 10:12.15 | ChannelZ | me either. |
| 10:15.23 | ChannelZ | https://issues.asterisk.org/view.php?id=14411 |
| 10:16.00 | ChannelZ | "there was lots of discussion on the mailing lists about 'i' extension in Gosub(). IIRC the final decision was that the i extension should not trigger. Thus, this bug report became invalid." |
| 10:26.49 | *** join/#asterisk gaetronik (n=gaetan@LPuteaux-156-15-5-233.w80-14.abo.wanadoo.fr) |
| 10:27.11 | gaetronik | thanks ChannelZ |
| 10:28.49 | tzafrir_laptop | zoap, what version of zaptel? |
| 10:29.11 | tzafrir_laptop | what's the output of: zaptel_hardware -v |
| 10:30.30 | gaetronik | is there any way to check if an extension exist in the dialplan |
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| 10:32.31 | gaetronik | if an extension command return -1 |
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| 10:34.44 | gaetronik | is there anyway to catch the error? |
| 10:37.42 | zoap | tzafir_laptop: it's 1.4.12.9:svn.r4653 |
| 10:38.07 | zoap | zaptel_hardware -v gives me no output |
| 10:41.28 | gaetronik | DIALPLAN_EXISTS for my answer |
| 10:47.45 | shadey_ | hi guys, maybe a silly question, but i'm trying to make an outgoing call via a SIP trunk and the FROM header must be sip:CLI@public_IP, my calls originate from sip:internal_sip@public_ip.. how can i set it so that the calls originate from the CLI and not the SIP ID? |
| 10:50.05 | gaetronik | labels in Gosubif is the same as gosub for 1.6 or just like in 1.4 gosub? |
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| 11:13.34 | tzafrir_laptop | zoap, here? |
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| 11:16.59 | pawpro | Hello, Is it possible to send to SIP proxy only some calls (outgoing) and not all SIP signalling? |
| 11:17.06 | zoap | yes, was lunch time, sorry :) |
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| 11:18.26 | zoap | tzafrir_laptop: zaptel_hardware -v gives me no output at all... |
| 11:20.53 | zoap | tzafrir_laptop: will be back in five, got to show up somewhere... hope to see you here then. will not leave the seat for a while then |
| 11:27.40 | zoap | tzafrir_laptop: here |
| 11:30.47 | *** join/#asterisk guyvdb_ (n=guy@196.211.15.234) |
| 11:32.06 | guyvdb_ | When I call a PSTN through DAHDI via a SIP the sidetone is too high. I can hear myself talking on the SIP. What action can I take? |
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| 12:30.20 | guyvdb_ | If I initiate a call from a PSTN line and then hang up, asterisk keeps ringing the extention. I believe I must change something in indicators.conf but what? |
| 12:41.40 | tzafrir_laptop | zoap, now I'm back. Do you have the outputs of those commands? |
| 12:41.45 | tzafrir_laptop | zoap, what version of zaptel? |
| 12:41.47 | tzafrir_laptop | what's the output of: zaptel_hardware -v |
| 12:42.12 | zoap | tzafir_laptop: hold on a sec, just rebooting the server |
| 12:43.32 | zoap | zaptel version is from http://updates.xorcom.com/astribank/zaptel-xpp.tgz |
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| 12:45.15 | tzafrir_laptop | guyvdb_, which type of PSTN: analog? digital? |
| 12:45.53 | zoap | tzafir_laptop: output: http://pastebin.com/d80f9550 |
| 12:46.14 | zoap | did alot in the meantime, so got it working this fa |
| 12:46.42 | tzafrir_laptop | zoap, looks basically ok |
| 12:47.15 | tzafrir_laptop | fixed after the reboot? |
| 12:47.22 | tzafrir_laptop | ah, ok |
| 12:47.29 | zoap | tzafir_laptop: it does. googled a bit and got this far :) |
| 12:47.50 | zoap | have one more problem though, maybe you know where to look. genzaptelconf -svdM gives me: |
| 12:48.03 | zoap | line 11: Framing(T1)/Signalling(E1) must be one of 'd4', 'esf', 'cas' or 'ccs', not '' |
| 12:48.25 | zoap | but when i change it in /etc/zaptel.conf, it just gets removed again... ?? |
| 12:49.28 | *** join/#asterisk michel2 (n=michel2@nl06sr01.targetmedia.nl) |
| 12:49.31 | michel2 | hi there |
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| 12:58.47 | tzafrir_laptop | zoap, please use 'zapconf' rather than genzaptelconf |
| 12:59.17 | *** join/#asterisk voipmonk (n=voipmonk@69.172.114.221) |
| 12:59.24 | tzafrir_laptop | zapconf; dahdi_cfg |
| 12:59.51 | zoap | tzafir_laptop: k, will give that one a try |
| 13:03.22 | zoap | tzafrir_laptop: ok, did zapconf -v, it tells me it generates files, but no channels in zap show channels |
| 13:03.53 | michel2 | hi there, we were making use of asterisk with PHP. Sometimes when I call and I start with a voicefile the script will skip the first 1 or 2 seconds of the voicefile, Does anyone knows how that is possible |
| 13:03.56 | tzafrir_laptop | now run: ztcfg |
| 13:03.56 | zoap | tzafrir_laptop: sure it's me, but hell, this drives me nuts! documenting everything though, so don't have to ask again... :/ |
| 13:04.32 | voipmonk | michel2 add a wait time of about 1 -2 seconds before playing the file |
| 13:05.18 | michel2 | I know, but I wanna know why that the script is doing that. |
| 13:05.21 | zoap | ok, it tells me: 86 channels to configure. |
| 13:05.44 | michel2 | had it something todo with php or is it a systemfailure or just the connection of the caller? |
| 13:06.12 | voipmonk | no |
| 13:06.17 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:07.05 | voipmonk | you have to account for the wacky call setup times |
| 13:07.23 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 13:08.04 | voipmonk | even without your script you will notice a similar outcome when people call in or when you call someone and play a message |
| 13:08.22 | voipmonk | add a wait before the playback and you'll be fine |
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| 13:14.22 | zoap | tzafrir_laptop: ok... did echo '#include zapata-channels.conf' >>/etc/asterisk/zapata.conf et voila! tnx for you patience and help |
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| 13:48.02 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 13:48.16 | NirS | hello all |
| 13:48.18 | NirS | anybody home ? |
| 13:48.33 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:48.39 | oej | Yes, I'm home |
| 13:48.42 | voipmonk | yes |
| 13:49.18 | *** join/#asterisk NirS (n=NirS@87.69.148.100) |
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| 13:49.30 | NirS | this is kinda funky |
| 13:49.38 | NirS | anyone used mplayer to stream MOH music ? |
| 13:49.49 | NirS | all the information available on the web is either outdated or totally wrong |
| 13:50.29 | coppice | but if you delete all the obsolete and confusing info the web would look sooooo small |
| 13:50.59 | NirS | true, true |
| 13:51.19 | NirS | but I would expect a wiki like voip-info.org to be somewhat in the right direction for doing things |
| 13:51.43 | NirS | I've managed to get mplayer to stream into a file, then playing that using Playback |
| 13:51.45 | ManxPower-work | NirS: Not at all. Asterisk changes too fast for the Wiki |
| 13:51.48 | NirS | however, that's not live streaming |
| 13:52.01 | NirS | you have a point there |
| 13:52.30 | NirS | so, anyone has experience with this ? |
| 13:52.31 | ManxPower-work | The most important (and the only official) docs are part of the Asterisk source tarball anyway |
| 13:52.58 | ManxPower-work | NirS: Oh, I learned long ago that trying to stream music into MoH is just more trouble than it's worth |
| 13:53.38 | NirS | well, how about creating some form of WAV based PIPE that can pipe from an internet radio source using mplayer |
| 13:53.48 | NirS | this way, we can use the Playback application |
| 13:53.51 | ManxPower-work | For me it would work for a while, but as soon as there was a glitch in the source I'd have to restart MoH |
| 13:53.57 | NirS | however, couldn't find a way to make it work |
| 13:54.15 | ManxPower-work | NirS: Patches are welcome |
| 13:54.16 | NirS | I'm willing to live with that |
| 13:54.19 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 13:54.33 | NirS | is good, not that good yet - sorry to say |
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| 13:54.49 | NirS | needing 2 gallons of coffee |
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| 13:55.20 | ManxPower-work | joy, our exchange server looks like it's down |
| 13:55.25 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 13:55.45 | *** join/#asterisk rocksfrow (n=kyle@pool-71-179-183-143.bltmmd.fios.verizon.net) |
| 13:55.50 | rocksfrow | hi |
| 13:56.11 | rocksfrow | i currently have a asterisk box setup at my office, and i'm about to setup load balancing across two connections. |
| 13:56.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:56.38 | ManxPower-work | rocksfrow: Are you using something like AsteriskGUI or FreePBX or Trixbox? |
| 13:56.41 | rocksfrow | i'm curious about what problems I'll face (if any), allowing the phone server to bounce across the connections as well. |
| 13:56.45 | rocksfrow | freepbx |
| 13:56.53 | ManxPower-work | ~freePBX |
| 13:56.54 | 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 |
| 13:57.02 | rocksfrow | i'm not asking anything about freepbx |
| 13:57.09 | *** join/#asterisk RypPn2 (i=TuMbL@rosscom.co.uk) |
| 13:57.09 | rocksfrow | i'm asking about the core of asterisk |
| 13:57.15 | rocksfrow | lol |
| 13:57.29 | rocksfrow | immediately push me off to another channel? |
| 13:57.45 | rocksfrow | the question i have concerns asterisk, NOT freepbx bro. |
| 13:58.04 | ManxPower-work | rocksfrow: yes. You are. Any kind of failover would be part of the dialplan. The dialplan is 0wn3d(sp!) by FreePBX. But I'm sure we can help you with the ASTERISK part. |
| 13:58.41 | rocksfrow | My apologees, the old sysadmin set this system up and kind of just...left |
| 13:58.42 | ManxPower-work | There shouldn't odd issue with "load balancing" |
| 13:58.58 | rocksfrow | right, because all of the actual phone traffics going through my t1, right? |
| 13:59.08 | rocksfrow | (i have a separate t1 from my primary internet provider) |
| 13:59.18 | rocksfrow | then a dsl connection, and i'm getting a cable connection |
| 13:59.20 | ManxPower-work | rocksfrow: It depends on how you want to "load balance" |
| 13:59.41 | rocksfrow | ...well, using the 3com vpn firewall, i'd configure what they call 'load balance' vs 'fail over' |
| 13:59.58 | rocksfrow | i guess my main question is... |
| 14:00.14 | ManxPower-work | Ah. Yes that would fail horribly if you are trying to keep active calls running during a "failover" |
| 14:00.15 | rocksfrow | the phone server also is connected to the internet via ethernet |
| 14:00.21 | rocksfrow | ah, really? |
| 14:00.29 | rocksfrow | wait, so traffic is going through the internet connection? not the t1? |
| 14:00.33 | ManxPower-work | Asterisk does not support moving calls between IP addresses. |
| 14:00.40 | rocksfrow | yeah, i figured. |
| 14:00.56 | rocksfrow | so i'm obviously confused, i thought the server was using the t1 for it's traffic |
| 14:00.57 | ManxPower-work | when you say "T-1" do you mean a Voice (CAS) T-1, a PRI, or a DATA T-1 |
| 14:01.05 | rocksfrow | ah.. |
| 14:01.11 | rocksfrow | yea, i'd assume it's a voice |
| 14:01.31 | rocksfrow | which means it's using the dsl for it's communication, and only using that for the call initialization? |
| 14:01.34 | ManxPower-work | OK then. Asterisk does not support moving a call from a voice T-1 to VoIP in the middle of a call. |
| 14:01.43 | rocksfrow | yeah totally, i didn't think so. |
| 14:01.45 | rocksfrow | so basically.. |
| 14:01.56 | rocksfrow | i can just configure the servers port to stick on the one connection then |
| 14:02.11 | ManxPower-work | "configure the server's port"? |
| 14:02.18 | voipmonk | chuckles |
| 14:02.31 | rocksfrow | uh, yeah..the port it's plugged into the switch, configure that port to not load balance across the connections |
| 14:02.33 | ManxPower-work | ~guis |
| 14:02.33 | infobot | well, guis is "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS" |
| 14:02.48 | rocksfrow | i'm not talking about within asterisk bro, lol |
| 14:02.52 | ManxPower-work | rocksfrow: I wish you the BEST of luck. |
| 14:02.56 | rocksfrow | i meant on the switch |
| 14:03.11 | rocksfrow | real quick |
| 14:03.11 | rocksfrow | say i do have a data t1 |
| 14:03.28 | voipmonk | rocksfrow: are you trying to load balance the internet connection or the T1 |
| 14:03.34 | rocksfrow | sorry for the ignorance, i haven't had to touch this thing for a year (yay asterisk) |
| 14:03.51 | rocksfrow | i'm load balancing the internet connections |
| 14:03.57 | rocksfrow | the t1 is separate |
| 14:03.59 | rocksfrow | from both of my connections. |
| 14:04.06 | voipmonk | I see |
| 14:04.38 | voipmonk | u have voip traffic coming into this asterisk system and going out the T1 and vice versa? |
| 14:04.39 | rocksfrow | yeah so i was just trying to figure out whether the server relies on the internet connection for the calls, or on the t1 |
| 14:04.48 | ManxPower-work | voipmonk: Another one that's trying to fly a fighter jet but only knows how to fly a crop duster. They are both airplanes they can't be very different. |
| 14:05.01 | rocksfrow | lol, another smartass |
| 14:05.03 | voipmonk | puts his helmet on |
| 14:05.09 | ManxPower-work | rocksfrow: how would we know what your server is doing? |
| 14:05.12 | rocksfrow | i thought i explained this to you |
| 14:05.25 | rocksfrow | i was sort of left in this situation, i can't just bail out of the plane just bc i dont know how to fly it |
| 14:05.29 | rocksfrow | gotta figure that shit out fool |
| 14:05.48 | ManxPower-work | rocksfrow: You should read the Asterisk book. |
| 14:05.50 | ManxPower-work | ~book |
| 14:05.51 | 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 |
| 14:05.57 | rocksfrow | manxpower-work: yeah..i ordered one,. |
| 14:06.01 | rocksfrow | i think thats it |
| 14:06.29 | ManxPower-work | rocksfrow: Go read the PDF version while you wait for the dead tree version to arrive. |
| 14:06.42 | rocksfrow | dead tree oh no |
| 14:07.25 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-jsajkzqeyeesmseo) |
| 14:08.00 | drmessano | Oh, nice |
| 14:08.12 | ManxPower-work | drmessano: Welcome to Monday on #Asterisk |
| 14:08.16 | *** join/#asterisk Strykar_ (n=wakka@122.169.81.18) |
| 14:08.25 | drmessano | "Whats a phone system?" "That box on the wall next to the Windows 98 server that scares me" |
| 14:08.50 | rocksfrow | windows, lol |
| 14:08.52 | Strykar_ | What's the starting price range for VoIP over PSTN cards for Asterisk? |
| 14:09.08 | ManxPower-work | Strykar_: Assume about $300 |
| 14:09.22 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:09.26 | rocksfrow | dude, you guys seriously are acting like i'm a complete idiot bc i've never managed an asterisk server before that's ridiculous |
| 14:09.29 | ManxPower-work | It will vary based on many issuea. |
| 14:09.30 | Strykar_ | ManxPower-work, Do those do h/w echo cancellation? |
| 14:09.44 | drmessano | rocksfrow: Yes, and yes |
| 14:09.55 | ManxPower-work | Strykar_: no. Those are more expensive. You need to look at the Digium store for pricing info |
| 14:10.03 | rocksfrow | you're lame dude, just bc i don't know shit abt asterisk doesn't mean i'm a common idiot |
| 14:10.10 | rocksfrow | and don't know wtf a server is and use windows |
| 14:10.18 | rocksfrow | 1. windows can suck it |
| 14:10.20 | TJNII | gets popcorn |
| 14:10.22 | *** join/#asterisk Akiraa (i=Akiraa@79.112.30.186) |
| 14:10.23 | ManxPower-work | rocksfrow: no, you're a complete idiot because you are asking for help on a system that you don't know anything about. |
| 14:10.24 | rocksfrow | 2. i've been managing linux servers for years |
| 14:10.26 | [TK]D-Fender | Strykar_: There is no such thing as VoIP over PSTN |
| 14:10.36 | drmessano | 1. Stop jumping up and down and pouting |
| 14:10.37 | rocksfrow | 3. i got left in this situation managing this asterisk server from a previous administrator |
| 14:10.41 | drmessano | 2. Windows 98, ftw |
| 14:10.45 | Strykar_ | ManxPower-work, is h/w echo cancellation neccesary? |
| 14:11.01 | rocksfrow | WTF? lol |
| 14:11.07 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 14:11.07 | rocksfrow | i'm an idiot for asking help on a system i dont know anything about? |
| 14:11.08 | rocksfrow | haahaha |
| 14:11.08 | ManxPower-work | Strykar_: I don't feel that Digium's EC is all that good. I prefer Sangoma. |
| 14:11.11 | rocksfrow | that makes no fucking sense. |
| 14:11.23 | voipmonk | ok I'm done. |
| 14:11.24 | Strykar_ | [TK]D-Fender, Yes, I meant a VoIP card to connect to an analog PSTN network |
| 14:11.27 | rocksfrow | of course i'd ask for help on something i know nothing about, lol. |
| 14:11.28 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:11.29 | ManxPower-work | voipmonk: me too. |
| 14:11.32 | [TK]D-Fender | rocksfrow: chances are 8 is not going to do your load-balancing for you in any appreciable way. It is the wrong tool for the job |
| 14:11.37 | [TK]D-Fender | rocksfrow: SER <------- |
| 14:11.39 | [TK]D-Fender | * |
| 14:11.49 | drmessano | rocksfrow: If it did make sense, we wouldn't nned to call you an idiot |
| 14:11.54 | ManxPower-work | [TK]D-Fender: he doesn't know if he as a voice T-1 or a data t-1. |
| 14:11.56 | [TK]D-Fender | Strykar_: What is a "VoIP card"? |
| 14:12.17 | rocksfrow | what? |
| 14:12.25 | drmessano | ManxPower-work: "Are you wiring this place with 220?" "Yeah, 220, 221, whatever it takes" |
| 14:12.34 | rocksfrow | lol, again..wtf |
| 14:12.37 | [TK]D-Fender | manxpowerAnd hasn't confirmed which direction if not both to balance. Which providers are involved. Confirmed tech... or well... anything pertinent |
| 14:13.06 | ManxPower-work | [TK]D-Fender: correct. |
| 14:13.18 | [TK]D-Fender | rocksfrow: Find out what you've got... then ask. |
| 14:13.18 | drmessano | [TK]D-Fender: STOP ASKING FOR REAL DATA PLZ |
| 14:13.39 | rocksfrow | yeah, as i said previously |
| 14:13.48 | ManxPower-work | maybe if we just ignore him, he'll go away. |
| 14:13.59 | rocksfrow | i'm pretty positive its a voice only :-/, but i have to call the provider today |
| 14:14.10 | rocksfrow | such welcoming channel this is, lol. |
| 14:14.12 | Strykar_ | [TK]D-Fender, ManxPower-work, Can I make Asterisk work over an sDSL line? |
| 14:14.35 | [TK]D-Fender | Strykar_: * doesn't care what kind of tech your packets flow over |
| 14:14.57 | ManxPower-work | rocksfrow: the best way to "load balance" (which I think you mean failover) is to check the result of DIALSTATUS and HANGUPCUSE after the Dial line and decide in your dialplan if you want to send the call via another destination or not. |
| 14:15.03 | ManxPower-work | Now yo have your answer. |
| 14:15.10 | Strykar_ | [TK]D-Fender, That's what I gleaned, hence the build a PC, add card and test idea |
| 14:15.14 | [TK]D-Fender | ManxPower-work: Umm.... ew |
| 14:15.33 | [TK]D-Fender | Strykar_: I fail to see why you need some sort of "card" |
| 14:15.45 | ManxPower-work | [TK]D-Fender: Umm...Total control over call |
| 14:15.57 | Strykar_ | [TK]D-Fender, how do I connect to the PSTN here? |
| 14:16.21 | [TK]D-Fender | Strykar_: How do you want to connect to it? |
| 14:16.44 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 14:16.47 | [TK]D-Fender | ManxPower-work: That's outbound for one, second, that results-based retry-ing |
| 14:17.00 | Strykar_ | I'd like the ability to use an IVR when calls come locally, and for me to use the Internet connection to dial to another Asterisk box in Canada |
| 14:17.19 | ManxPower-work | [TK]D-Fender: you prefer a non-results based methof? |
| 14:17.32 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:17.48 | *** join/#asterisk DocAwesome (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:17.48 | *** mode/#asterisk [+o DocAwesome] by ChanServ |
| 14:18.02 | [TK]D-Fender | ManxPower-work: You await failure to "load-balance" instead of checking current usage. tahts "failure balancing" in my books, and mono-directional. |
| 14:18.10 | Strykar_ | [TK]D-Fender, Let me rephrase that, what are telephony cards, like Digium's 1TDM410PELF required for? |
| 14:18.13 | rocksfrow | i don't mean failover, i mean load balance. but yeah thanks for the answer. dial plan it is. |
| 14:18.39 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 14:18.50 | [TK]D-Fender | Strykar_: that model # is way off, but I do know which you are referring to. That is for plugging in a T! link you already HAVE so * can use it. |
| 14:18.54 | Katty | morning |
| 14:19.03 | rocksfrow | manxpower-work: http://www.3com.com/products/en_US/detail.jsp?tab=features&pathtype=purchase&sku=3CREVF100-73 |
| 14:19.20 | [TK]D-Fender | Strykar_: You don't need a card if you don't already have, or are not planning on getting a PHYSICAL LINK of some sort. |
| 14:19.49 | *** join/#asterisk wopsynator (n=MEGAWOPS@cap31-3-82-227-199-59.fbx.proxad.net) |
| 14:20.01 | [TK]D-Fender | rocksfrow: Not necessarily. |
| 14:20.18 | rocksfrow | [TK]D-Fender, what? |
| 14:22.03 | [TK]D-Fender | rocksfrow: We don't have a solid picture of what you're even running, what your implementation of "load-balancing" should work like, or much else of value. |
| 14:22.03 | [TK]D-Fender | rocksfrow: You are spouting solutions to problems you haven't even clearly defined. |
| 14:23.02 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) [NETSPLIT VICTIM] |
| 14:23.17 | rocksfrow | [TK]D-Fender, my original question was, whether or not asterisk can be configured to NOT rely on the internet connection for the actual phone communication or not |
| 14:23.29 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 14:24.13 | rocksfrow | [TK]D-Fender, basically, so it wouldnt' be affected at all by the connection changing, as the pbx is connected to it's own t1 (separate from the dsl and cable connection) |
| 14:24.22 | rocksfrow | BUT, i've got to figure out whether the t1 the old guy setup is a voice, or data |
| 14:24.25 | drmessano | Sure, it can work over a PRI, GSM, Bluetooth, POTS, and any number of physical connections other than ethernet |
| 14:24.33 | [TK]D-Fender | rocksfrow: And most of those load-balancing is not generally going to work for VoIP because of source-addressing and other issues, etc. Those are generally for balancing TCP communications |
| 14:24.43 | drmessano | You've just shown us shit and asked us to fold it into a duck |
| 14:25.10 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 14:25.11 | rocksfrow | ... |
| 14:25.17 | rocksfrow | is it possible to use the t1 for all communication |
| 14:25.24 | rocksfrow | besides the local voip traffic |
| 14:25.26 | Kobaz | sure why not |
| 14:25.29 | rocksfrow | i mean, to the phones |
| 14:25.34 | rocksfrow | in the office. |
| 14:25.36 | rocksfrow | yes? |
| 14:25.38 | drmessano | Where else? |
| 14:25.44 | [TK]D-Fender | rocksfrow: What kind of T1? You don't need ANY internet connection to run *. |
| 14:25.44 | rocksfrow | huh? |
| 14:25.50 | Kobaz | t1 to a channel bank, will split off channels to analog |
| 14:25.58 | rocksfrow | well |
| 14:26.03 | *** join/#asterisk high-freq (n=hfreq@99.188.101.168) |
| 14:26.03 | [TK]D-Fender | rocksfrow: Feel free to run a complete analog TDm solution if you want to. |
| 14:26.23 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk guyvdb_ (n=guy@196.211.15.234) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 14:26.23 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk hfb (n=hfb@cpe-98-151-255-13.socal.res.rr.com) [NETSPLIT VICTIM] |
| 14:26.23 | *** join/#asterisk ChannelZ (i=channelz@burner.com) [NETSPLIT VICTIM] |
| 14:26.42 | [TK]D-Fender | rocksfrow: Again you are not being CLEAR. T1 is a CARRIER TECH. It is not an end protocol in and of itself |
| 14:26.42 | [TK]D-Fender | rocksfrow: What is on TOP of it? |
| 14:26.51 | rocksfrow | again, sorry for my ignorance on this..BUT, theres a box with the phone line running into it, then i have a regular phone line coming out for my dsl, and a t1 (rj45) coming out of the same box to a t1 card in the server |
| 14:26.53 | Kobaz | [TK]D-Fender: depends how asterisk is configured... heh |
| 14:27.05 | ManxPower-work | [TK]D-Fender: rocksfrow is also running FreePBX, BTW. |
| 14:27.07 | rocksfrow | i'm trying to explain to you, lol. |
| 14:27.09 | Kobaz | [TK]D-Fender: before i found out about dnsmanager, asterisk would completely stop working without an internet connection |
| 14:27.22 | [TK]D-Fender | rocksfrow: YES. * calls out over whatever tech you tell it to and have connectivity to |
| 14:27.30 | rocksfrow | well shit. |
| 14:27.46 | ManxPower-work | Kobaz: there are several ways to fix that issue. |
| 14:28.17 | rocksfrow | [TK]D-Fender, but if i have a voice t1 and not a data t1, that's something i would have to change, correect? |
| 14:28.35 | Kobaz | oh... and there was a nasty bug in 1.4.something, where if you didn't have a default route defined, then asterisk couldn't get it's own ip address, and sip would not work at all |
| 14:28.39 | drmessano | Youre gonna dump a PRI for data? |
| 14:28.44 | [TK]D-Fender | rocksfrow: your "box" breaekdown sounds broken |
| 14:28.52 | *** join/#asterisk mpe_ (n=mpe@gate.ipvision.dk) |
| 14:29.00 | rocksfrow | .... |
| 14:29.04 | rocksfrow | what sounds broken? |
| 14:29.05 | Kobaz | ManxPower-work: other than dnsmanager, what's the other ways? |
| 14:29.14 | rocksfrow | ask me anything and I'll try to answer, thanks for your patience with me |
| 14:29.26 | [TK]D-Fender | rocksfrow: Box has a "phone line" running into it? What kind of "line"? And then how the hell does it spit OUT DSL & T1? |
| 14:29.32 | ManxPower-work | Kobaz: Since at least Asterisk 0.65 if Asterisk could not look up the hostname associated with an IP on the system then it would hang. Most people consider this "poorly setup system" rather than a bug in Asterisk. |
| 14:29.47 | [TK]D-Fender | rocksfrow: Thats like removing a Jumbo-jet from the back of your Volkswagon Beetle |
| 14:29.47 | Kobaz | heh |
| 14:29.51 | Kobaz | i would consider that a bug |
| 14:29.58 | rocksfrow | no no |
| 14:30.00 | rocksfrow | not a phone line |
| 14:30.04 | [TK]D-Fender | rocksfrow: You don't get bigger things from smaller things |
| 14:30.04 | rocksfrow | a rj45 from the BOX |
| 14:30.07 | rocksfrow | man i said this.. |
| 14:30.11 | rocksfrow | again, sorry for my ignorance on this..BUT, theres a box with the phone line running into it, then i have a regular phone line coming out for my dsl, and a t1 (rj45) coming out of the same box to a t1 card in the server |
| 14:30.14 | drmessano | rocksfrow: Does it have a DSL LINE <> running into it, or ETHERNET from the network, which happens to use DSL for inet, connected to it??? |
| 14:30.20 | [TK]D-Fender | rocksfrow: Sorry your description sucks something fierce. |
| 14:30.26 | Kobaz | hehe |
| 14:30.27 | drmessano | ..... |
| 14:30.28 | rocksfrow | this "box" is like...a little gray box that is converting the line to a rj45 |
| 14:30.29 | Kobaz | goes back to coding |
| 14:30.34 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 14:30.34 | rocksfrow | i really sound stupid i know, but i've never seen one b4 |
| 14:30.39 | Kobaz | rocksfrow: smart jack? |
| 14:30.40 | rocksfrow | no experience with t1, at all. |
| 14:30.44 | [TK]D-Fender | rocksfrow: Sounds like a T1 SmartJack |
| 14:30.44 | drmessano | Converting the line? |
| 14:30.49 | rocksfrow | uh yeah |
| 14:30.58 | rocksfrow | said in the dumbest way possible, yes. lol |
| 14:31.20 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:31.36 | [TK]D-Fender | rocksfrow: No, I'm sure you could sink a little further if you tried.....or tried less... either really |
| 14:31.40 | drmessano | and you thought that we may or may not think or not think that you may or may not be stupid |
| 14:31.46 | rocksfrow | ok, from OUTSIDE a regular data line is coming in, you know like cat5e looking, that's going to the gray box owned from the provider, which has BOTH the dsl pair coming out, and then an rj45 |
| 14:32.04 | ManxPower-work | rocksfrow: there is no such thing as a "regular data line". |
| 14:32.05 | drmessano | How misguided or incredibly correct you may or may not be |
| 14:32.10 | rocksfrow | drmessano, call me stupid with ASTERISK / pbx systems all you want |
| 14:32.17 | rocksfrow | but in general, i like to think not |
| 14:32.24 | mchou | lordy |
| 14:32.25 | rocksfrow | what are you talking about |
| 14:32.29 | rocksfrow | a cat5e |
| 14:32.32 | rocksfrow | a fuckign phone line man |
| 14:32.45 | mchou | who's on first? |
| 14:32.45 | rocksfrow | you can use it as phone line atleast |
| 14:32.45 | [TK]D-Fender | rocksfrow: This miracle gray box of yours could jsut be a pretty casing for completely separate lines. |
| 14:32.49 | rocksfrow | just a little thicker, lol |
| 14:32.51 | drmessano | heads for the dryness <> wetness converter box next to the toilet in the bathroom |
| 14:32.59 | ManxPower-work | rocksfrow: if a line has a dialtone on it then it's not a T-1. |
| 14:33.06 | rocksfrow | ..... |
| 14:33.15 | rocksfrow | i can't plug the line into a phone, tis rj45 |
| 14:33.20 | rocksfrow | i mean the voip will not have a dial tone |
| 14:33.23 | rocksfrow | obviously |
| 14:33.31 | mchou | oh dear |
| 14:33.41 | rocksfrow | would a picture be better? lol, geez |
| 14:33.52 | [TK]D-Fender | rocksfrow: YES |
| 14:34.01 | drmessano | [TK]D-Fender: dont hurt him too bad.. the plates in his skull arent completely fused |
| 14:34.03 | rocksfrow | i mean who cares wtf line is going into the box, its COPPER probably 4 pair |
| 14:34.03 | drmessano | BBL |
| 14:34.08 | rocksfrow | thats going to the communication box i'll call it |
| 14:34.18 | Kobaz | rocksfrow: it matters... quite a bit |
| 14:34.22 | rocksfrow | from that box, a single copper pair comes out |
| 14:34.23 | rocksfrow | to the dsl modem |
| 14:34.27 | rocksfrow | and from the SAME box |
| 14:34.30 | rocksfrow | an rj45 |
| 14:34.35 | Kobaz | t1 != dsl |
| 14:34.36 | [TK]D-Fender | rocksfrow: kNo the more generic you are the faster your IQ drops |
| 14:34.46 | rocksfrow | there are two lines coming from the box, that one line is going into |
| 14:35.01 | [TK]D-Fender | rocksfrow: TAKE THE FUCKING PICTURE ALREADY. |
| 14:35.05 | Kobaz | hehe |
| 14:35.10 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 14:35.13 | [TK]D-Fender | rocksfrow: Your descriptions sucks and its getting WORSE |
| 14:35.17 | rocksfrow | w/e bro |
| 14:35.19 | rocksfrow | later |
| 14:35.28 | [TK]D-Fender | rocksfrow: Now pull your head out of your ass and do the smart thing and just SHOW US |
| 14:35.33 | mchou | haha |
| 14:35.34 | Kobaz | too late |
| 14:35.39 | [TK]D-Fender | fuck-tard |
| 14:35.42 | mchou | he's got fios :) |
| 14:36.41 | mchou | gotta love his description |
| 14:36.48 | mchou | copper ibnto a grey box |
| 14:36.49 | drmessano | So he doesnt have copper at ALL? |
| 14:36.56 | mchou | into* |
| 14:36.57 | drmessano | Now I know he's a retard |
| 14:37.05 | drmessano | :) |
| 14:37.08 | [TK]D-Fender | mchou: It may or may not be "4-WIRE" too! |
| 14:37.18 | drmessano | [TK]D-Fender: close |
| 14:37.21 | mchou | [TK]D-Fender: hahah |
| 14:37.26 | drmessano | <PROTECTED> |
| 14:37.31 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 14:37.34 | [TK]D-Fender | Are we sure its grey and not beige... you know how tricky those beige boxes are |
| 14:37.38 | mvanbaak | ~phones |
| 14:37.39 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 14:37.42 | mchou | I thought the company is called 2-wire :) |
| 14:38.13 | drmessano | "What do you have running into your PBX?" "Fios" |
| 14:38.47 | mchou | drmessano: I only said fios cause when he signed off that's what the addr showed up :) |
| 14:38.47 | *** join/#asterisk robl^laptop (n=robl@m3e5336d0.tmodns.net) |
| 14:39.03 | *** part/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 14:39.31 | *** join/#asterisk MGSsancho (i=www-data@www.z4znet.com) |
| 14:39.35 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 14:39.42 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 14:39.46 | drmessano | Yes, I know......... |
| 14:39.56 | MGSsancho | removes his clothes in a sexy manner |
| 14:40.54 | drmessano | mchou: You're a little slow if I had to explain your own joke to you |
| 14:42.54 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 14:47.03 | high-freq | lol is this the comedian chan? |
| 14:47.18 | MGSsancho | only if you put in on youtube |
| 14:47.33 | mchou | no, it's the comedy channel |
| 14:48.13 | high-freq | forgot who pointed me to the 15meg asterisk pdf manual...yall got link? |
| 14:48.38 | mchou | ~book |
| 14:48.39 | 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 |
| 14:48.45 | high-freq | yup yup that be it |
| 14:48.49 | mchou | look at that |
| 14:48.54 | mchou | magic |
| 14:49.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 14:50.08 | high-freq | lol u da man ;) course i'll be havin some questions later after I install this asteriskNOW "testing it out kinda thing" lol...i have access to numerous computers i can use in my lab |
| 14:50.18 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 14:50.21 | high-freq | so i hope u have some smartass remarks too |
| 14:50.40 | *** join/#asterisk montu (n=dpearson@support.databalance.com) |
| 14:51.30 | [TK]D-Fender | high-freq: And it is not a "manual". It is a 3rd party guide to help in unsderstanding * |
| 14:51.46 | *** join/#asterisk telnettech (n=telnette@office.callcopy.com) |
| 14:51.50 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:51.58 | coppice | I've met asses smarter than some of the folk around here |
| 14:52.08 | Naikrovek | that's nice |
| 14:52.19 | [TK]D-Fender | high-freq: And AsteriskNOW is a complete OS + * + FreePBX, etc. tons of 3rd party stuff we don't support here |
| 14:52.24 | high-freq | one thing is do I necessarily need a phone line run to the server? or can i tunnel SIP somewhere cuz i got handed me two D-Link DPH-125MS fones |
| 14:52.38 | high-freq | fender, thx man your making me smarter already :) |
| 14:52.44 | MGSsancho | high, yes |
| 14:52.45 | [TK]D-Fender | high-freq: * can talk with whatever you set it up to talk to |
| 14:53.01 | [TK]D-Fender | high-freq: All VoIP. All Analog. Any kind of mix of techs. |
| 14:53.06 | drmessano | ~youre |
| 14:53.14 | drmessano | :( |
| 14:53.37 | high-freq | hmm cuz they won't allow me to connect the schools fone to the server and wanted to call out to friends and such to their house phones |
| 14:54.13 | *** join/#asterisk gaetronik (n=gaetan@LPuteaux-156-15-5-233.w80-14.abo.wanadoo.fr) |
| 14:54.58 | high-freq | <---nooooobx2 but if i can get this workin today then maybe the school will sponsor me a couple lines and build a pbx system for them lol |
| 14:55.25 | high-freq | i'm fast learner |
| 14:55.34 | drmessano | Are you gonna run voip lines under the tennis courts? |
| 14:55.39 | [TK]D-Fender | high-freq: that phone of yours seems proprietary and unlikely to work with * |
| 14:56.11 | robl^laptop | Grandstream..er.. umm.. Polycom for the win! |
| 14:56.20 | high-freq | hmm not good |
| 14:56.41 | ManxPower-work | not to mention that it takes a week or 8 to understand Asterisk in any meaningful way |
| 14:56.57 | high-freq | got this layer3 switch by netgear the other day, what a pain in the ass to configure. I did notice a VoIP section in there tho i might go check it out |
| 14:57.00 | [TK]D-Fender | high-freq: Well... digging deeper they do mention SIP... its a "maybe". Do you have admin access to configure the phones? |
| 14:57.58 | high-freq | yes everything on my lab i'm under my own subnet which i built with this old cisco2600, layer3 switch netgear GSM7324 and 2 servers, 3 workstations, and now 2 fones |
| 14:57.59 | [TK]D-Fender | high-freq: It does look like a pretty crappy phone either way |
| 14:58.23 | high-freq | ya i know these are the hand-me-downs, the second year students got the brand new ones |
| 14:58.29 | [TK]D-Fender | high-freq: Ok, quite possibly usable, and you can setup whatever you want as far as VoIP goes |
| 14:58.34 | eppigy | BINGO |
| 14:59.10 | montu | Hello all. Anyone got a trick to check if an extension exists before executing a DIAL() in version 1.4.xx of asterisk? Trying to keep asterisk from hanging up on people when they fat finger an extension. |
| 14:59.17 | high-freq | ya finishing up the d/l of the asteriskNOW so i'll be installin it here soon |
| 14:59.35 | *** join/#asterisk SuPrSLuG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:59.44 | [TK]D-Fender | high-freq: Be aware that its centered around a GUI that the book won't help you with, and isn't supported here |
| 15:00.21 | high-freq | ya i been in linux for awhile now so i understand the terminal ;) but then again i do luv my gui |
| 15:01.47 | high-freq | well got 1 week before xmas break so hope i get somethin workin before then |
| 15:02.00 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 15:02.03 | drmessano | Not with those phones you wont |
| 15:02.20 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:02.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:02.25 | high-freq | well i'll just go jack me 2 fones from second year students lol |
| 15:02.28 | [TK]D-Fender | drmessano: They speak SIP, they've been used in lab before and have admin access I don't see why not... |
| 15:03.03 | drmessano | They're not TCP only, designed to work with responsepoint? |
| 15:03.22 | high-freq | i do got an ip on the phones via DHCP if that matters |
| 15:04.15 | [TK]D-Fender | high-freq: Thats like asking "My car has tires on it... so it runs, right?" |
| 15:04.45 | high-freq | lol second smartass comment ;) god i luv yall already |
| 15:04.48 | [TK]D-Fender | high-freq: Yes... congratulations you aren't missing that important, yet single step" |
| 15:05.11 | drmessano | "Its got an internet connector on it, if that matters" |
| 15:05.23 | high-freq | hold on sec...i'm gonna go get me a diff fone lol |
| 15:06.01 | Katty | hmmmm...the squirrels look happy this morning (= |
| 15:06.26 | *** join/#asterisk danj1980 (n=dan@91.108.0.113) |
| 15:07.31 | eppigy | Katty: morning :> |
| 15:07.56 | danj1980 | Hi everyone. |
| 15:08.11 | danj1980 | If got a problem with call dropping. |
| 15:08.17 | danj1980 | I've got* |
| 15:08.58 | Katty | hi eppigy |
| 15:08.58 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:08.59 | Katty | heh, my boss said i had a nice hairdo this morning |
| 15:08.59 | Katty | and it's the same way i put my hair up everyday. |
| 15:09.00 | Katty | some people sure are fruitcakes |
| 15:09.05 | danj1980 | I've enabled debugging in logger.conf and found that whenever the call drops, it doesnt get logged in the mysql csv. |
| 15:09.18 | danj1980 | sorry, in the mysql cdr |
| 15:09.30 | danj1980 | does anyone have any ideas why? |
| 15:09.36 | eppigy | Katty: are you pregnant? |
| 15:10.14 | high-freq | teach asks if these fones gonna work or not? |
| 15:10.59 | [TK]D-Fender | high-freq: grammar fail |
| 15:11.18 | high-freq | ya i know i'm not in engrish class |
| 15:11.39 | high-freq | instead i can be typing 1's and 0's to u if u could understand |
| 15:11.51 | TSM2 | whats a good method to work out ammount of SIP trunks i require for a PBX |
| 15:12.03 | [TK]D-Fender | high-freq: You're definitely a few bits short of a full byte.... |
| 15:12.13 | ManxPower-work | TSM2: how many calls/from the outside world do you need to have at one time? |
| 15:12.14 | [TK]D-Fender | TSM2: .... pardon? |
| 15:12.21 | high-freq | lol |
| 15:12.28 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 15:12.34 | ManxPower-work | ~siptrunk |
| 15:12.35 | infobot | extra, extra, read all about it, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 15:13.05 | [TK]D-Fender | TSM2: You don't require any. I certainly don't |
| 15:13.17 | [TK]D-Fender | TSM2: So neither do you, right? |
| 15:13.40 | defswork | got strangeness with my nokia n97 - if I receive a call when I am remote (over VPN) the calling party cant hear me - I can hear them ok. If I originate a call it's fine. Probably a codec selection issue - any ideas ? |
| 15:13.43 | TSM2 | external SIP service i mean |
| 15:13.48 | high-freq | well no luck on gettin a new fone well i'll try this and if it don't work to hell with it |
| 15:14.02 | TSM2 | defswork: n97 sip software is not very good |
| 15:14.12 | TSM2 | defswork: n97 shit phone |
| 15:14.20 | defswork | TSM2, it-s worked fine for a couple of years |
| 15:14.28 | ManxPower-work | notice how TSM2 didn't answer the most important question |
| 15:14.29 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444445.dsl.bell.ca) |
| 15:14.32 | Katty | eppigy: ^_- |
| 15:14.36 | Katty | eppigy: what? |
| 15:14.45 | TSM2 | defswork: cant be a few years, n97 has only been around for a few months |
| 15:14.57 | dlynes | damn that was a nasty netsplit...there was only 7 or 8 of us left on the other server |
| 15:14.59 | defswork | TSM2, sorry - it's n95 :o |
| 15:15.09 | Katty | eppigy: where did that question come from |
| 15:15.10 | defswork | n97 is shite |
| 15:15.17 | defswork | doesn't ship with sip client |
| 15:15.25 | ManxPower-work | TSM2: you know it's useless to be ACCURATE and CLEAR and CORRECT when asking telecom questions! |
| 15:15.39 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:15.40 | TSM2 | ManxPower: sorry sec |
| 15:16.15 | eppigy | Katty: well it is just oen of those things. Like if you have a glow about you some dude may be like, "Oh I like you're hair today." |
| 15:16.39 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:16.43 | Katty | eppigy: trust me, i am not glowy this morning |
| 15:16.52 | Katty | eppigy: i am grumpy and irritable as usual in the am |
| 15:16.52 | TSM2 | ManxPower: what i mean, is ratio of clients vs external lines |
| 15:17.37 | leifmadsen | Katty: I'm only grumpy and irritable when I'm awake |
| 15:19.04 | *** join/#asterisk newsmafia (n=newsmafi@wsip-98-173-56-18.sd.sd.cox.net) |
| 15:20.18 | dlynes | Katty: btw....jblack was looking for you yesterday |
| 15:20.38 | [TK]D-Fender | TSM2: Ratio is up to you. noone can tell you what you want |
| 15:20.41 | dlynes | Katty: he seemed to be worried about you |
| 15:20.44 | [TK]D-Fender | TSM2: make up your own mind |
| 15:20.53 | danj1980 | Can anyone help with a call dropping issue |
| 15:20.54 | danj1980 | ? |
| 15:21.05 | danj1980 | I'd really appreciate it |
| 15:21.32 | high-freq | Are D-Link phones comapatable with asterisk??? |
| 15:21.47 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 15:22.10 | ManxPower-work | TSM2: that depends on your needs. For large numbers of endpoints IIRC 10% in use is common, but that only is valid for large scale stuff (like a CO). If you have a call center then the numbers will be totally different. |
| 15:22.22 | ManxPower-work | high-freq: any phone that uses standard SIP protocol should work with Asterisk. |
| 15:22.26 | [TK]D-Fender | high-freq: Don't see why not so far. Go try |
| 15:22.38 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 15:22.42 | high-freq | sorry that was my teach lol yes i'll be installin the NOW version soon |
| 15:22.51 | dlynes | high-freq: yeah...they're compatible |
| 15:22.57 | dlynes | high-freq: they're just not great phones |
| 15:22.59 | ManxPower-work | high-freq: head over to #AsteriskNow. |
| 15:23.16 | ManxPower-work | dlynes: I thought many d-link phones ran MGCP |
| 15:23.23 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 15:23.29 | dlynes | ManxPower-work: we've got one in our office running sip |
| 15:23.36 | TSM2 | ManxPower: higher the ammount of clients then then proportian of external lines reduces usualy, ive been thinking about 6-8 lines for 18 clients |
| 15:23.49 | dlynes | ManxPower-work: it just sounds like shit, and has limited programmability |
| 15:24.40 | ManxPower-work | TSM2: I've had installs where 2% of people were on external calls, at other locations 90% of the people were on external calls. |
| 15:25.01 | ManxPower-work | And I've seen sites where 99% of phones were on external calls at the same time. |
| 15:25.13 | TSM2 | defswork: ive got N97, im trying to send it back as its so bad, slow, ive tried SIP and it keeps loosing its registrations, its realy bad |
| 15:25.18 | *** join/#asterisk traxx (n=traxx@91-64-130-76-dynip.superkabel.de) |
| 15:25.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:26.31 | ManxPower-work | TSM2: I'm fond of a mix of PSTN access technologies. PRI as primary PSTN access with rollover to SIP in times of excessive call volume. |
| 15:26.58 | TSM2 | ManxPower: yup it can be a mess, there is no way for me to monitor it at the moment as we just have 8 analog lines (because they wanted 8 DID), ile have to start at 6 or 8 and just upgrade as needed, im just putting through a costings to upgrade from our Nortel CICS to a voip solution |
| 15:27.36 | defswork | TSM - can't have PRI ? |
| 15:27.52 | TSM2 | ManxPower: this is all a cost saving excersise and ive given them an option to run fully on our internet fiber |
| 15:27.57 | *** join/#asterisk michael-i (n=michael-@208.53.198.95) |
| 15:28.15 | ManxPower-work | TSM2: Good thing reliability isn't one of your factors. |
| 15:28.34 | TSM2 | defswork: this will be for US office, i have thought pri but not enough cost savings for them |
| 15:28.51 | TSM2 | ManxPower: reliability is very good, without internet we are dead in the water anyway |
| 15:29.06 | traxx | hi. i have two asterisk servers connected with different extensions registered to them. if a SIP call is placed from an extension on ServerA to a peer on ServerB, then the call is forwarded via an IAX trunk with Dial(). trouble is, the caller id is lost. anybody know what can be done ? |
| 15:29.07 | TSM2 | ManxPower: its a cogent on-net fiber service for the building |
| 15:29.07 | defswork | TSM2, PRI is all I would consider - especially for 8 lines |
| 15:29.19 | eppigy | Katty: :< |
| 15:29.50 | TSM2 | defswork: thats what we have in the UK office but we have more seats than them, aprox 30 |
| 15:29.52 | ManxPower-work | VoIPOverInternet is never as reliable as real telecom lines |
| 15:30.24 | ManxPower-work | VoIPOverPrivateNetwork can be nearly as reliable as real telecom lines. |
| 15:30.43 | TSM2 | does anyone use bandwidth.com , so far ive not heard any negatives, but if they exist i want to hear |
| 15:30.44 | defswork | but far more expensive |
| 15:31.19 | ManxPower-work | defswork: not always. |
| 15:32.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:34.57 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 15:36.59 | *** join/#asterisk lordmortis (n=lordmort@gateway.qsix.com.au) |
| 15:42.21 | TSM2 | so the recomendation is to not use VOIP service providers for reliability, in our case i need to weigh out that without internet at our office no one can do any work at all anyway |
| 15:43.04 | MGSsancho | POTS lines work with out power as well |
| 15:43.13 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 15:43.31 | TSM2 | MGSsancho: not realy if you have a PBX |
| 15:44.34 | MGSsancho | true. very easy to put a UPS on 1 device |
| 15:44.40 | *** join/#asterisk michael-i (n=michael-@208.53.198.95) |
| 15:44.41 | *** join/#asterisk ticoit (n=ticoit@201.191.190.126) |
| 15:44.57 | MGSsancho | how many people protect evert network device they have? :) |
| 15:45.26 | n3hxs | Me |
| 15:46.00 | TSM2 | MGSsancho: yup our pbx and switch cabs are UPS'd up |
| 15:46.21 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:46.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:46.26 | MGSsancho | wait wait wait... are you talking about a compentent network? |
| 15:46.58 | MGSsancho | am i awake? most places i go to a mess or a few work group switched ziptied to a desk |
| 15:47.37 | TSM2 | HP now have a new 8P desk switch that can be POE powered |
| 15:47.45 | TSM2 | i think its HP |
| 15:47.47 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:47.55 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:47.58 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 15:48.41 | Naikrovek | TSM2: those aren't new. cisco's had one of those for a while, gigabit even |
| 15:48.46 | Naikrovek | $pendy tho |
| 15:50.00 | MGSsancho | well it comes down to this, if you care about cost then go for it. if you care about reliable then have a diffrent system |
| 15:51.56 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 15:52.42 | *** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk) |
| 15:52.55 | areay | hi all.. how can i put a call on hold using the manager? |
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| 15:57.50 | montu | anyone know a way to check if an extension exists before DIAL() is executed in ast v. 1.4.2x |
| 15:58.10 | kannan | hello, all on a client xlite softphone, i would like to find if the phone is being used on one line, and if so, go to the next priority in the extensions context. Currently, it rings xlite on line 2 . |
| 15:58.23 | ManxPower-work | montu: nothing in "core show applications" or "core show FUNCTIONS" was helpful? |
| 15:58.29 | kannan | ast 1.4.21.2 |
| 15:58.42 | ManxPower-work | kannan: the easiest thing to do is disable call waiting on the phone |
| 15:58.49 | montu | will double check |
| 15:59.09 | kannan | ManxPower-work, thanks i feel like an idiot, will check |
| 16:02.45 | areay | is it possible to place a call on hold using the manager or cli? |
| 16:07.11 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:07.44 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
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| 16:09.28 | Naikrovek | areay: no idea. not to sound sarcastic but have you asked google? |
| 16:09.47 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 16:09.52 | areay | Naikrovek, yeah lol |
| 16:10.13 | Naikrovek | it probably is possible through the manager api |
| 16:10.36 | areay | i've looked at the manager api commands list but can't find anything relevent |
| 16:10.40 | areay | i can park calls fine |
| 16:10.44 | areay | just not hold em |
| 16:11.34 | [TK]D-Fender | areay: If its not in the docs, odds are it doesn't exist |
| 16:11.34 | Naikrovek | just set up extension 420. huhuhuhhuhhuhhuhuhhu |
| 16:11.51 | [TK]D-Fender | Naikrovek: POTS I presume.... |
| 16:12.12 | Naikrovek | no. not that funny |
| 16:12.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:13.27 | Naikrovek | chocolate give anyone else heartburn? |
| 16:13.42 | n3hxs | Only when you put too much vodka in it. |
| 16:13.43 | Naikrovek | i ate a kit kat about 90 seconds ago and now i have severe heartburn |
| 16:13.47 | *** join/#asterisk pyite_hsv (n=pyite_hs@unaffiliated/pyite) |
| 16:14.16 | kannan | :( , the facility fdoesnt seem to be avilable in xlite phone. I was wondering if there is any settings in the sip.conf that can limit the soft phone to one call |
| 16:14.33 | ManxPower-work | A drunk driver will go right thru a stop sign. A stoned driver will stop at the stop sign and wait for it to turn green. |
| 16:14.44 | kannan | facility = call waiting disabled for line of xlite |
| 16:15.01 | Naikrovek | use ekiga? another free soft phone |
| 16:15.04 | ManxPower-work | kannan: I assumed you looked at all the call limiting features listed in sip.conf.sample. |
| 16:15.39 | ManxPower-work | You can also do it in the dialplan or by having the phone register as a different user for each line. |
| 16:15.50 | Naikrovek | true dat |
| 16:16.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:16.30 | kannan | i guess i have to do it in the dialplan , as the setting in sip.conf prevents the phone from making simultaneous outbound calls and conferencing them also |
| 16:17.27 | kannan | xlite cannot register multiple accounts |
| 16:17.41 | ManxPower-work | kannan: try using a real phone. |
| 16:17.48 | Naikrovek | no kidding |
| 16:17.53 | Naikrovek | how does everyone find xlite |
| 16:18.03 | n3hxs | google |
| 16:18.13 | Naikrovek | yes but why is it always the only one people try |
| 16:18.27 | ManxPower-work | Naikrovek: all softphones suck. xlite sucks less than most. |
| 16:18.40 | kannan | Naikrovek, the users want only xlite , (users = employees) |
| 16:18.43 | Naikrovek | they all suck yes |
| 16:18.56 | ChannelZ | Zoiper is not bad, I like it better than x-lite |
| 16:18.57 | Naikrovek | my users all want photoshop and indesign |
| 16:18.59 | Naikrovek | just for fun |
| 16:18.59 | kannan | zoiper and firefly can easily sort for me |
| 16:19.02 | Naikrovek | they're not getting it |
| 16:19.08 | Naikrovek | users != bosses |
| 16:19.36 | kannan | pl provide a clue on how to get it done in diaklplan? |
| 16:19.42 | kannan | :) |
| 16:19.42 | Naikrovek | CHANISAVAIL |
| 16:19.46 | Naikrovek | or something like that |
| 16:20.06 | ManxPower-work | kannan: "core show applications" "core show FUNCTIONS" |
| 16:20.12 | kannan | it rings straightaway on Line 2 , |
| 16:20.33 | kannan | ManxPower-work , kindly next clue ? |
| 16:20.59 | ManxPower-work | kannan: Every suggestion I have won't work with x-lite. |
| 16:21.10 | came0 | Anyone familiar with the AMI interface? I'm attempting to connect two outside lines (lines 1-5 http://pastie.org/742717 the rest is the response). I can ring my cellphone and then when I answer it rings an extension on the system fine, but when I try to do the same thing using a different outside line it rings my cell phone then when I answer instead of ringing the other line it just says "goodbye" (line 34-42 I assume??) Any idea what I |
| 16:21.10 | came0 | doing wrong here? I'm connecting here via telnet and typing in the command manually. |
| 16:21.10 | Nugget | telnet is eeeeeeevil! |
| 16:21.30 | kannan | ManxPower-work , oh ok, thanks |
| 16:21.39 | came0 | telnet |
| 16:21.45 | Naikrovek | nugget has that on an autoresponse i think |
| 16:21.49 | Naikrovek | only once every few minutes |
| 16:21.53 | Naikrovek | or he's power-lurking |
| 16:21.54 | came0 | Nugget: telnet |
| 16:21.57 | came0 | ah, gotcha |
| 16:22.44 | *** join/#asterisk bzing2 (n=dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 16:22.44 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 16:22.44 | *** join/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 16:22.44 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 16:22.58 | drfreeze | Have a wierd problem: Have a client who says when they are on the phone, when a new call comes in, that niether party of the existing call cannot hear until the new call is answered |
| 16:23.05 | drfreeze | Anyone heard of that before? |
| 16:23.25 | voipmonk | whats inbetween the callers? |
| 16:23.35 | came0 | does anyone use the AMI interface in production servers? or is it still alpha or beta or what? I'm having a hard time finding info on it |
| 16:23.35 | voipmonk | what devices are in between the callers? |
| 16:23.56 | Chainsaw | came0: Interfaces like AsteriskNow seem to use them. |
| 16:23.57 | drfreeze | voipmonk: what do you mean? |
| 16:23.59 | voipmonk | what do you plan on doing with it came0 ? :) |
| 16:24.02 | ManxPower-work | REMEMBER! POTS lines are considered ANSWERED as soon as the DTMF is sent. |
| 16:24.17 | Chainsaw | came0: I'm considering using it for my web interface if it turns out to be faster then calling asterisk -rx. |
| 16:24.33 | voipmonk | drfreeze: what devices such as routers, gateways, ASA's are between the two callers that dont hear anything until the call is connected? |
| 16:24.44 | voipmonk | not hearing anything - like Ringing? drfreeze ? |
| 16:25.31 | Naikrovek | came0: yes, production ready, but not for more than a few simultaneously connected apps |
| 16:25.36 | Naikrovek | apparently it starts to flip out |
| 16:26.00 | came0 | voipmonk: ultimately I'm integrating my asterisk server into my web app with a 'click to dial' interface (similar to google voice and other services) where you type in (or select from previously used numbers) a number to be called on then it rings your phone and then when you answer rings the second party |
| 16:26.11 | drfreeze | voipmonk: one caller is on the other end of the PRI |
| 16:26.36 | came0 | so I'm experimenting with a python package that connects to the AMI but I'm having trouble getting it to work the way I expect so I'm testing by just connecting via telnet |
| 16:26.37 | *** join/#asterisk muiro (n=muiro@rrcs-24-56-88-130.ma.biz.rr.com) |
| 16:26.38 | voipmonk | came0 and Naikrovek , you can use AMI but if you plan to do lots of calls, use astman proxy |
| 16:26.46 | Naikrovek | yes |
| 16:26.48 | came0 | Naikrovek: yeah I was going to say about the proxy |
| 16:27.08 | drfreeze | the internal extension goes thru 2 routers and a pt2pt |
| 16:27.28 | came0 | Chainsaw: what do yo mean by "calling asterisk -rx"? |
| 16:27.34 | [TK]D-Fender | came0: People have been using it for well over half a decade... |
| 16:27.51 | [TK]D-Fender | came0: Works just fine, and plenty of dos on it |
| 16:28.18 | Chainsaw | came0: Exactly that. Calling asterisk -rx "database show agent" or similar. |
| 16:28.21 | drfreeze | voipmonk: also happens with the phones on the same switch as asterisk |
| 16:28.22 | *** join/#asterisk darkskiez (n=mhb@cpc4-broo7-2-0-cust263.know.cable.virginmedia.com) |
| 16:28.49 | ManxPower-work | drfreeze: sounds to me like you have an allow=all or allow=g729 |
| 16:29.13 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-078-211.pools.arcor-ip.net) |
| 16:29.24 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 16:29.26 | came0 | Ok let me research asterisk -rx.. maybe thats a better option... but I assume I'll still have to get mailbox info about messages from the AMI? |
| 16:29.52 | Blackvel | hi. is there any way to dynamically change the language from DE to EN while calling VoicemailMain and recording custom unavailable/busy message? |
| 16:29.57 | drfreeze | ManxPower-work: these three lines are in sip.conf: disallow=all, allow=ulaw, allow=alaw |
| 16:30.08 | Blackvel | cli console set language? |
| 16:30.11 | ManxPower-work | drfreeze: why are you allowing both alaw and ulaw? |
| 16:30.29 | *** join/#asterisk cusco_ (n=trilili@2001:0:53aa:64c:10a6:6bce:a077:acde) |
| 16:30.33 | cusco_ | hi |
| 16:30.41 | cusco_ | queue show - shows me a extensiona s in use |
| 16:30.44 | cusco_ | "in use" |
| 16:30.50 | cusco_ | as in fact it is not in use |
| 16:30.52 | cusco_ | ... |
| 16:32.59 | Blackvel | found it |
| 16:33.07 | [TK]D-Fender | came0>Ok let me research asterisk -rx.. maybe thats a better option... but I assume I'll still have to get mailbox info about messages from the AMI? <- huih? Whats this about "mailboxes" now? |
| 16:33.28 | *** join/#asterisk boch (n=fran@200.61.191.9) |
| 16:34.30 | came0 | member:identifier:[tk]d-fender: Well besides the click to dial I also wanted a little icon in my web app showing number of voicemails in each users mailbox |
| 16:34.57 | drfreeze | ManxPower-work: boilerplate. What do you suggest? |
| 16:35.00 | came0 | [TK]D-Fender ^^ whoops |
| 16:35.26 | ManxPower-work | drfreeze: I suggest allowing the codec you want to use. |
| 16:35.39 | ManxPower-work | drfreeze: "boilerplate"?? |
| 16:35.47 | ManxPower-work | You mean like "from sip.conf.sample"? |
| 16:35.56 | *** join/#asterisk Corydon76-lap (n=Corydon7@nat/digium/x-ffozxcxekjyljeab) |
| 16:35.56 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 16:36.10 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 16:36.56 | cusco_ | where can I read the difference between static agent and dynamic agent? |
| 16:37.07 | ManxPower-work | Or do you mean "That's what that POS GUI I'm using defaults to"? |
| 16:37.20 | drfreeze | ManxPower-work: I haven't edited that code in years |
| 16:37.50 | drfreeze | and it gets copied to new projects |
| 16:38.17 | voipmonk | came0: you can do that with a mysql query... that is if your voicemail is in a mysql db |
| 16:38.21 | ManxPower-work | drfreeze: then it sounds like you have a network, routing, NAT, port forward, or OS issue. |
| 16:39.08 | kannan | I have asterisk with two E1 single span cards. Card -> telco pri , card 2 -> Siemens Hipath PBX. It works fine for a while , the i get console full of errors, and the voice quality is extremely bad. I het the same issue as in -https://issues.asterisk.org/view.php?id=15498. Is it enough if one card has h/w echo cancel module , to be able to cancel out any echo issues etc? |
| 16:39.42 | *** join/#asterisk tftech (n=tftech@63.214.236.169) |
| 16:39.51 | voipmonk | kannan: are you using dahdi or zaptel? |
| 16:40.06 | kannan | dahdi |
| 16:40.34 | voipmonk | are you using that echotraining stuff , too? |
| 16:40.34 | Blackvel | voicemailmain unavailable msg will be recorded to /var/spool/asterisk/voicemail/default/<number>/unavail.gsm/wav? how to make it multi language available? |
| 16:40.38 | ManxPower-work | kannan: What SPECIFIC errors? |
| 16:41.04 | Blackvel | is it better to use playback and copy custom de/en message to /var/lib/asterisk/sounds/ and de/en folders? |
| 16:41.10 | [TK]D-Fender | came0: Yuo can use either. |
| 16:41.20 | voipmonk | Blackvel: what have you tried? |
| 16:41.24 | kannan | pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 floods the CLI |
| 16:41.29 | voipmonk | turns off the SpoonFeed |
| 16:42.02 | ManxPower-work | kannan: Ah. That means "Digium card sucks" or "Interrupt latency too high". |
| 16:42.06 | kannan | voipmonk , i tried disabling the vpmsuport totally, but then there is a echo if people call in |
| 16:42.13 | kannan | the alarms are always OK |
| 16:42.14 | cusco_ | err... the extension should be available |
| 16:42.21 | cusco_ | but shows as "In use" |
| 16:42.21 | Blackvel | voipmonk: recorded my message: please record your message after tone (without the message ppl seem having problem listing just for the tone) |
| 16:42.24 | cusco_ | why? |
| 16:43.02 | voipmonk | Blackvel: because they dont understand the language? |
| 16:43.04 | kannan | if we remove the h/w echo cancel on span 2 ( it is the onbe that provides timing) the other card is fine. If we interchange , then also the same situation |
| 16:43.13 | Blackvel | but i need that as multi-language version. just used voicemailmain to record my custom message. for my whole ivr is used custom records/prompts |
| 16:43.37 | voipmonk | Blackvel: have you recorded your message in different languages? |
| 16:43.40 | Blackvel | with voicemail and silent option there is no message. that is a little bit of a problem for some users (as it seems) |
| 16:43.44 | Blackvel | no. dunno how to do that |
| 16:43.47 | kannan | i mean if we remove the vpm module on span 2 , then it is no prob, will one echo cancelor do the job fully? |
| 16:43.49 | ManxPower-work | kannan: HDLC abort errors are caused by two possible things. 1) errors on the PRI line. 2) corrupted data coming from PSTN card. |
| 16:44.16 | voipmonk | Blackvel: I pray you didnt think Asterisk would record your message magically into different languages for you... |
| 16:44.25 | kannan | ManxPower-work : the whole thing works fine for an hour or more , then it happens |
| 16:44.31 | muiro | in what version of asterisk was cmd Originate put in? |
| 16:44.34 | Blackvel | <PROTECTED> |
| 16:44.35 | ManxPower-work | kannan: That does not change my statement. |
| 16:44.53 | Blackvel | can i create an "en" folder in spool directory? |
| 16:45.02 | Blackvel | that works for /var/lib/asterisk/sounds |
| 16:45.05 | voipmonk | Blackvel: ok backup - if I call into your app, what do you want me to hear |
| 16:45.09 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 16:46.18 | Blackvel | if the person pressed "1" or something in ivr i call voicemail. i need a custom record prompt (either automatically by using custom prompts like busy message / unavailable message) or by recording my own de/en prompts and using Playback |
| 16:46.36 | Blackvel | not sure what is the better / easier option |
| 16:46.45 | kannan | ManxPower-work : does 2) corrupted data = bad card? . We tested the pbx with sip phones and ATAs , with the PSTN card (span 1 ) it is ok. |
| 16:46.51 | Blackvel | can custom prompt be recorded for both DE/EN ? |
| 16:46.55 | [TK]D-Fender | Blackvel: make your recordings in VoiceMailMain like everybody else |
| 16:47.07 | Blackvel | yes did that with DE language |
| 16:47.08 | ManxPower-work | kannan: no 2) usually means something is locking interrupts for so long the data is getting corrupted. |
| 16:47.14 | Blackvel | it recorded unavail.gsm |
| 16:47.36 | Blackvel | how can i make a new custom recording for language EN? and where does it record? will it overwrite old DE prompt? |
| 16:47.38 | kannan | ManxPower-work : how to check that is happening? |
| 16:48.42 | ManxPower-work | kannan: this is one of the most complicated issues in Asterisk to solve. Many people never solve it. Contact Digium support. |
| 16:49.04 | *** join/#asterisk cusco__ (n=trilili@2001:0:53aa:64c:4cc:1941:2ac0:762d) |
| 16:49.07 | cusco__ | re |
| 16:49.08 | kannan | ManxPower-work : thank you again |
| 16:49.22 | drfreeze | ManxPower-work: the calls work fine |
| 16:49.35 | drfreeze | The problem does not happen until the new call comes in |
| 16:50.12 | kannan | ManxPower-work : does the same thing happen with Sangoma cards also , any idea? |
| 16:50.32 | ManxPower-work | kannan: I've never had HDLC aborts in a Sangoma card. |
| 16:51.27 | kannan | ManxPower-work : good to hear that, we can even get new cards. Coz Siemens charges much more for a new cord , that can do something else like qsig, cas etc |
| 16:51.34 | kannan | card |
| 16:52.04 | kannan | for each swtich type etc, we have to get a new or additional card |
| 16:52.12 | ManxPower-work | kannan: At least go thru the motions of trying Digium support. My guess is all they will tell you to do is "upgrade to the latest DAHDI", but at least you will have tried. |
| 16:52.50 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:53.05 | x86 | SIP over TCP is _way_ more NAT friendly than UDP it turns out |
| 16:53.13 | x86 | works very, very well for me thus far |
| 16:53.24 | Naikrovek | nice |
| 16:53.30 | kannan | sip trunking also very costly in siemens |
| 16:53.40 | Qwell | x86: SIP isn't the problem. RTP is. |
| 16:53.40 | x86 | would like to get SIP over TLS working well too, but so far haven't been very successfull at getting that setup |
| 16:53.44 | x86 | Qwell: right |
| 16:53.56 | Blackvel | can i call set application on CLI? |
| 16:53.57 | x86 | Qwell: well, fundamental design flaw of SIP, really ;) |
| 16:54.20 | Qwell | what I'm suggesting is that simply switching signaling to TCP won't do anything to help NAT issues |
| 16:54.33 | x86 | it sure helped me out |
| 16:54.40 | [TK]D-Fender | Blackvel: Yuo can't. that is channel based and there is no channel on CLI |
| 16:54.43 | cusco__ | hi |
| 16:54.44 | *** join/#asterisk levity (n=levity@unaffiliated/canuck) |
| 16:55.02 | x86 | had NAT on both ends and the call would get hung up after a few seconds with UDP because of a re-trans issue |
| 16:55.05 | leifmadsen | TCP and UDP are transport layers, and have nothing to do with SIP other than transporting messages |
| 16:55.20 | x86 | still have NAT on both ends, now using TCP instead of UDP, and no problems at all |
| 16:55.30 | leifmadsen | it's not SIPs fault if your network allows TCP through better than UDP -- that's a network issue |
| 16:55.30 | x86 | leifmadsen: exactly my point |
| 16:55.49 | leifmadsen | you said it was a fundamental design flaw of SIP |
| 16:56.08 | cusco__ | queue in use... |
| 16:56.16 | cusco__ | a extension is showing in use forever |
| 16:56.21 | cusco__ | no queue is ringing on it |
| 16:56.26 | *** part/#asterisk robl^laptop (n=robl@m3e5336d0.tmodns.net) |
| 16:56.28 | cusco__ | how can I debug that? |
| 16:56.30 | x86 | leifmadsen: yes, to use UDP and a different random port for every stream of RTP |
| 16:57.31 | p3nguin | That's not a flaw in SIP/RTP, but it is a flaw in your networking if you don't accommodate those ports. |
| 16:57.32 | voipmonk | hrmm rtp.conf ight help you narrow the randomness down a bit |
| 16:57.47 | voipmonk | ight = might |
| 16:57.47 | x86 | well sure |
| 16:58.03 | x86 | p3nguin: I think it's a flaw in SIP to use UDP, period |
| 16:58.10 | x86 | design flaw, rather |
| 16:58.26 | Blackvel | [TK]D-Fender: ok |
| 16:58.30 | voipmonk | design a better way, x86 , please. |
| 16:58.44 | p3nguin | I don't see a problem with SIP using TCP, but for voice packets, TCP really doesn't make sense. |
| 16:59.06 | voipmonk | voice packets...? |
| 16:59.07 | x86 | not saying there are ways to properly implement it, but with TCP you have proper connection tracking and you can allow related connections (which is the default behavior for most NAT/firewall devices) |
| 16:59.24 | p3nguin | cusco__: "a extension" ... perhaps you mean a channel. |
| 16:59.55 | voipmonk | x86: - learn the law of the land first, then attempt to change it :) |
| 17:01.09 | Blackvel | so voicemailmain unavailable message was not designed to be recorded as a multi language version? :) |
| 17:01.29 | x86 | so with TCP, subsequent RTP streams after the SIP session is setup, are considered "related" and are just simply allowed through the NAT/firewall device without a gaping range of holes pre-made for RTP traffic in the firewall |
| 17:01.39 | cusco__ | p3nguin: yes, I guess I do |
| 17:01.53 | ManxPower-work | x86: what universe are you in? |
| 17:02.28 | p3nguin | cusco__: What does "core show channels verbose" say about the channel you're having trouble with? |
| 17:02.30 | x86 | ManxPower-work: what do you not understand about my logic? |
| 17:03.16 | ManxPower-work | x86: TCP isn't going to magically allow UDP to work. |
| 17:03.34 | ManxPower-work | it's just as easy to make RTP packets "related" to SIP/UDP as it is for SIP/TCP. |
| 17:03.43 | TommyBotten | x86: How would the firewall know that the RTP is related to the TCP/SIP? |
| 17:03.45 | x86 | ManxPower-work: I never said "TCP is going to allow UDP to work"! |
| 17:03.57 | x86 | ManxPower-work: i was saying TCP WORKS BETTER THAN UDP IN NATTED ENVIRONMENTS! |
| 17:04.03 | x86 | ManxPower-work: DUE TO CONNECTION TRACKING |
| 17:04.12 | x86 | ManxPower-work: (perhaps if all caps, you can read it better?) |
| 17:04.15 | p3nguin | That's a retarded thing to say. |
| 17:04.15 | TommyBotten | x86: You need a firewall that understands the protocol. Like connection tracking |
| 17:04.55 | x86 | ManxPower-work: you can't do related connections with UDP because UDP is "connectionless" |
| 17:05.06 | p3nguin | x86: Actually, it has been scientifically proven that a sentence written in all caps is more difficult to read than one which has been correctly capitalized. |
| 17:05.19 | x86 | p3nguin: noted ;) |
| 17:05.26 | cusco__ | p3nguin: how can I know wich channel is being taken by the extension? |
| 17:05.35 | cusco__ | that does not show the extension |
| 17:05.39 | cusco__ | it is not in use |
| 17:05.51 | ManxPower-work | x86: AHA! You are correct! That's why you can't make a response to a DNS request get automatically forwarded back to the device making the DNS request when in a NAT'd enviroment! |
| 17:05.51 | kannan | in a SIP DID, i used Background for IVR, it sometimes takes quite a few seconds to start ringing the dialled extension, other times it dials out at once, what settings could be used to ensure immediate dial out? |
| 17:05.52 | cusco__ | but "queue show" shows it in use... " |
| 17:05.53 | p3nguin | cusco__: If you look at both the channel and extension columns, the information is listed. |
| 17:06.04 | cusco__ | no, the extension is not there |
| 17:06.19 | ManxPower-work | kannan: stop using variable length overlapping extensions. |
| 17:06.32 | voipmonk | kannan: match more appropriately in your dialplan |
| 17:06.42 | x86 | ManxPower-work: that has nothing to do with TCP connection tracking |
| 17:06.51 | x86 | ManxPower-work: you totally missed my point... |
| 17:06.54 | ManxPower-work | x86: I'm taking about UDP tracking. |
| 17:06.57 | voipmonk | kannan: Background allows you to enter dtmf during the message playback and will look insdie the context for a match |
| 17:06.59 | x86 | I'm not |
| 17:07.28 | voipmonk | kannan: for example... if u have _123XXXX. instead of _123XXXX it will take longer based on the timeouts u preset |
| 17:07.36 | areay | is there really no way of placing a call on hold using the manager api? would creating an extension and using MusicOnHold() in extensions.conf work? |
| 17:07.38 | voipmonk | kannan: see how that works? |
| 17:07.41 | p3nguin | Even though UDP is "connectionless," iptables can still keep information about the UDP "connection." |
| 17:07.52 | ManxPower-work | p3nguin: amazing, isn't it. |
| 17:07.57 | p3nguin | quite |
| 17:08.08 | ChannelZ | It's magic, bitches! |
| 17:08.25 | kannan | there is only one priority other than the DID (by Caller ID lookup) , which is a _1xx,1,Dial(SIP/my-extension) in the context |
| 17:08.43 | kannan | plus t and i extensions |
| 17:08.50 | ManxPower-work | kannan: and I bet when you pick option 1 in your IVR it takes a whle. |
| 17:08.50 | p3nguin | Is this a loop, or what? |
| 17:09.08 | *** join/#asterisk Tim_Toady (n=moi@77.49.232.13.dsl.dyn.forthnet.gr) |
| 17:09.09 | voipmonk | ok kannan - not that i dont trust your deduction skills....... ok Im lying - show me some pastebin.... :) show me your dialplan |
| 17:09.30 | ManxPower-work | voipmonk: I bet he has an include |
| 17:09.33 | kannan | voipmonk, ok , 2 mins pl |
| 17:09.46 | voipmonk | or worse |
| 17:09.50 | voipmonk | <PROTECTED> |
| 17:09.52 | voipmonk | statement |
| 17:09.54 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 17:10.14 | kannan | yes, but not in that context .. |
| 17:10.27 | voipmonk | or a php script that needs an ack from x10 when the toilet flushes |
| 17:10.29 | kannan | #include and switch used , but other contexts.. |
| 17:11.08 | ManxPower-work | kannan: remove the include and the switch |
| 17:12.26 | kannan | ManxPower-work, voipmonk, ok i will try |
| 17:12.46 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 17:12.53 | kannan | but then it doesn 't happen always ... |
| 17:13.48 | ManxPower-work | kannan: "dialplan show <context>" |
| 17:15.40 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:20.24 | areay | is there really no way of placing a call on hold using the manager api? would creating an extension and using MusicOnHold() in extensions.conf work? |
| 17:20.59 | voipmonk | you can transfer them to very special place |
| 17:21.27 | areay | thats what i thought... create an extension 7777 or something that just plays music on hold |
| 17:21.35 | areay | and then transfer the channel back when you unhold |
| 17:22.05 | p3nguin | Why not use the hold button on the phone? |
| 17:22.10 | drfreeze | This is just crazy |
| 17:22.47 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
| 17:22.50 | areay | p3nguin, i'm using a command-line phone on auto answer |
| 17:23.00 | drfreeze | 2 people on the phone, when a third call comes in, the two existing calls cannot hear the caller. When the 3rd call is answered, audio returns after about 4 seconds |
| 17:23.00 | areay | p3nguin, ideally i'd like to keep all phone controls away from the user |
| 17:23.29 | areay | other than the ones that are needed of course |
| 17:23.53 | areay | but i want to restrict even those... the asterisk manager api is perfect, but the one missing function is hold/unhold :( |
| 17:24.45 | kannan | ManxPower-work, voipmonk , Dialplan show my-context ->http://pastebin.ca/1714598 |
| 17:25.09 | drfreeze | Could this be the new polycom firmware? |
| 17:25.15 | kannan | http://pastebin.ca/1714598 |
| 17:26.48 | Blackvel | i recorded my custom de/unavail.alaw and en/unavail.alaw message. do i have to active it in voicemailmain? it plays vm-theperson + digits + vm-is-unavail + beep |
| 17:27.11 | Blackvel | maybe alaw does not work? it has to be wav or gsm? |
| 17:31.30 | Corydon76-lap | Blackvel: why de/unavail ? |
| 17:31.49 | p3nguin | German unavailable message? |
| 17:32.14 | Corydon76-lap | I don't know that for voicemail, you can do separate languages |
| 17:33.00 | Corydon76-lap | Blackvel: what version are you running? |
| 17:34.38 | Blackvel | i recorded my own message (REcord application) but want voicemail to play unavail msg :) |
| 17:37.47 | Corydon76-lap | I can pretty much guarantee that 1.4 doesn't support multiple languages, and given the codepath, it's unlikely that has changed at all. |
| 17:38.27 | *** join/#asterisk Alagar (n=Administ@122.164.34.230) |
| 17:40.21 | Corydon76-lap | Yep, no go on multiple languages for the unavail message, sorry |
| 17:41.53 | Blackvel | http://www.voip-info.org/wiki/view/Asterisk+Voicemail |
| 17:41.54 | Corydon76-lap | relatively easy fix, if you want to file an issue on it |
| 17:41.58 | Blackvel | http://www.voip-info.org/wiki/view/Asterisk+Voicemail |
| 17:42.12 | *** join/#asterisk frag4now (n=frag4now@unaffiliated/frag4now) |
| 17:42.32 | Blackvel | it works with normal playback/record |
| 17:43.19 | cusco__ | logger.c won't log both to mysql and to queue_log |
| 17:43.22 | Corydon76-lap | I understand you, but the fact is that we check whether the file exists with a NULL language component, which doesn't check for the presence of a language directory |
| 17:43.25 | cusco__ | the abandons incommings etc |
| 17:43.36 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 17:43.45 | Blackvel | not sure on voicemail. i want to activate the "play unavailable message" for the application |
| 17:43.52 | cusco__ | we keep changing the sourcecode (jsut taking the else off) to make it write in both msyql and queue_log file |
| 17:43.58 | Blackvel | not sure how it recognized that there is a custom file |
| 17:44.01 | Corydon76-lap | Blackvel: you could trick it with an unavail message in a non-subdirectory |
| 17:44.15 | voipmonk | or review the source and make changes |
| 17:44.23 | Corydon76-lap | That is, en/unavail, de/unavail, and unavail |
| 17:44.32 | Blackvel | does it scan /var/spool/asterisk/voicemail/default/<exten> unavail.wav? |
| 17:44.52 | Blackvel | for the moment it just plays default |
| 17:44.53 | Corydon76-lap | It doesn't scan. It calculates and verifies that path |
| 17:45.24 | Blackvel | how does voicemail know when to play standard or custom msg? |
| 17:45.45 | Corydon76-lap | Blackvel: it calculates the path and checks whether that specific path exists or not |
| 17:46.17 | Corydon76-lap | As I said, no scanning. |
| 17:47.00 | Corydon76-lap | Blackvel: read everything I said above and it should make sense |
| 17:47.36 | frag4now | can i use an asterisk server to act as answering service? |
| 17:47.57 | Blackvel | Corydon76-dig: so it checks path with default unavail.wav file? |
| 17:48.18 | Blackvel | oh |
| 17:48.25 | voipmonk | frag4now: you can configure asterisk to be an answering system, yes |
| 17:48.29 | Blackvel | i missed one sentenance of you above....sorry |
| 17:49.01 | frag4now | voipmonk: is this configuration difficult for a newbie(like me)? can i find some docs somewhere? |
| 17:49.37 | Blackvel | Corydon76-dig: wont it play unavail and ignore de/unavail or en/unavail? |
| 17:50.49 | Corydon76-lap | Blackvel: why don't you try it? |
| 17:51.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:51.49 | m0e | i have a quick question :) Does this look correct? It's supposed to match when the first digit is a 9 and then remove the 9 and dial the number through the dahdi group 1 channels |
| 17:51.51 | m0e | exten => _9|.,3,Dial(DAHDI/G1/${EXTEN:7}) |
| 17:51.59 | Blackvel | and your guess is without custom unavail file in default spool mailbox directory it moves forward with standard msg? |
| 17:52.10 | Blackvel | yes sure...may be an option |
| 17:52.26 | *** join/#asterisk pyite_hsv (n=pyite_hs@unaffiliated/pyite) |
| 17:52.38 | ManxPower-work | m0e: | is not valid |
| 17:52.48 | *** join/#asterisk dlynes (n=chatzill@bas5-hamilton14-1242444445.dsl.bell.ca) |
| 17:53.05 | m0e | ok.. i thought it would be the same as for freepbx.. could you guide me towards a document for matching? |
| 17:53.07 | p3nguin | m0e: That's wrong. It should be _9. and ${EXTEN:1} |
| 17:53.26 | ManxPower-work | ${EXTEN:numberofdigitstostrip} |
| 17:53.32 | m0e | what is that :1? i read a tutorial that had :7 and didnt explain it |
| 17:53.33 | m0e | ooooh |
| 17:53.35 | m0e | perfect! |
| 17:53.37 | m0e | thanks :) |
| 17:53.46 | ManxPower-work | m0e: read channelvariables.tex in the asterisk source tarball. |
| 17:53.54 | p3nguin | m0e: _9NXXXXXX if you want to dial 9 and 7 more digits. |
| 17:53.54 | m0e | will do, thanks |
| 17:54.17 | voipmonk | ok |
| 17:54.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:54.22 | ManxPower-work | ~guis |
| 17:54.23 | infobot | it has been said that guis is "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS" |
| 17:54.31 | voipmonk | be back later... |
| 17:54.44 | voipmonk | glances at manxpower ....good luck |
| 17:54.53 | voipmonk | dangles the SpoonFeed in front of TK |
| 17:54.56 | Corydon76-lap | ManxPower-work: If I use ${EXTEN:a:b} and both a and b are negative, what does it do? |
| 17:55.14 | ManxPower-work | Corydon76-lap: I would have to look at channelvariables.tex. 8-) |
| 17:56.02 | p3nguin | m0e: Also, consider ignorepat in that particular context. |
| 17:56.20 | m0e | will go read about that as well, thanks |
| 17:56.30 | ManxPower-work | p3nguin: ignorepat is only for ZAP / DAHDI channels |
| 17:56.39 | m0e | i'm sorry for being such a noob, but the gui has made me ignorant.. i'm trying to get a feel for how things actually work |
| 17:56.45 | p3nguin | and he's using DAHDI, so we're good. |
| 17:56.52 | Katty | hi |
| 17:57.19 | Corydon76-lap | ManxPower-work: it doesn't say ;-) |
| 17:58.11 | cusco__ | ls |
| 17:58.17 | cusco__ | er.. |
| 17:58.33 | cusco__ | why does an extension on "queue show" has (In use) on it? |
| 17:58.46 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 17:58.59 | ManxPower-work | cusco__: stop saying "extension" when you means something else. |
| 17:59.03 | p3nguin | I don't think queue show lists anything abou extensions. |
| 17:59.08 | cusco__ | 611 (Local/611@agents) with penalty 110 (realtime) (In use) has taken no calls yet |
| 17:59.30 | cusco__ | ok, what is it, a channel? |
| 17:59.39 | ManxPower-work | that is a channel sort of. |
| 17:59.43 | cusco__ | member |
| 17:59.49 | cusco__ | ok |
| 17:59.52 | p3nguin | Could be a channel, could be a device. Only you know, but it's certainly not an extension. |
| 18:00.00 | cusco__ | why is the member's channel marked as "In use"? |
| 18:00.07 | cusco__ | that should only happen when its ringging, right? |
| 18:00.15 | ManxPower-work | cusco__: an EXTENSION in Asterisk is exactly one thing. an exten => line in extensions.conf. |
| 18:00.19 | p3nguin | Maybe they forgot to hang up the phone. |
| 18:00.35 | *** join/#asterisk etfonhomey (n=etfonhom@32.179.6.247) |
| 18:00.51 | ManxPower-work | cusco__: what device is dialed from extnsion 611 in context agents? |
| 18:01.24 | cusco__ | dialed from 611 ? |
| 18:01.27 | cusco__ | or to 611 ? |
| 18:01.41 | cusco__ | im not sure, how do I check? |
| 18:01.42 | p3nguin | 611 is an exten in extensions.conf. |
| 18:01.43 | ManxPower-work | in exten => 611 in context [agents] what is the Dial line? |
| 18:01.54 | p3nguin | What does it do when you call 611 from a phone? |
| 18:02.02 | cusco__ | ah, hold |
| 18:02.16 | ChannelZ | my garage door opens |
| 18:02.24 | p3nguin | MINE TOO! |
| 18:02.43 | p3nguin | I wondered why it kept doing that! Stop dialing 611, please. |
| 18:02.54 | cusco__ | that happens: http://paste.debian.net/53979/ |
| 18:03.16 | p3nguin | holy wtf |
| 18:03.31 | cusco__ | :/ I know... |
| 18:03.37 | Katty | holywtf? |
| 18:03.48 | Katty | holywtfbatman |
| 18:03.48 | ManxPower-work | looks like you set your SIP USERID to be the same as the "extension". |
| 18:04.02 | ManxPower-work | so does "sip show peer 611" show that it's in use or anything like that? |
| 18:05.08 | cusco__ | http://paste.debian.net/53980/ |
| 18:05.37 | cusco__ | doesn«t have a "in use" information |
| 18:06.01 | Naikrovek | never saw a french quote in an english word before |
| 18:06.17 | ManxPower-work | cusco__: try "sip show channels" |
| 18:06.21 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:06.41 | cusco__ | user 611 not listed |
| 18:07.00 | [TK]D-Fender | Irrelevant |
| 18:07.00 | *** join/#asterisk loather (n=loather@wsip-98-175-250-116.sd.sd.cox.net) |
| 18:07.28 | cusco__ | http://paste.debian.net/53981/ |
| 18:07.42 | ManxPower-work | cusco__: then chances are you do have an actual queue issue rather than just a device issue |
| 18:07.48 | *** join/#asterisk Alagar (n=Administ@122.164.34.230) |
| 18:08.10 | [TK]D-Fender | Looking at SIP = waste so far |
| 18:08.27 | ManxPower-work | [TK]D-Fender: dude you want a full sip debug for everything. |
| 18:08.38 | cusco__ | lol |
| 18:08.52 | ManxPower-work | User: I'm having a problem with [TK]D-Fender: SIP DEBUG NEEDED! |
| 18:08.56 | ManxPower-work | 8-) |
| 18:11.17 | [TK]D-Fender | manxpowerNo I don't |
| 18:11.42 | [TK]D-Fender | ManxPower-work: He has a MEMBER issue and we are not looking at CALLS in progress |
| 18:11.45 | p3nguin | only for things relating to SIP, right? |
| 18:11.58 | [TK]D-Fender | NO |
| 18:12.04 | p3nguin | :) |
| 18:12.21 | ManxPower-work | <PROTECTED> |
| 18:12.31 | [TK]D-Fender | ManxPower-work: Still irrlevent |
| 18:12.59 | came0 | Ok I guess I'm having the same problem initiating a call from CLI or asterisk -rx as I was with the AMI. I can do "originate SIP/9bs638/14075551212 extension 100@default" and it will ring my cell and then ring extension 100 when I answer. But what do I use for the extension argument if I want to dial an outside line? originate SIP/9bs638/14075551212 extension SIP/9bs638/14075552222@default? or something similar.. I cant get it to ring t |
| 18:13.00 | came0 | second line |
| 18:13.02 | loather | ok, all my phones are set up to allow g.729 for calls. In my sip.conf I have disallow=all ; allow=g729,g723,gsm,ulaw |
| 18:13.18 | loather | why are the majority of calls being negotiated as ulaw instead of g729? |
| 18:13.36 | p3nguin | loather: Where did you set the preferred codec? |
| 18:13.47 | loather | sip.conf general |
| 18:13.55 | ManxPower-work | loather: Asterisk will usually pick the codec with the least CPU overhead. |
| 18:14.10 | [TK]D-Fender | came0: originate SIP/9bs638/14075551212 extension 14075552222@whatevercontexthasanextentomatchanddowhatyouwantdefault |
| 18:14.13 | ian6 | lol bhab |
| 18:14.14 | p3nguin | loather: If you notice in sip.conf, it says that the order you list the codecs does not constitute preference. |
| 18:14.25 | [TK]D-Fender | came0: And you should NEver have ro use a context named [default] |
| 18:14.45 | loather | ManxPower-work: I clearly don't want it to do that. If I tell it to pick g729, there's a reason. |
| 18:14.48 | loather | lol zip |
| 18:15.10 | came0 | [TK]D-Fender: Ok thanks again man. I think I need to do some reading on contexts. I dont know what they mean. |
| 18:15.26 | loather | p3nguin: and incedentally, it doesn't say that in my sip.conf. |
| 18:15.54 | ManxPower-work | loather: you are not telling it to pick G729, yo are telling it pick any of these codecs. |
| 18:16.11 | loather | Then how do I tell it to prefer g.729? |
| 18:16.23 | ManxPower-work | loather: I don't know if you can tell Asterisk that. |
| 18:16.27 | p3nguin | Well, I thought it was in sip.conf, but I guess it's possible that it is in another piece of documentation. |
| 18:17.01 | ManxPower-work | loather: do you have a g729 license or transcoder card? |
| 18:17.06 | loather | ManxPower-work: All I know is it worked perfectly on asterisk 1.2. Ever since the move to 1.6, it's been nondeterministic. |
| 18:17.08 | loather | licenses. |
| 18:17.31 | loather | 48 concurrent |
| 18:17.52 | ManxPower-work | loather: why not force all calls to be g729? |
| 18:17.55 | cusco__ | any hint? what should I be looking at, in the queues configuration? |
| 18:18.37 | loather | because some of my phones don't support g.729 |
| 18:18.43 | ian6 | get better phones. |
| 18:18.55 | loather | haha, working on it :) |
| 18:18.55 | ManxPower-work | loather: so set the peers for those phones to not use g729. |
| 18:19.09 | Corydon76-lap | loather: in sip.conf, setvar=__SIP_CODEC=g729 |
| 18:19.11 | loather | yeah... that means I have to set up like 130 different peers. |
| 18:19.24 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:19.29 | p3nguin | Don't you need to have a peer for every device, anyway? |
| 18:19.31 | ManxPower-work | loather: yes. Do it the right way. |
| 18:19.34 | Corydon76-lap | loather: or set it in your dialplan |
| 18:19.37 | ManxPower-work | p3nguin: you do if you are sane. |
| 18:19.41 | loather | Corydon76-lap: that'll make it prefer g.729? |
| 18:19.48 | ManxPower-work | Corydon76-lap: the SIP_CODEC works for incoming calls? |
| 18:19.48 | Corydon76-lap | loather: or use templates |
| 18:20.31 | Corydon76-lap | ManxPower-work: Maybe. Depends upon when the channel is created, before or after codec negotiation |
| 18:21.25 | ManxPower-work | Corydon76-lap: As I understand it the codec is already set in stone by the time the call hits the dialplan. |
| 18:21.46 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 18:22.44 | Corydon76-lap | I'm not asking about dialplan. I'm talking about sip.conf |
| 18:23.06 | *** join/#asterisk darkskiez (n=mhb@cpc4-broo7-2-0-cust263.know.cable.virginmedia.com) |
| 18:23.21 | Corydon76-lap | and order of allows should still determine codec order |
| 18:23.48 | Corydon76-lap | Just remember that the codecs specified in the default context are first |
| 18:24.27 | loather | Corydon76-dig: That's what I thought! At least, that's exactly how it worked in 1.2. |
| 18:24.55 | Corydon76-lap | That code hasn't changed much |
| 18:24.57 | loather | If I wanted it to prefer 729, I'd list 729 first. GSM? GSM first. |
| 18:25.12 | Corydon76-lap | loather: but codecs in the default context come first |
| 18:25.14 | loather | Now it seems altogether nondeterministic between ulaw and 729. |
| 18:25.28 | loather | yah, the only place I have codecs listed is in the default context. |
| 18:25.30 | *** join/#asterisk tgunr (n=tgunr@c-68-37-9-16.hsd1.nj.comcast.net) |
| 18:25.42 | Corydon76-lap | loather: then that's the order used |
| 18:26.07 | loather | It never used to be a problem, because the 729-incapable phones would tell asterisk "FU, use something else" then later negotiate ulaw |
| 18:26.35 | loather | but now it picks one or the other without regard to the ordering |
| 18:27.10 | Corydon76-lap | I'd need to see a 'sip set debug on' with it picking the 'wrong' one to know anything more |
| 18:27.26 | loather | ok, hang tight,. |
| 18:29.11 | *** join/#asterisk gkeartland (n=gkeartla@188-220-19-194.zone11.bethere.co.uk) |
| 18:31.32 | *** join/#asterisk muiro (n=muiro@rrcs-24-56-88-130.ma.biz.rr.com) |
| 18:32.18 | loather | alright, looks like it happens on incoming calls only |
| 18:32.19 | muiro | cmd Originate isn't in 1.6.11 is it? |
| 18:32.36 | p3nguin | I think I would allow only g729 in default and then set ulaw in peers that don't support g729. |
| 18:32.39 | loather | that is, incoming calls to the PRI, which should be transcoded to 729 before being sent to the phone. |
| 18:33.52 | leifmadsen | muiro: 1.6.2.x |
| 18:34.09 | muiro | leifmadsen: many thanks |
| 18:34.21 | leifmadsen | muiro: at least I'm pretty sure that's where it shows up first |
| 18:35.40 | loather | looks like in the SDP for calls originating in ulaw from PRI, it's preferring *not* to transcode. It lists ulaw first, then g729. |
| 18:35.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:36.10 | p3nguin | Not transcoding is usually a good thing, so I can see why that would be preferred. |
| 18:36.48 | loather | Not in this case. Calls are eating an inordinate amount of bandwidth. |
| 18:37.01 | loather | The machine has plenty of CPU cycles to spare. |
| 18:40.42 | *** join/#asterisk gooph (n=gooph@pool-71-96-244-205.dfw.dsl-w.verizon.net) |
| 18:41.55 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-078-211.pools.arcor-ip.net) |
| 18:42.25 | ManxPower-work | I still don't see how you can have 130 endpoints without 130 entries in sip.conf |
| 18:43.06 | ManxPower-work | Unless you are doing something like...... |
| 18:43.08 | ManxPower-work | ~users.conf |
| 18:43.09 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 18:43.11 | Blackvel | hey guys, multi language voicemail unavail message works now :) |
| 18:43.59 | leifmadsen | Blackvel: yay! |
| 18:44.09 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:44.15 | leifmadsen | should really change that users.conf description to something slightly more sane and usable |
| 18:44.26 | Blackvel | but to be honest i used my own record application to record DE + EN :) |
| 18:44.45 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:45.44 | Blackvel | leifmadsen: absolutely fantastic. who drinks a champagne or beer with me? cheers! |
| 18:45.59 | leifmadsen | Blackvel: I would if it wasn't 2pm :) |
| 18:46.10 | leifmadsen | (that, and i had beer in the fridge) |
| 18:46.19 | ManxPower-work | leifmadsen: We don't worry about silly things like being factually correct around here. |
| 18:46.33 | high-freq | anyone here use the simple tool freepbx ;) |
| 18:47.08 | [TK]D-Fender | high-freq: Remember what I told you... NO SUPPORT HERE |
| 18:47.15 | [TK]D-Fender | ~freepbx |
| 18:47.15 | 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 |
| 18:47.17 | [TK]D-Fender | ^^^^^ |
| 18:47.20 | high-freq | damnit fender why u always gotta be on ;( |
| 18:47.31 | ManxPower-work | high-freq: most everyone else will tell you the same thing |
| 18:47.47 | high-freq | damn tried to squeeze a question in ;( lol |
| 18:47.51 | ManxPower-work | If support on #FreePBX is so terrible then why are you using? |
| 18:47.55 | high-freq | just testing u guys makin sure u guys are on your toes |
| 18:48.09 | ManxPower-work | high-freq: I'll test my kick/ban in a min. |
| 18:48.29 | high-freq | shit i haven't been to that chan lol. |
| 18:48.33 | high-freq | aaaw ;( |
| 18:48.46 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 18:49.55 | Blackvel | high-freq: if i where you i would try to knock the question down to native dialplan (no gui clicking). where is a 50:50 chance that one could help you here (but not for gui clicking) |
| 18:50.04 | nny | ok dumb question, but I had someone ask it and wanted to make sure I knew the right answer. If someone is forwarding calls from, say a cellphone, to asterisk, there is no info retained in the data about the original called number, right? I figured the CID name/num gets mangled, but there is no bit for "original number" |
| 18:50.06 | Blackvel | there is a... |
| 18:50.14 | high-freq | no worries i will ask my retarded question there ;) lol |
| 18:50.38 | ManxPower-work | ~guis |
| 18:50.39 | infobot | extra, extra, read all about it, guis is "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS" |
| 18:50.47 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 18:50.55 | high-freq | lmao nice one |
| 18:50.56 | bcrisp | ~wee |
| 18:50.56 | infobot | wee is, like, urine |
| 18:51.04 | nny | ~nny |
| 18:51.13 | nny | damn it got removed :\ |
| 18:51.20 | ManxPower-work | ~manxpower |
| 18:51.21 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
| 18:51.27 | high-freq | simply put i am just testing ;) but i got an ubuntu system with asterisk on it already, so all in all i got 2 systems just testing |
| 18:51.33 | bcrisp | ~bcrisp |
| 18:51.36 | high-freq | gui first tho click clickaroo |
| 18:51.42 | Blackvel | 2001? |
| 18:51.44 | Blackvel | wow |
| 18:51.46 | p3nguin | ~p3nguin |
| 18:51.58 | Blackvel | is it already that old |
| 18:51.59 | nny | tl:dr is there any CID info outside of name/num? |
| 18:52.00 | ManxPower-work | Blackvel: Oct 2001, IIRC. |
| 18:52.01 | nny | er |
| 18:52.04 | nny | tl: dr |
| 18:52.11 | bcrisp | ya when i google * questions i see really old chat logs with manxpower's name in em |
| 18:52.36 | p3nguin | nny: Yes, but it does not exist in all channel types nor all media types. |
| 18:52.42 | high-freq | damnit i gotta get off xchat i got a final in 10 mins ;( off to studyin guys bbl |
| 18:53.00 | bcrisp | good luck on your underwater basketweaving exam |
| 18:53.16 | nny | p3nguin: which media or channel types retain that data? |
| 18:53.31 | high-freq | wtf how'd u know? asjf;;kljf goddamnit stop talkin to me |
| 18:53.51 | Corydon76-lap | 2001 is about when I started working on Asterisk |
| 18:54.01 | bcrisp | i was in japan in 2001 |
| 18:54.37 | high-freq | oorah |
| 18:54.37 | nny | p3nguin: i assume this pertains to the question http://lists.digium.com/pipermail/asterisk-ss7/2009-July/002930.html |
| 18:54.40 | high-freq | haha |
| 18:54.51 | nny | p3nguin: which means "test the carrier first" |
| 18:55.24 | *** join/#asterisk clive- (i=ident@dsl-242-143-178.telkomadsl.co.za) |
| 18:56.48 | clive- | hi guys... I am having some strange issues with a tdm410p card which has a fxo module. I think my zaptel version is maybe too old. |
| 18:57.25 | clive- | does anyone know the best version of zaptel; to use with asterisk 1.2 |
| 18:57.37 | ManxPower-work | clive-: the latest 1.2.x zaptel |
| 18:58.01 | ManxPower-work | clive-: what problems? DTMF issues, one way audio, or ports locking up? |
| 18:58.44 | clive- | Manx, hi... the audio has clicks about a second appart, and the sound is shocking |
| 18:58.56 | Blackvel | leifmadsen: but it will be funny having a peer on 2pm...will make you work faster ;) |
| 18:59.28 | Blackvel | you US guys are always at least 6 hours behind Europe :) |
| 18:59.53 | dlynes | Blackvel: there's plenty of euros on here, too |
| 19:00.01 | clive- | Manx, version SVN-branch-1.2-r4587 of zaptel |
| 19:00.24 | Blackvel | btw....saw voicemail live on voip-info.org |
| 19:00.34 | Blackvel | has been built into asterisk in standard branch? |
| 19:00.39 | Blackvel | e.g 1.4/1.6? |
| 19:00.54 | dlynes | Blackvel: what is 'voicemail live'? |
| 19:01.01 | ManxPower-work | clive-: I meant the latest RELEASED zaptel |
| 19:01.05 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 19:01.22 | Blackvel | dlynes: lets you hear what the caller says and lets you pickup call |
| 19:01.35 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 19:01.37 | dlynes | Blackvel: oh...you mean spy on voicemail? |
| 19:01.43 | clive- | Manx, let me go figure out which version is the latest relaeased:) |
| 19:01.44 | *** join/#asterisk dekoi (n=droid@metroswitch.com) |
| 19:01.48 | Blackvel | read about it (1st day) |
| 19:01.55 | Blackvel | dlynes: yes |
| 19:02.08 | dlynes | Blackvel: you can probably do that using a standard dialplan |
| 19:02.21 | ayeso | I am NATing my call traffic, how can i change the source IP in the SIP packet to show my external NAT IP address? |
| 19:02.28 | Blackvel | if i hear the caller |
| 19:02.37 | Blackvel | and find a way to takeover chan, yes |
| 19:02.41 | danj1980 | externip? |
| 19:02.42 | ManxPower-work | ayeso: by following the NAT instructions |
| 19:02.45 | ManxPower-work | ~answers |
| 19:02.46 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 19:02.50 | dlynes | Blackvel: if you don't hear the call, how would you know someone's leaving a voicemal? |
| 19:03.01 | ayeso | ManxPower-work: hey thanks! |
| 19:03.04 | Blackvel | snoms read led is blinking |
| 19:03.15 | ayeso | ManxPower-work: Real problem solver you are! |
| 19:03.22 | Blackvel | sure...only when its finished |
| 19:03.23 | dlynes | Blackvel: that says that someone's already left one and hung up |
| 19:03.38 | dlynes | Blackvel: how are you going to interrupt a non-existent conversation? |
| 19:03.42 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 19:04.06 | clive- | how long does digium usually take to respond to an email support request? |
| 19:04.26 | dlynes | clive-: usually in less than 24 hours (assuming you don't email on the weekend) |
| 19:04.26 | dekoi | shouldn't take longer than 30 business days |
| 19:04.42 | clive- | lol |
| 19:04.53 | clive- | thanks dlynes |
| 19:05.31 | dlynes | clive-: or a holiday, for that matter |
| 19:05.32 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:05.42 | *** join/#asterisk CrazyTux (n=brandon@70-138-119-70.lightspeed.hstntx.sbcglobal.net) |
| 19:05.51 | dlynes | clive-: make sure you're not using any non-digium hardware, too |
| 19:05.57 | dlynes | clive-: otherwise they'll refuse to help you |
| 19:06.15 | clive- | dlynes, no its digium card messing up all on its own |
| 19:06.28 | loather | ManxPower-work: I have over 400 entries in sip.conf. I just don't want to go through and manually update all of them with that codec line that may or may not solve the problem. |
| 19:06.44 | dlynes | loather: you're not using templates? |
| 19:07.00 | loather | apparently not :P |
| 19:07.02 | ManxPower-work | loather: easy search and replace. |
| 19:07.08 | leifmadsen | loather: learn how to use templates |
| 19:07.13 | p3nguin | If you only allow one default codec and you only allow one peer codec, how can that not solve the problem? |
| 19:07.13 | loather | hard to S&R on something that's not there. |
| 19:07.13 | bcrisp | i love templates |
| 19:07.31 | ManxPower-work | p3nguin: oh it will solve the problem. He just doesn't want to do it that way |
| 19:07.51 | dlynes | loather: your life would be 1000x simpler using templates...then your codes would only be defined in the template, and all your sip peers/users/friends would inherit that template |
| 19:08.00 | dlynes | s/codes/codecs/ |
| 19:08.09 | loather | dlynes: that may be something I need to do. |
| 19:08.18 | p3nguin | ~templates |
| 19:08.31 | dlynes | loather: they're covered in about 2 or 3 pages in 'thebook' |
| 19:08.33 | dlynes | ~thebook |
| 19:08.34 | infobot | i guess thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 19:09.44 | dlynes | loather: between templates, macros and astdb, you can trim down the size of the average sip/iax file and dialplan astronomically |
| 19:10.03 | dlynes | loather: not to mention drastically simplifying them |
| 19:10.33 | bcrisp | is there any special way to set up voicemail for queues? or would i just need to create a vm box and give access to all members? |
| 19:11.05 | p3nguin | But you're going to spend more time converting over to a template than you will if you just add disallow=all and allow=ulaw in each peer definition. |
| 19:11.31 | bcrisp | id like to have a group voicemail box |
| 19:11.58 | p3nguin | bcrisp: Set a timeout on the Queue() command, and then follow it with a VoiceMail() command for a "general" mailbox where everyone can check it. |
| 19:11.58 | dlynes | p3nguin: yes, but the search and replace suggestion from ManxPower-work is the proper fix for now....however, loather should switch over to templates gradually, when he can, to make life easier for himself |
| 19:12.06 | loather | dlynes: oh, i make extensive use of macros :) |
| 19:12.53 | loather | macros, subroutines, variable substitution ... when i inherited this system the config files were around 4k lines long. Now I've got them down to about a quarter of that. |
| 19:12.58 | dlynes | bcrisp: another thing you can do, is set up a general email address that they can all check via imap, and have that voicemail deliver the voicemails via a wav file attachment to that general email address |
| 19:13.09 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:13.13 | ManxPower-work | dlynes: setting a default codec of g729 with codec specified for the few peers that require ulaw is what I would do. |
| 19:13.14 | loather | Including a hell of a lot more functionality. |
| 19:13.26 | hardwire | http://hphotos-snc3.fbcdn.net/hs108.snc3/15561_1215561306608_1156089753_30671293_302835_n.jpg <- Devil kitty commands you all to step away from the computer and pet something fluffy |
| 19:13.27 | bcrisp | p3nguin, very good. Am using the queues.conf context to allow callers to dial 1 to leave a voicemail |
| 19:13.47 | bcrisp | ya in voicemail.conf im setting up a mailing list address for the particular vm box |
| 19:14.18 | p3nguin | bcrisp: You could also combine what I suggested with what you are now doing. That would allow them to press the button or to wait long enough and still get that voicemail box. |
| 19:14.21 | *** join/#asterisk jjg_ (n=jjg@12.40.200.74) |
| 19:14.41 | dlynes | bcrisp: or better yet, have that email automatically processed by a trouble ticket system, have it entered into the trouble ticket system, so you can audit whether the dummies on the phones actually do anything with the message. |
| 19:14.49 | bcrisp | p3nguin: i'll probably use it in combination with some gotoiftime for hours of operation |
| 19:15.10 | bcrisp | dlynes: good idea |
| 19:15.16 | dlynes | bcrisp: just imap email or just voicemail messages on the phone are not auditable |
| 19:15.23 | jjg_ | hi, we are currently using voxeo for vxml/ccxml, but i've been tasked with finding a scalable replacement ... any recommendations for something that we can host in-house that will act as a media gateway? |
| 19:15.36 | p3nguin | bcrisp: I have my Queue() timeout set to 300, so that if I or one of my agents cannot get to a call in five minutes, they don't need to keep sitting there thinking someone will eventually answer. |
| 19:15.54 | bcrisp | p3nguin: but they are first in line! haha |
| 19:16.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.38 | p3nguin | hehe yeah |
| 19:16.48 | dlynes | bcrisp: there's also a third-party add-on allowing you to have the queue call them back when it's their turn in the queue |
| 19:17.04 | drfreeze | Is this a known problem: while my desk phone was ringing, the party I was talking to from my desk phone could not hear me |
| 19:17.18 | bcrisp | dlynes: ive been looking for that... whats it called? |
| 19:17.20 | dlynes | drfreeze: that's a phone problem |
| 19:17.24 | p3nguin | Sounds like a call waiting problem on the phone. |
| 19:17.29 | drfreeze | or |
| 19:17.56 | drfreeze | Is this a known problem: while my desk phone was ringing, I could not hear the party I was talking to from my desk phone |
| 19:17.59 | dlynes | bcrisp: nvm...there's a freely available version, too |
| 19:18.00 | p3nguin | Is the phone actually ringing while you are on the phone, or do you just get the call waiting tone? |
| 19:18.18 | drfreeze | p3nguin: let me ask |
| 19:18.53 | *** join/#asterisk pyite_hsv (n=pyite_hs@unaffiliated/pyite) |
| 19:18.53 | dlynes | bcrisp: free: http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback non-free, supported: http://www.orderlyq.com/asteriskqueues.html |
| 19:19.07 | bcrisp | dlynes, ty |
| 19:19.18 | CrazyTux | exten => _pstntrunkX.,1,NoOp(This is a test) <--- why is "pstntrunk33158366700" not matching? |
| 19:19.45 | CrazyTux | Gives me an extension not found etc |
| 19:20.04 | dlynes | CrazyTux: probably because 'n' matches [2-9]? |
| 19:20.14 | CrazyTux | dlynes: thats an X ? |
| 19:20.27 | p3nguin | What's an X? |
| 19:20.35 | p3nguin | X matches 0-9. |
| 19:20.36 | dlynes | CrazyTux: X is 0-9, N is 2-9, and Z is 1-9 |
| 19:20.49 | CrazyTux | dlynes: ok so how is that a problem/ |
| 19:21.00 | dlynes | CrazyTux: never checked to see if lower case also matches |
| 19:21.14 | p3nguin | <PROTECTED> |
| 19:21.31 | CrazyTux | p3nguin: _pstndrunk-X. did not work either? |
| 19:21.45 | p3nguin | It's nonsense. Use something sane. |
| 19:21.56 | p3nguin | Like your phone number. |
| 19:21.56 | CrazyTux | p3nguin: what is wrong with that/ |
| 19:22.03 | dlynes | crazytux: try _psttrukX., instead |
| 19:22.11 | CrazyTux | p3nguin: that did not work either |
| 19:22.13 | dlynes | p3nguin: there's nothing wrong with it, per se |
| 19:22.16 | p3nguin | Then use 's' |
| 19:22.19 | dlynes | p3nguin: you can have letters in extensions |
| 19:23.31 | CrazyTux | I think I may see the problem |
| 19:23.43 | dlynes | CrazyTux: can you pastebin your entire dialplan? |
| 19:23.48 | CrazyTux | dlynes: I can not |
| 19:23.54 | dlynes | CrazyTux: why not? |
| 19:24.01 | Katty | peeks in |
| 19:24.03 | Katty | :> |
| 19:24.07 | Katty | hi! :> |
| 19:24.10 | CrazyTux | Call from 'XX.XX.XX.XX' to extension 'pstntrunk-33158366700' rejected because extension not found. <-- why would it not show the context that the sip.conf is routing to? |
| 19:24.10 | dlynes | peers at Katty. |
| 19:24.13 | Katty | distributes muffins |
| 19:24.18 | CrazyTux | instead it is showing the IP |
| 19:24.30 | CrazyTux | Usually it shows the context |
| 19:24.39 | [TK]D-Fender | CrazyTux: because it doesn't. Now pay attention to SIP DEBUG |
| 19:24.45 | [TK]D-Fender | CrazyTux: never has |
| 19:25.02 | chuckf | mmmm.....muffins |
| 19:25.34 | CrazyTux | [TK]D-Fender: ah |
| 19:25.48 | CrazyTux | [TK]D-Fender: I am looking at the sip call with ngrep, I see it coming through as it should |
| 19:26.07 | dlynes | CrazyTux: Try also as the first line in your dialplan context 'exten => _.,1,Noop(The exten dialed was: ${EXTEN})' |
| 19:26.17 | dlynes | CrazyTux: it's just temporary though, so you can see what's coming in |
| 19:26.17 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 19:26.44 | dlynes | CrazyTux: do not make that a permanent entry, or get in the habit of using '_.', because it's a big enough pattern match to clobber you |
| 19:26.46 | [TK]D-Fender | CrazyTux: Do you think an ASTERISK DIALPLAN CONTEXT will show up in SIP PACKETS? No, you'll see this in * CLI <- |
| 19:27.24 | CrazyTux | alright let me take a look |
| 19:27.33 | dlynes | CrazyTux: now what does the 'noop' output for you? |
| 19:27.36 | p3nguin | Yay! Compliance! |
| 19:27.58 | CrazyTux | :) |
| 19:28.00 | Katty | oh dear :< |
| 19:28.03 | dlynes | CrazyTux: if it didn't output anything, it's because you've got something fubar in your sip.conf file, or because you put it in the wrong context |
| 19:28.06 | Katty | it's only noon and all the feeders appear EMPTY |
| 19:28.20 | Katty | how is this possibles?! i filled them up at 7 |
| 19:28.31 | p3nguin | badgers |
| 19:28.32 | dlynes | Katty: they're storing everything for the winter? |
| 19:28.44 | bcrisp | thinks someone is stealing katty's critter food and selling it |
| 19:28.46 | Katty | yes, that is possible i suppose |
| 19:28.51 | Katty | bcrisp: it's not worth it |
| 19:28.56 | Katty | bcrisp: i will calculate price per pound |
| 19:29.05 | jblack | thinks someone is stealing katty's critter food and eating it |
| 19:29.14 | dlynes | thinks maybe it's jblack. |
| 19:29.21 | jblack | people are critters too |
| 19:29.22 | *** join/#asterisk cusco_ (n=trilili@2001:0:53aa:64c:1c6f:5a82:a077:acde) |
| 19:29.22 | CrazyTux | [TK]D-Fender: Looking for pstntrunk-33158366700 in incoming-route |
| 19:29.30 | [TK]D-Fender | CrazyTux: \o/ |
| 19:29.38 | p3nguin | If it's anything like my area, people steal shit just for the sake of stealing it. It doesn't even need to have value. |
| 19:29.40 | cusco_ | Dec 14 19:27:51] NOTICE[25480]: rtp.c:1796 ast_rtp_read: Unknown RTP codec 126 received from '84.90.32.13' |
| 19:30.03 | bcrisp | someone stole my lawn lights |
| 19:30.10 | CrazyTux | [TK]D-Fender: ok which in that context I have: exten => _pstntrunk-X.,1,Answer |
| 19:30.11 | bcrisp | (only 3 of em) |
| 19:30.15 | jblack | I keep meaning to steal the baby jesus out of the nativity scene at the courthouse |
| 19:30.27 | cusco_ | according to http://www.asteriskguru.com/tutorials/unknown_codec_received.html - the devices decide to send dtmf as dynamic |
| 19:30.30 | bcrisp | jblack = felony |
| 19:30.37 | CrazyTux | lol |
| 19:30.41 | p3nguin | not if it's under $300 value |
| 19:30.51 | Katty | it costs me .3364 per pound of squirrel food |
| 19:30.52 | bcrisp | its a federal courthouse? |
| 19:30.53 | jblack | no, jblack = seperating church from state |
| 19:30.54 | dlynes | baby jesus is priceless |
| 19:30.59 | Katty | err .3664 |
| 19:31.02 | dlynes | so it would be over $300 value :0 |
| 19:31.03 | Katty | so like... 37 cents a pound |
| 19:31.06 | [TK]D-Fender | CrazyTux: "n" = reserved letter |
| 19:31.17 | Katty | bcrisp: actually you probably oculd make quite a profit on that |
| 19:31.20 | [TK]D-Fender | CrazyTux: And you aren't showing your actual whole context |
| 19:31.20 | CrazyTux | [TK]D-Fender: ahhh ok |
| 19:31.22 | p3nguin | [tk]d-fender: Good luck getting that across. |
| 19:31.25 | drfreeze | p3nguin: here's more info: 230 calls 221. Outside call comes in and dials both 230 and 221 |
| 19:31.34 | dlynes | [TK]D-Fender: i've already told him that...but he doesn't listen so good :) |
| 19:31.38 | drfreeze | They both here the call waiting tone, but then both lose audio |
| 19:31.41 | Katty | bcrisp: several stores sell 8lb bags of squirrel / critter feed for at least 5 bucks |
| 19:31.59 | dlynes | [TK]D-Fender: besides...his dialplan is a national secret |
| 19:32.02 | jblack | I don't understand feeding wild animals, when so many people are going hungry these days. |
| 19:32.03 | drfreeze | When the outside call is hung up, audio returns in about a second |
| 19:32.16 | ManxPower-work | jblack: feed the wild animals to the hungry |
| 19:32.20 | Katty | bcrisp: 7lbs critter mix feed at lowes is 5.78 |
| 19:32.32 | jblack | ManxPower-work: I'd be ok with that. |
| 19:32.48 | bcrisp | nice |
| 19:32.50 | beek | jblack: because I like watching wild animals outside my window. Starving people, not so much. |
| 19:33.07 | bcrisp | CLI went nuts just now |
| 19:33.13 | jblack | You're a traitor to your species, beek |
| 19:33.15 | *** join/#asterisk Tim_Toady (n=moi@188.4.89.184.dsl.dyn.forthnet.gr) |
| 19:33.17 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 19:33.40 | bcrisp | beek... wild people? |
| 19:33.43 | p3nguin | bcrisp: core set verbose 0 |
| 19:33.53 | bcrisp | p3nguin.. it was an infinite loop haha |
| 19:33.59 | beek | bcrisp: Girls gone wild? |
| 19:34.13 | dlynes | beek: you've been watching too much late night tv |
| 19:34.23 | beek | hangs head in shame. |
| 19:34.35 | dlynes | beek: you probably own about 10 foreclosures in vegas, i bet too, eh? |
| 19:34.48 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 19:34.56 | dlynes | beek: and buy houses in detroit and cleveland on ebay? |
| 19:35.08 | bcrisp | my IVR is ghetto |
| 19:35.11 | bcrisp | 3 different voices |
| 19:35.12 | beek | dlynes: Why would I want houses in detroit? I'd never be able to sell them. |
| 19:35.23 | p3nguin | bcrisp: lol |
| 19:35.26 | Katty | if there were starving people in my yard...i'm not sure what i'd do to be honest |
| 19:35.28 | bcrisp | beek, they would make good firewood |
| 19:35.39 | p3nguin | katty: put out more feed. |
| 19:35.40 | Katty | i would feel inclined to help them... |
| 19:35.48 | Katty | but that one episode of south park keeps coming back |
| 19:35.54 | bcrisp | "Support mental illness, or we'll stand in your yard" |
| 19:37.00 | bcrisp | was actually at a volunteer mental illness center and one of them had that idea to raise money :) |
| 19:37.04 | Katty | i'm not really sure how to help them |
| 19:37.30 | Katty | especially when so many of them don't really want money for food |
| 19:37.56 | Katty | and even if they did, i don't make enough money to feed a whole platoon of them |
| 19:38.06 | bcrisp | (they want $$ for WOW) |
| 19:38.32 | CrazyTux | [TK]D-Fender: that was the issue, thank you! |
| 19:38.50 | dlynes | CrazyTux: so when I told you that, why the hell didn't you listen to me? |
| 19:38.56 | cusco_ | can I just change the devstate for device? |
| 19:39.03 | CrazyTux | dlynes: you never said n was reserved |
| 19:39.04 | cusco_ | from In use to NOT_Inuse |
| 19:39.05 | cusco_ | ' |
| 19:39.10 | dlynes | CrazyTux: I DID SO |
| 19:39.12 | p3nguin | dlynes: Your nick is shorter than [tk]d-fender's. |
| 19:39.19 | dlynes | CrazyTux: N = 2-9 |
| 19:39.30 | dlynes | slaps CrazyTux. |
| 19:39.33 | CrazyTux | N not n |
| 19:39.45 | CrazyTux | but apparently so? |
| 19:39.47 | dlynes | yes |
| 19:39.48 | CrazyTux | lowercase is the same |
| 19:39.51 | beek | p3nguin: You did say 'nick', right? |
| 19:40.01 | p3nguin | uh yes |
| 19:40.35 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 19:40.36 | p3nguin | tries to remain without vulgarity as often as possible. |
| 19:40.56 | dlynes | p3nguin: He's thinking about his own little problems....that's why he spends so much time watching late night tv...waiting for the viagra ads interspersed within the girls gone wild ads |
| 19:41.45 | p3nguin | I've noticed they play and Extenze infomercial and then a GGW infomercial after it. |
| 19:42.02 | dlynes | coincidence? |
| 19:42.09 | p3nguin | I think not. |
| 19:42.45 | p3nguin | Man, that would have been so bad if I would have typoed that statement earlier. |
| 19:43.04 | drfreeze | p3nguin: Just verified that the same problem can be recreated with all sip calls |
| 19:43.19 | p3nguin | drfreeze: What kind of phones are you using? |
| 19:43.33 | drfreeze | 230 calls 201, then another sip ext calls 888 which dials 230 and 221 |
| 19:43.35 | drfreeze | Polycom |
| 19:43.57 | drfreeze | All of these happen to be 550's with 3.2.3 cfg files |
| 19:44.32 | drfreeze | they lose audio until the call is answered or hangs up |
| 19:45.01 | p3nguin | Firmware bug, maybe? Did you try earlier and later firmware versions? |
| 19:45.41 | ManxPower-work | you don't GET audio until the call is answered. |
| 19:45.51 | ManxPower-work | Do you mean "ringback"? |
| 19:46.21 | p3nguin | manxpower-work: There is an existing call in progress. When call waiting tone sounds, it seems to kill the audio on the current call. |
| 19:46.44 | ManxPower-work | p3nguin: Sounds like an episode of the twilight zone. |
| 19:47.44 | CrazyTux | dlynes: p3nguin thank you guys too, sorry Im just having a bad morning -- not feeling so well |
| 19:48.09 | CrazyTux | and yes the dial plan is a national secret :) |
| 19:48.10 | CrazyTux | haha |
| 19:48.14 | p3nguin | crazytux: The important part is that you learned something AND you got the problem resolved. |
| 19:48.32 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.163.240.dsl.dyn.forthnet.gr) |
| 19:52.29 | bcrisp | i love gotoiftime |
| 19:53.05 | p3nguin | It works nicely for setting up an after-hours dialplan. |
| 19:53.31 | drfreeze | p3nguin: where do I get firmware files? |
| 19:53.43 | drfreeze | I have tried 4 different config files |
| 19:53.44 | p3nguin | No idea; I don't use Polycom phones. |
| 19:53.51 | drfreeze | 3.2.3, 3.1.3 |
| 19:53.55 | drfreeze | and a few other |
| 19:54.19 | *** join/#asterisk high-freq (n=hfreq@99.188.101.168) |
| 19:54.39 | *** join/#asterisk korihor (n=korihor@190.205.251.97) |
| 19:54.57 | [TK]D-Fender | drfreeze: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 19:55.21 | drfreeze | [TK]D-Fender: that's the config files. Is firmware included? |
| 19:55.34 | [TK]D-Fender | drfreeze: Yes. |
| 19:56.54 | drfreeze | I have tried 3.1.3RevC, 3.2.1, 3.2.2, 3.2.1RevB |
| 19:57.39 | Katty | how sad |
| 19:57.42 | Katty | my front yard is empty )= |
| 19:58.05 | drfreeze | The 3.1.3RevC works with the 301, 501 and 550 which is what this office has |
| 19:58.08 | Katty | oh wait! there's 2 on the feeders!~ |
| 19:59.39 | drfreeze | Anyone have an idea(s) on what to be looking at for this problem |
| 20:00.00 | drfreeze | I'm at a loss at this point |
| 20:00.27 | *** join/#asterisk Alagar (n=Administ@122.164.35.197) |
| 20:00.36 | drfreeze | The only think I have to try is completely rebuild the phone server and try again assuming the problem is not with the phones |
| 20:01.47 | Katty | drfreeze: do what? |
| 20:01.49 | Katty | scrolls up |
| 20:03.02 | Katty | drfreeze: what happens if you take one of your phones off the ftp/tftp server? |
| 20:03.07 | Katty | drfreeze: and then format it, and reprogram it |
| 20:03.09 | drfreeze | Katty: twilight zone like problem. Incoming call knocks out audio on existing calls |
| 20:03.20 | Katty | drfreeze: does it still snickerdoodle up when it's not pointing to a server? |
| 20:03.40 | Katty | oh i see. |
| 20:03.47 | Katty | i wouldn't think that would have anything to do with your phones |
| 20:04.01 | Katty | or at least the firmware. |
| 20:04.15 | drfreeze | we've ruled out PRI card |
| 20:04.22 | Katty | but i can't say i've ever had that issue |
| 20:04.22 | *** join/#asterisk bprice20 (n=bprice20@64.1.25.210.ptr.us.xo.net) |
| 20:04.26 | drfreeze | happens when only sip phones are involved |
| 20:04.45 | Katty | working under that assumption |
| 20:04.49 | Katty | what does a sip software phoen do |
| 20:04.54 | drfreeze | It's like basic *. 3 line extension. Answer, Dial, Hangup |
| 20:04.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:05.11 | Katty | a software phone will be completely seperate from your polycom setup |
| 20:05.14 | Katty | might tell you something |
| 20:05.30 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 20:05.31 | drfreeze | have a reccomendation for one to use with winders? |
| 20:05.36 | Katty | zoiper |
| 20:05.44 | eppigy | ZOINK |
| 20:05.55 | Katty | hi dave |
| 20:06.20 | drfreeze | Katty: cool, I'll give it a try |
| 20:10.43 | bcrisp | ~woot |
| 20:10.43 | infobot | \o/ |
| 20:13.46 | eppigy | Katty: hi der |
| 20:21.24 | drfreeze | Katty: how do I configure this zoiper softphone? |
| 20:21.31 | bcrisp | right click it |
| 20:21.46 | drfreeze | I created a sip account. What do I use for the domain? |
| 20:21.57 | bcrisp | the ip address/ domain name of your * server |
| 20:23.00 | drfreeze | bcrisp: ok, did that |
| 20:23.12 | bcrisp | k |
| 20:23.17 | bcrisp | is it registering? |
| 20:23.23 | drfreeze | still says registering sip |
| 20:23.32 | bcrisp | do you have CLI open? |
| 20:23.59 | drfreeze | what do you mean by CLI? |
| 20:24.12 | bcrisp | command line interface |
| 20:24.20 | drfreeze | I have that open for * |
| 20:24.30 | bcrisp | ok pastebin what is happening |
| 20:25.38 | drfreeze | 288 (Unspecified) D 0 UNKNOWN |
| 20:26.08 | bcrisp | ? |
| 20:26.12 | bcrisp | what is that |
| 20:26.18 | drfreeze | that is sip show peers |
| 20:26.43 | drfreeze | ok, I found the advanced options |
| 20:26.48 | drfreeze | maybe this will help |
| 20:26.51 | bcrisp | hm |
| 20:26.58 | bcrisp | ya make sure you have the right codecs specified |
| 20:27.11 | bcrisp | you shouldnt have to do anything advanced tho |
| 20:28.19 | drfreeze | so, ulaw |
| 20:28.24 | drfreeze | or alaw |
| 20:28.35 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 20:28.47 | bcrisp | pastebin your sip.conf |
| 20:29.51 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 20:31.34 | drfreeze | bcrisp: http://pastie.textmate.org/private/9fiarf7qonflxuwp3gd2dg |
| 20:32.15 | bcrisp | drfreeze: ok good, can you pastebin your sip.conf file now? |
| 20:32.25 | drfreeze | sure |
| 20:33.35 | drfreeze | bcrisp: http://pastie.textmate.org/private/rfddduduvnbxmhggjdooza |
| 20:34.15 | bcrisp | ok now pastebin your extensions.conf |
| 20:35.39 | drfreeze | bcrisp: extensions.conf is empty for now |
| 20:35.58 | bcrisp | ok.. |
| 20:36.14 | bcrisp | change nat=yes |
| 20:36.21 | bcrisp | in sip.conf for 288 |
| 20:36.30 | drfreeze | k |
| 20:36.41 | bcrisp | then from CLI do sip reload |
| 20:36.58 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 20:37.10 | drfreeze | there is no nat'ing tho, all on same switch (or is this std with soft phones) |
| 20:37.30 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 20:38.00 | bcrisp | thats been the case with mine |
| 20:38.22 | *** join/#asterisk kephu (n=Keph@chello089075152208.chello.pl) |
| 20:38.25 | kephu | hi |
| 20:39.08 | beek | hi |
| 20:39.08 | kephu | I've just installed asteriskNOW and I'm having a bit of a trouble getting around in it. Can anyone here show me the ropes or point to a good 'getting started' guide? |
| 20:39.18 | [TK]D-Fender | kephu: ... |
| 20:39.20 | [TK]D-Fender | ~freepbx |
| 20:39.21 | 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 |
| 20:39.22 | [TK]D-Fender | ^^^^^^^^^^^ |
| 20:39.31 | drfreeze | still getting the no matching peer found |
| 20:39.32 | drfreeze | Dec 14 14:39:05] NOTICE[2858]: chan_sip.c:15933 handle_request_register: Registration from '"288"<sip:PlcmSpIp@192.168.121.50;transport=UDP>' failed for '192.168.121.178' - No matching peer found |
| 20:39.48 | kephu | ...freepbx? |
| 20:39.57 | bcrisp | grrrrrrrrrr |
| 20:40.35 | ManxPower-work | drfreeze: do you have a [288] section of sip.conf? |
| 20:40.40 | [TK]D-Fender | kephu: If its up and running and you need help configuring it, then #freepbx . If you have problems getting it installed and started in basic form : #asterisknow |
| 20:41.09 | kephu | I guess it installed, technically |
| 20:41.16 | drfreeze | ManxPower-work: yes |
| 20:41.33 | drfreeze | ManxPower-work: http://pastie.textmate.org/private/rfddduduvnbxmhggjdooza |
| 20:42.13 | ManxPower-work | drfreeze: try changing [288] to [PlcmSpIp] and do a sip reload, see what happens |
| 20:42.23 | drfreeze | k |
| 20:42.33 | bcrisp | thats his secret? |
| 20:42.55 | ManxPower-work | drfreeze: if that works then fix your phone config |
| 20:43.17 | ManxPower-work | bcrisp: Looks like it. Don't worry, he'll get hacked pretty fast with a secret like that |
| 20:43.18 | drfreeze | ManxPower-work: that worked |
| 20:43.26 | ManxPower-work | drfreeze: fix your phone config |
| 20:43.45 | ManxPower-work | drfreeze: are you using polycoms? |
| 20:43.48 | bcrisp | manxpower, is he entering his password instead of authorized user name? |
| 20:44.04 | ManxPower-work | bcrisp: no idea. |
| 20:44.22 | bcrisp | by changing the sip device name to his secret i would imagine so |
| 20:44.38 | ManxPower-work | bcrisp: all I KNOW is what he pastebin'd |
| 20:45.00 | bcrisp | off to see the wizard |
| 20:45.05 | *** join/#asterisk pagec (n=chatzill@ip67-152-24-98.z24-152-67.customer.algx.net) |
| 20:45.08 | ManxPower-work | http://www.fnords.org/~eric/polycom-config-examples/ for any that are interested. |
| 20:45.11 | cusco_ | I still can't figure why the device state is "In use" |
| 20:45.20 | ManxPower-work | These config files use the MAC of the phone as the userid, BTW. |
| 20:45.25 | cusco_ | We restarted asterisk, changed teh extension configuratione tc |
| 20:45.53 | pagec | i have a PRI card from digium with echo cancelation hardware, is it possible to make adjustments to the echo cancel on their hardware? |
| 20:45.55 | *** join/#asterisk TimToady_ (n=moi@188.4.79.86.dsl.dyn.forthnet.gr) |
| 20:46.01 | ManxPower-work | pagec: no. |
| 20:46.03 | *** part/#asterisk kephu (n=Keph@chello089075152208.chello.pl) |
| 20:46.24 | pagec | so if people are still hearing echo there isn't a lot that can be done about it? |
| 20:46.45 | ManxPower-work | pagec: A1: No. A2: Contact Digium. This issue is something they will help with. |
| 20:46.46 | drfreeze | ManxPower-work: yes |
| 20:46.58 | ManxPower-work | drfreeze: look at my config examples URL: |
| 20:47.38 | cusco_ | can I manually change the device state? |
| 20:47.55 | ManxPower-work | cusco_: perhaps you will have better luck on the mailing lists? |
| 20:48.12 | cusco_ | well I guess so.. |
| 20:50.03 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-155-79.dsl.stlsmo.sbcglobal.net) |
| 20:50.15 | cusco_ | ManxPower-work: that is the devstate right? |
| 20:51.06 | LemensTS | hey all. Does anyone have a script that runs on cron that checks for too many 'sip show channels' open and does something if so? Or a script somewhat close that I can modify? |
| 20:51.20 | voipmonk | err |
| 20:51.31 | voipmonk | too any sip show channels? - what are you doing, LemensTS ? |
| 20:53.23 | LemensTS | voipmonk: http://forum.asterisk2billing.org/viewtopic.php?f=21&t=6765&p=26525#p26525 There is the problem. As far as I can see its an A2Billing issue. Until I can get it fixed I was going to run a script that would restart asterisk if they sip channels built up real high (looks like REGISTERS mainly) |
| 20:53.56 | bcrisp | hmm |
| 20:54.10 | voipmonk | might need to redesign the query |
| 20:54.20 | voipmonk | to use MySQL instead of REALTIME |
| 20:54.23 | drfreeze | bcrisp: speaking of secrets, do you manually enter a new secret in each phone? |
| 20:54.39 | drfreeze | Katty: Ok, the soft phone appears to be working |
| 20:54.43 | drfreeze | no audio problems |
| 20:55.00 | Katty | k |
| 20:55.20 | bcrisp | drfreeze: if you want |
| 20:55.26 | drfreeze | So, I noticed on the last polycom reboot that it did not update sip cfg. It is still at 3.2.3. I want to have it load 3.1.3RevC |
| 20:55.40 | drfreeze | Do I need to force it some how to downgrade? |
| 20:56.03 | bcrisp | drfreeze: i set dif pwds for sip phones |
| 20:56.30 | drfreeze | oh, ok. I just set this up as a test. |
| 20:56.42 | *** part/#asterisk CrazyTux (n=brandon@70-138-119-70.lightspeed.hstntx.sbcglobal.net) |
| 20:57.37 | *** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com) |
| 20:57.46 | Katty | is cranky, coworkers are acting like douchebags. |
| 20:57.55 | bcrisp | drfreeze: in your sip phone (zoiper right?) - did you set all the user name type fields to 288? |
| 20:58.14 | ManxPower-work | drfreeze: sip.cfg comes with your polycom firmware |
| 20:58.44 | drfreeze | not sure why I get this message: Application sip.ld is not is incompatible with the phone |
| 20:59.18 | drfreeze | bcrisp: set username to 288 and caller id to the ftp username |
| 21:00.28 | ManxPower-work | drfreeze: that's EXACTLY what it says. |
| 21:00.39 | ManxPower-work | Sounds like you have 300/301/500/501/600/601 phones. |
| 21:01.05 | ManxPower-work | But you already know about these issues since you carefully read the release notes everytime you upgrade, right?? |
| 21:02.48 | drfreeze | I have 301, 501 and 550 phones |
| 21:03.05 | drfreeze | the 3.1.3RevC file is supposed to work with all those phones |
| 21:05.32 | *** join/#asterisk rethus1 (n=contio@p5087398C.dip.t-dialin.net) |
| 21:05.41 | rethus1 | hi there, |
| 21:05.49 | rethus1 | have problems with 1.6 and musiconhold |
| 21:06.03 | rethus1 | got the error: Music class requested but no musiconhold loaded |
| 21:06.16 | rethus1 | seems no class loaded. can anybody help? |
| 21:06.22 | rethus1 | rund asterisk on ubuntu 9.10 |
| 21:06.50 | ChannelZ | rethus1: module load res_musiconhold |
| 21:07.42 | ManxPower-work | drfreeze: what BootROM version are you using? |
| 21:08.08 | ManxPower-work | drfreeze: are you using the SPLIT or the COMBINED firmware? |
| 21:11.21 | jjg_ | anyone maintaining their own media gateway for vxml/ccxml? |
| 21:14.11 | *** join/#asterisk ivan_paes (n=paes@201-67-161-42.fnsce703.dsl.brasiltelecom.net.br) |
| 21:14.15 | clive- | are "IRQ misses " common when starting up ? |
| 21:16.22 | Naikrovek | that should be in the channel topic. polycom phones should always use split bootrom and SIP firmwares |
| 21:16.48 | ManxPower-work | Naikrovek: good thing drfreeze isn't watching the channel. |
| 21:16.50 | drfreeze | ManxPower-work: I have the following bootroms |
| 21:16.50 | drfreeze | ["4.1.1.0232", "3.2.3.0002", "3.2.2.0019", "4.1.3.0052"] |
| 21:17.04 | drfreeze | Am using split firmware |
| 21:17.10 | Naikrovek | good man |
| 21:17.13 | drfreeze | I man combined |
| 21:17.19 | drfreeze | *mean |
| 21:17.21 | bcrisp | you man combined? |
| 21:17.22 | bcrisp | :D |
| 21:17.26 | bcrisp | eewwww |
| 21:17.30 | drfreeze | which, after reading the docs, it looks like I should change that |
| 21:17.33 | ManxPower-work | drfreeze: I'd comment, but I closed all my polycom browser windows. You want to use SPLIT with Bootrom 4.0 or higher. |
| 21:17.52 | Naikrovek | ew combined = frought with trouble |
| 21:18.01 | Naikrovek | do the split, even though it's a bigger download |
| 21:18.07 | Naikrovek | it will just work (tm) |
| 21:18.47 | bcrisp | queue members are about to get a rude awekening... |
| 21:19.02 | bcrisp | im gonna spawn off some calls to test their answer times |
| 21:19.21 | *** join/#asterisk hardwire (n=spencers@39.183.dowl.anc.borealisbroadband.net) |
| 21:19.49 | Naikrovek | audit |
| 21:19.52 | bcrisp | brb coffee .. want to be awake for this :) |
| 21:20.28 | *** join/#asterisk joobie (n=joobz@mx01.anric.com.au) |
| 21:20.33 | bcrisp | nah not for audit really.. just want to see how they react :) - they dont know they are queue members yet :D |
| 21:21.34 | bcrisp | speak of the devil |
| 21:22.45 | bcrisp | one of you just do that? |
| 21:23.00 | Naikrovek | just do what |
| 21:23.13 | bcrisp | right after i mentioned the queue thingie got a call on our unlisted # |
| 21:23.15 | *** join/#asterisk voipmonk (n=voipmonk@CPE001c10479603-CM0014f8cd6db2.cpe.net.cable.rogers.com) |
| 21:23.23 | bcrisp | (unlisted outside of my pastebins here heh) |
| 21:23.26 | Naikrovek | ah |
| 21:23.27 | Naikrovek | heh |
| 21:23.31 | Naikrovek | hope your system is secure |
| 21:24.03 | Naikrovek | there are pranksters in here who use your hostname/ip to give you something to do/worry about |
| 21:24.08 | *** join/#asterisk cusco_ (n=trilili@2001:0:53aa:64c:1c6f:5a82:a077:acde) |
| 21:24.14 | cusco_ | hi! |
| 21:24.25 | Naikrovek | i don't subscribe to that philosophy however, I give you my word. |
| 21:24.25 | cusco_ | ast_debug() sents debug messages... where to?? |
| 21:24.35 | Naikrovek | cusco_: probably the log |
| 21:24.41 | Naikrovek | cusco_: /var/log/asterisk/full |
| 21:24.54 | cusco_ | well the mensage Im looking for its not there |
| 21:25.06 | Naikrovek | then i have no idea |
| 21:25.46 | bcrisp | well they dialed into the IVR and apparently hung up |
| 21:26.27 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:28.43 | [TK]D-Fender | checkout time, BBIAB |
| 21:30.02 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 21:30.31 | drfreeze | Hrm, looks like the bootroms are new enough |
| 21:30.52 | Naikrovek | drfreeze: do you use tftp ftp http or what |
| 21:30.58 | drfreeze | vsftpd |
| 21:31.02 | Naikrovek | ftp |
| 21:31.07 | Naikrovek | okay cool |
| 21:31.12 | Naikrovek | best results with ftp i've found |
| 21:31.28 | drfreeze | so, I'll update with the split config and see what happens |
| 21:31.37 | Naikrovek | drfreeze: i suspect it'll work fine |
| 21:31.59 | drfreeze | sure has a been a wasted day |
| 21:38.32 | Naikrovek | drfreeze: status? |
| 21:40.41 | phix | I use proftpd |
| 21:40.43 | phix | it is win |
| 21:40.58 | phix | never used vsftpd |
| 21:41.45 | Naikrovek | vsftpd is good as well |
| 21:43.55 | leifmadsen | I like vsftpd |
| 21:44.37 | bcrisp | muahaha |
| 21:44.44 | Naikrovek | DIGIUM RECOMMENDS VSFTPD you saw it here first |
| 21:44.47 | Qwell | leifmadsen: what happened to DocAwesome? |
| 21:44.47 | Naikrovek | :) |
| 21:44.50 | phix | lol |
| 21:44.53 | Naikrovek | yeah i was wondering that as well |
| 21:44.57 | leifmadsen | Qwell: just changed it up :) |
| 21:45.01 | bcrisp | i freed my queue logic and it finally got through to the only person to answer... the big boss man! |
| 21:45.12 | phix | (Naikrovek || leifmadsen) == Digium? |
| 21:45.17 | leifmadsen | people seem to love "DocAwesome" |
| 21:45.21 | Deeewayne | I used to use vsftpd but had to switch to pure-ftpd to work around some audiocodes gateway issues |
| 21:45.32 | Naikrovek | phix: liefmadsen = digium contractor i think |
| 21:45.37 | phix | nice |
| 21:45.44 | leifmadsen | aye |
| 21:45.53 | phix | so my statement is true :D |
| 21:47.26 | Naikrovek | drfreeze: well? |
| 21:48.38 | phix | leifmadsen: nice, perhaps you can help me out then :) I am getting an added echo by my asterisk server, if I bypass it, no echo |
| 21:49.08 | Naikrovek | phix: digium should not be your first line of support if you're in here |
| 21:49.15 | drfreeze | Naikrovek: just finished downloading... |
| 21:49.16 | Naikrovek | lots of folks in here are very talented and can help |
| 21:49.23 | Naikrovek | drfreeze: ah okay |
| 21:49.37 | phix | Naikrovek: true, but you can answer it too if you know the answer :) |
| 21:49.49 | Naikrovek | phix: if i knew i would answer, I assure you. |
| 21:50.09 | Naikrovek | phix: do you have phone hardware in your server |
| 21:50.45 | phix | ok, well that goes to everyone then :) I am just just talking to leifmadsen :) I just needed to address it to someone to ensure nick highlight detection on IRC and therefore some one will actually read it and comment on it :P |
| 21:51.06 | drfreeze | Naikrovek: do I need to specify the sip.ld files anywhere in my configs or is that taken care of in the sip or phone1 cfg? |
| 21:51.14 | phix | Naikrovek: yes |
| 21:51.15 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 21:51.16 | leifmadsen | phix: that's bad policy |
| 21:51.26 | Deeewayne | highlights leifmadsen's nick |
| 21:51.26 | bcrisp | why does sip show channels show 1 entry whiel sip show channelstats shows 0 entries? |
| 21:51.38 | leifmadsen | I tend to ignore questions directed to me when they should be asked more generally |
| 21:51.42 | leifmadsen | Deeewayne: I ignore you |
| 21:51.46 | Deeewayne | :-( |
| 21:51.50 | leifmadsen | Deeewayne: <3 |
| 21:51.59 | Deeewayne | hugs leifmadsen |
| 21:52.11 | leifmadsen | goes back to trying to get cacti to monitor his dell poweredge servers |
| 21:52.15 | leifmadsen | I'm installing MIBs! |
| 21:52.24 | hardwire | ew |
| 21:52.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:52.37 | phix | leifmadsen: yeah it may be half flawed, but it works for me :P |
| 21:52.42 | phix | [169170.743040] Port 1: Installed -- AUTO FXS/DPO |
| 21:52.45 | phix | [169171.797420] Port 4: Installed -- AUTO FXO (FCC mode) |
| 21:53.36 | phix | zaptel 200904 8 wctdm24xxp,ztdummy |
| 21:53.54 | phix | Naikrovek: Is that the info you were after? |
| 21:55.24 | phix | Linux werd 2.6.26-2-amd64 #1 SMP Thu Nov 5 02:23:12 UTC 2009 x86_64 GNU/Linux |
| 21:55.37 | phix | Debian Lenny |
| 21:56.19 | phix | Intel(R) Core(TM)2 Duo CPU E7400 @ 2.80GHz |
| 21:56.45 | drfreeze | Naikrovek: you there? |
| 21:57.00 | Naikrovek | drfreeze: yes |
| 21:57.03 | Naikrovek | phix: i dunno |
| 21:57.04 | phix | <PROTECTED> |
| 21:57.04 | phix | Mem: 3960 3858 102 0 533 2167 |
| 21:57.09 | voipmonk | sneezes |
| 21:57.14 | Naikrovek | phix: pastebin.com or pastie.org or whatever |
| 21:57.15 | Naikrovek | ~pb |
| 21:57.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 21:57.20 | Naikrovek | phix: ^^^^^^^ |
| 21:57.44 | rethus1 | i have convert a file for musiconhold |
| 21:57.47 | rethus1 | sox -V 4092.wav -r 8000 -c 1 --norm musik.raw |
| 21:57.52 | phix | Naikrovek: oh, am I flooding? I spread those lines over 5 minutes though :) |
| 21:57.56 | drfreeze | Naikrovek: which sip.ld file do I use? |
| 21:57.56 | rethus1 | but its much to loud |
| 21:58.07 | rethus1 | how ca i reduce the loudness |
| 21:58.15 | Naikrovek | drfreeze: okay here's where it gets cool. the phones know which one to use and they'll only obtain the one they need |
| 21:58.24 | phix | rethus1: reduce the amplitude |
| 21:58.35 | Naikrovek | drfreeze: so just point the phones at the proper place and they'll pull the proper firmware for whatever they are |
| 21:59.08 | ManxPower-work | Naikrovek: but you need BootROM 4.0 or later, IIRC |
| 21:59.17 | phix | ooohh provisioning, I need to set some of that stuff up |
| 21:59.22 | Naikrovek | no, 2.7 or later I think |
| 21:59.43 | rethus1 | phix: where can i do this |
| 21:59.46 | Naikrovek | ManxPower-work: you could be right but i SEEM to recall it working with 2.7. Either way, just upgrade the bootrom first |
| 22:00.35 | drfreeze | Naikrovek: in the cfg file there is the APP_FILE_PATH |
| 22:00.58 | drfreeze | is that ignored, or is that where I put the right sip.ld file. Or is there other magic at work |
| 22:01.12 | Naikrovek | drfreeze: one sec |
| 22:02.04 | Naikrovek | app_file_path="sip.ld" just leave it like that |
| 22:02.11 | Naikrovek | APP_FILE_PATH="sip.ld" |
| 22:02.25 | ManxPower-work | drfreeze: use the sip.cfg and other cfg files that came with the firmwre. notdoing so will screw up your phones. |
| 22:02.29 | Naikrovek | the phone will prepend it's model number in front of that, pulling the correct firmware file |
| 22:03.47 | bcrisp | <PROTECTED> |
| 22:03.49 | bcrisp | haha |
| 22:03.58 | bcrisp | doesnt know if his * server is secure |
| 22:04.00 | ManxPower-work | bcrisp: do you have allowguest=no in sip.conf? |
| 22:04.09 | bcrisp | yes manxpower |
| 22:04.21 | Chainsaw | bcrisp: If it is connected to the internet, the fine people on it will test it for you. |
| 22:04.30 | Chainsaw | bcrisp: Your phone bill will tell you if it isn't. |
| 22:04.33 | ManxPower-work | bcrisp: You get a gold star! |
| 22:04.35 | rethus1 | phix: where can i do this |
| 22:04.55 | Naikrovek | Chainsaw: that is a good test actually |
| 22:05.05 | bcrisp | ugh |
| 22:05.06 | Naikrovek | unfortunately that consultation can indeed cost a lot of money :) |
| 22:05.09 | Naikrovek | depending |
| 22:05.14 | ManxPower-work | bcrisp: the most important things are 1) non-gussable passwords 2) all calls from untrusted sources get sent to their own context(s). |
| 22:05.27 | drfreeze | Naikrovek: ok, but there is no sip.ld file, right |
| 22:05.43 | Naikrovek | drfreeze: there is, but just leave it |
| 22:05.56 | bcrisp | ok i also setup ip based security for connection into the manager interface |
| 22:05.57 | Naikrovek | polycoms, if given the proper config files, just plain work |
| 22:06.10 | drfreeze | ManxPower-work: what exactly do you mean. Did you get software media with your phones, or are you saying just look at the revision number? |
| 22:06.32 | rethus1 | my raw converted musiconhold is to loud, how can i get them much more silent |
| 22:06.38 | Naikrovek | drfreeze: the config files that are meant to be used with the firmware you downloaded are included with the firmware .zip |
| 22:06.43 | rethus1 | and how cani make them play randomly |
| 22:06.47 | rethus1 | asterisk 1.6 |
| 22:07.30 | *** join/#asterisk ticoit (n=ticoit@201.191.190.123) |
| 22:07.38 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 22:07.54 | drfreeze | Naikrovek: http://pastie.textmate.org/private/xisaj66pugzuwxsfdr1bq |
| 22:08.14 | ManxPower-work | drfreeze: in the zip file you downloaded are various .cfg files. Use those, not one from previous versions of firmware. |
| 22:09.07 | phix | rethus1: sox |
| 22:09.23 | Naikrovek | drfreeze: okay. sip.cfg should not be touched. 000000000000.cfg should be copied and adapted, one file for each phone (use scripts to make this easy) and all changes to sip.cfg should go into a file (I call it server.cfg) that should be referenced in the 00000...cfg file |
| 22:09.50 | phix | rethus1: man sox tells you how to do it in the example section |
| 22:10.13 | phix | rethus1: long story short, you append vol 0.5 at the end to reduce the volume by 50% :) |
| 22:10.23 | phix | but please read the man for more information and options you can use |
| 22:13.10 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 22:13.16 | rethus1 | phix: i have tried v 0.1, but still very loud |
| 22:13.25 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:16.57 | *** join/#asterisk rethus (n=contio@p50875567.dip.t-dialin.net) |
| 22:17.08 | rethus | so i'm back, my irc crashed |
| 22:18.29 | rethus | phix: have you another idea... this -v options take no effect |
| 22:19.44 | bcrisp | changes passwords to GUIDs |
| 22:21.05 | rethus | in asterisk docs are still the parm -w on converting wav via sox. |
| 22:21.17 | rethus | this parm does'nt exist (i got an error) |
| 22:21.32 | *** join/#asterisk vally (i=vally@ip-81-210-164-42.unitymediagroup.de) |
| 22:21.39 | rethus | maybe the loundness is cause this parameter is no longer supported? |
| 22:21.56 | bcrisp | would you guys suggest changing the bindport to non 5060 to avoid port scans? |
| 22:22.15 | Qwell | bcrisp: no |
| 22:22.50 | ChannelZ | if you don't care about receiving 'anonymous' SIP calls and want to reconfigure every other device on your local network |
| 22:23.17 | ManxPower-work | rethus: maybe you can do "man sox" |
| 22:24.16 | bcrisp | ok, im using a sip provider for incoming calls only.. should i use type = peer? |
| 22:24.31 | bcrisp | or just user? |
| 22:24.56 | [TK]D-Fender | bcrisp: peer |
| 22:25.02 | bcrisp | ty |
| 22:26.46 | *** join/#asterisk ruifreire (n=ruifreir@201-14-96-12.gnace702.dsl.brasiltelecom.net.br) |
| 22:27.13 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 22:28.13 | ruifreire | can somebody help me on how can i participate on google summer code? i saw this irc channel on http://svn.digium.com/svn/asterisk/team/group/gsoc-2009/ideas.txt |
| 22:28.33 | Chainsaw | ruifreire: I have bad news for you. |
| 22:28.41 | Chainsaw | ruifreire: Summer of Code 2009 is over. It has been for a while now. |
| 22:29.14 | ruifreire | sorry, but just one question. |
| 22:29.16 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 22:29.17 | phix | rethus: -v? I said vol |
| 22:29.26 | bcrisp | is that like summer of love, but for geeks? |
| 22:29.39 | Chainsaw | ruifreire: Do proceed. |
| 22:29.48 | rethus | phix: sox -vol ? |
| 22:30.18 | Chainsaw | ruifreire: Keep the discussion central, thank you. |
| 22:30.21 | phix | but yes, -v or --volume works as well |
| 22:30.22 | rethus | can u retype it, my irc crash, and i still see your earlyer postings anymore |
| 22:30.42 | phix | rethus: no, you append vol 0.5 at the end of the argument list to sox |
| 22:31.24 | phix | sox -V 4092.wav -r 8000 -c 1 --norm musik.raw vol 0.5 |
| 22:31.35 | phix | sox -V 4092.wav -r 8000 -c 1 --norm --volume 0.5 musik.raw |
| 22:31.39 | rethus | ok, i'll try it |
| 22:31.40 | phix | try both |
| 22:31.53 | phix | one or both should work |
| 22:32.09 | phix | if not try sticking it before 4092.wav |
| 22:32.13 | rethus | first seem to work, second not |
| 22:32.20 | phix | (the --volume 0.5 that is, not the vol) |
| 22:32.22 | phix | ok |
| 22:32.34 | phix | adjust the number to suit :) |
| 22:32.42 | rethus | now i call to hear |
| 22:32.54 | phix | ok |
| 22:32.55 | phix | gl |
| 22:33.14 | rethus | same nois, now i try 0.1 |
| 22:33.54 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 22:34.22 | rethus | mh, but think vol doesn't work too, cause the returnvalue of sox process say nothing about loudnes (gain) but do, if i use -v 0.1 |
| 22:34.34 | *** join/#asterisk mixi__ (n=mixi_@p5491EB5D.dip.t-dialin.net) |
| 22:34.53 | mixi__ | hi. somone needs a quadbri pci with powersupply for S0? |
| 22:35.35 | rethus | if i do this: sox -v 0.3 -V 4092.wav -r 8000 -c 1 musik.raw i see a line Level Adjust 0.3 (linear gain) |
| 22:36.01 | rethus | ir i do your sox -V 4092.wav -r 8000 -c 1 --norm musik.raw vol 0.1 i see i tnot |
| 22:36.36 | phix | rethus: ok, so swap the -v 0.3 before and after your output filename (4096.wav). Since to my knowledge one apply the option on input, the other mean apply on output, not that it should really matter which order you do it |
| 22:36.59 | phix | rethus: are you using SIP or ZAP device to test this? |
| 22:37.10 | rethus | sip |
| 22:37.20 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 22:37.24 | phix | ok, and the message is louder than normal voice? |
| 22:37.36 | phix | message == the audio file |
| 22:38.12 | bcrisp | what is the dialtone sound named? |
| 22:38.43 | rethus | phix: i test it, one moment |
| 22:39.18 | ManxPower-work | bcrisp: there is no dialtone sound file. |
| 22:40.09 | ManxPower-work | on zap and inband audio Asterisk generates the dialtone, on SIP the phone generates the dialtone |
| 22:40.33 | bcrisp | ok |
| 22:40.38 | rethus | phix: voice is very clearly, music very bad and loud |
| 22:40.52 | bcrisp | i guess i was trying to do something when the user types "9" |
| 22:43.11 | bcrisp | exten => _91NXXNXXXXXX,1,Dial(.............. this is how i curerntly define the outgoing exten, wasn't sure if i needed to do something special to play dialtone when they first hit 9 |
| 22:43.11 | ChannelZ | most MOH music is very bad |
| 22:43.55 | bcrisp | ChannelZ esp robot ditty |
| 22:45.21 | ChannelZ | I should replace mine with sound effects of snoring |
| 22:45.48 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 22:46.31 | Deeewayne | there are a couple not-horrible ones if you avoid the robot ditty |
| 22:46.34 | Pan3D | replace it with the sound of an endless glissando |
| 22:46.35 | bcrisp | or audio from a domestic dispute |
| 22:46.45 | rethus | any idea phix? |
| 22:47.33 | *** join/#asterisk Ccomp5950 (n=Ccomp595@71.14.144.195) |
| 22:48.44 | *** join/#asterisk jtodd (i=fg0mek4t@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 22:48.44 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:50.11 | ManxPower-work | bcrisp: zap or SIP? |
| 22:52.31 | bcrisp | sip |
| 22:54.17 | rethus | how can i test if my german language-pack for * is right installed? |
| 22:54.25 | rethus | can i set a command on cli |
| 22:54.29 | rethus | like speak hello |
| 22:56.52 | ManxPower-work | bcrisp: dialtone on SIP is controlled BY THE PHONE. |
| 22:57.07 | phix | rethus: hmmm |
| 22:57.20 | ManxPower-work | bcrisp: Asterisk doesn't even know you took the phone off hook until you are done dialing and the phone sends the block of numbers to asterisk |
| 22:57.34 | phix | bcrisp: domestic dispute that ends with gun shots imo |
| 22:57.51 | phix | s/gun shots/weapons fire/ |
| 22:58.14 | phix | oh nice, infobot knows regex |
| 22:58.30 | phix | infobot is shit |
| 22:58.34 | phix | s/shit/awesome/ |
| 22:58.39 | phix | hehe |
| 22:59.11 | bcrisp | manxpower, ok ok |
| 22:59.35 | bcrisp | i can't believe caller ID spoofing is still legal in the US |
| 22:59.58 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:00.52 | phix | bcrisp: yes, but if you spoof your caller id as lyrics to a song, you will get pwned :) |
| 23:01.11 | bcrisp | phix lol really? |
| 23:01.46 | phix | probably, lol they are currently trying to sue an ISP in AU for allowing users to download music illegally from torrents |
| 23:02.02 | phix | not to mention a few other cases like that in the UK |
| 23:02.16 | bcrisp | as well they should |
| 23:02.25 | phix | They will stop at nothing! I am not suprised if they start picking on caller ids next :P |
| 23:02.40 | phix | bcrisp: no, that is stupid |
| 23:02.51 | bcrisp | phix: not for the people that make what you're copying |
| 23:03.25 | bcrisp | its called lost income , anyways wrong channel for this discussion |
| 23:03.27 | phix | bcrisp: Lets sue the company that built a highway because it contains a long straight that cars can exceed the speed limit on |
| 23:03.28 | ManxPower-work | bcrisp: there are many, many, many legitimate reasons to callerid spoof. In fact most of my phones have their callerid spoofed. |
| 23:03.39 | rethus | so i switch to mp3 |
| 23:03.47 | ManxPower-work | For example the calleid of one of my vitelity accounts is the number of my cellphone |
| 23:03.58 | rethus | gsm and raw seems not a good choice |
| 23:04.19 | rethus | but i wonder, why the speaking files from asterisk in gsm so clear |
| 23:04.40 | rethus | anybody know a command to confert mp3 to the right 8000Hz format? |
| 23:04.43 | bcrisp | manxpower, i similarly think that TV advertisements that appear to be created by the Govt, and junk snail male that looks govt official should be illegal too |
| 23:04.54 | bcrisp | s/male/mail/ |
| 23:05.00 | manxpower | bcrisp: that I agree with |
| 23:05.01 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 23:05.06 | phix | bcrisp: really? that isn't illegal? |
| 23:05.13 | bcrisp | phix: nope |
| 23:08.02 | ecrane | Yeah... I hate it when some incredible, untrustworthy company tries to pass itself off as a credible, trustworthy institution like the government. |
| 23:11.03 | rethus | how can i create a 8000hz mono mp3 file |
| 23:11.27 | Corydon76-lap | Use sox(1) |
| 23:12.17 | Corydon76-lap | However, if you're going to that trouble, why not just create a signed linear file (sln)? |
| 23:12.50 | *** join/#asterisk Benny_132 (n=benny_13@59.167.161.153) |
| 23:12.59 | Benny_132 | anyone here have a wip310 |
| 23:13.11 | voipmonk | ecrane: it is dangerous when you are right and the government is wrong |
| 23:14.18 | bcrisp | id like to gather some DTMF from the user to check against an external system.. should i use something like fastAGI? |
| 23:15.44 | bcrisp | the purpose is to look up an acct # or something similar |
| 23:16.01 | rethus | how can i reconvert mp3 files to 8000hz for asterisk |
| 23:16.16 | [TK]D-Fender | Benny_132: Dountful. Got another question? |
| 23:18.26 | Benny_132 | i need to try and catch the error logs from the phone, and im not a cisco partner so i can't download the syslog server exe, is there anything else i can do |
| 23:19.55 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 23:23.42 | TommyBotten | Has anyone else had issues with Asterisk 1.6.1.x and recieving fax?. I have tried 1.6.1.7,10,11 and 12-rc with various versions of spandsp. |
| 23:24.23 | bcrisp | hmm no, i dont think fastagi is what im looking for |
| 23:24.26 | TommyBotten | Faxes are received OK, but an error is always returned using T.38. When not using T.38, it core dumps. |
| 23:26.36 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:34.44 | *** join/#asterisk epaphus (n=epaphus@190.10.68.228) |
| 23:35.01 | epaphus | Hello, is it possible to listen to conversations lively? instead of listening to recordings after the conversation ended? |
| 23:35.21 | p3nguin | ChanSpy() or ExtenSpy() |
| 23:35.33 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 23:35.48 | p3nguin | Each has its own usage, so choose depending on what exactly you want to hear. |
| 23:37.17 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 23:37.25 | doolittlework | hi there anyone active |
| 23:37.38 | doolittlework | my linux skills is letting me down again |
| 23:37.49 | p3nguin | Can you imagine if all 240 people answered such a silly question. |
| 23:37.56 | doolittlework | lol |
| 23:37.57 | p3nguin | s/./?/ |
| 23:38.01 | bcrisp | hello... am I here? |
| 23:38.25 | citywok | Benny_132: i know somebody that has the old linksys cordless phone, not sure if it was the 310 or 330 |
| 23:39.01 | doolittlework | i have a problem with my monitor folder i have recorded some call, i can see the files but cant move them of server using winscp |
| 23:39.45 | Benny_132 | citywok yeah im just having problems getting it to register, everything looks right but it just doesn't register and i cant find slogsrv.exe to download anywhere so i dont no what messages the phone is getting |
| 23:39.54 | doolittlework | strange thing the small file sizes can be moved but larger say 14 m up can't be moved, winsp looses connection to server |
| 23:40.13 | p3nguin | When someone calls the extension that dials my phone, if I don't answer and the call ends up going to FollowMe(), the original caller ID remains on the call and is sent to my mobile phone so I know who is actually calling me. Now, on the other hand, if I use the "forward all calls" button on my phone and forward to my mobile, it appears that my office is calling my mobile phone, so I never know who to expect. Is there some way to make ... |
| 23:40.19 | p3nguin | ... it so that the original callerID is transmitted through a SIP device's "forward" feature? |
| 23:41.45 | bcrisp | p3nguin cant you set the callerid in the dialplan? |
| 23:41.53 | p3nguin | sure |
| 23:42.11 | bcrisp | heh |
| 23:42.14 | p3nguin | Now how will I guess who is calling so that I can set the CID in the dialplan? |
| 23:42.53 | bcrisp | i guess im not familiar with your forward all calls button |
| 23:43.01 | p3nguin | Actually, if I were that psychic, I wouldn't need CID at all -- I would already know who is on the line before it rings. |
| 23:43.41 | p3nguin | SIP phones usually have a "forward" button, which allows the phone to receive a call and turn it around and dial the number you enter to forward to. |
| 23:44.16 | bcrisp | hmm |
| 23:44.17 | p3nguin | Since it seems like the phone is making a call to perform the actual forward, that is the caller ID that is being set on the call. |
| 23:44.31 | bcrisp | ic what u mean |
| 23:45.00 | p3nguin | I'm not sure it's possible to make it do anything else, but I would like to know if anyone has any ideas about it. |
| 23:45.39 | bcrisp | dont use phones, use telepathy |
| 23:45.41 | epaphus | p3nguin, iam new to asterisk.. what does that mean ChanSpy() or ExtenSpy() ?? |
| 23:45.48 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 23:45.55 | p3nguin | epaphus: Those are Asterisk applications. |
| 23:46.06 | bcrisp | ~book |
| 23:46.06 | 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 |
| 23:46.36 | p3nguin | epaphus: You can view their usages by entering on the CLI "core show application ChanSpy" and "core show application ExtenSpy", respectively. |
| 23:46.52 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 23:47.14 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 23:48.20 | HaMF | dlynes, this: https://issues.asterisk.org/view.php?id=14625 is exactly my problem. Note the last comment: "Resolved by reporter with hardware upgrade." |
| 23:49.45 | Chainsaw | HaMF: Random "Avoided initial deadlock" messages was strictly an Asterisk 1.2 issue for me. |
| 23:49.51 | Chainsaw | HaMF: I've not seen it in 1.6 |
| 23:49.53 | *** join/#asterisk etfonhomey (n=etfonhom@74-131-159-160.dhcp.insightbb.com) |
| 23:50.37 | HaMF | Chainsaw, did you use 1.4? |
| 23:50.48 | Chainsaw | Never did, no. I skipped 1.4 |
| 23:50.59 | HaMF | hmm |
| 23:51.51 | HaMF | Chainsaw, what hardware do you use? |
| 23:51.59 | Chainsaw | HaMF: Telephony or the box itself? |
| 23:52.29 | *** part/#asterisk rethus (n=contio@p50875567.dip.t-dialin.net) |
| 23:53.25 | Chainsaw | looks at HaMF |
| 23:53.30 | HaMF | Chainsaw, principally the box itself but I'd like to hear about your telephony-hardware aswell if you want do tell me :) |
| 23:53.46 | Chainsaw | HaMF: HP Proliant DL365 G1 (dual Opteron 2218, 2G RAM) |
| 23:54.00 | Chainsaw | HaMF: Telephony card is a Digium TDM400P with one FXO. |
| 23:54.31 | Chainsaw | HaMF: It interfaces with Patton gateways (4118, 4634) and handsets (Cisco 4960G) over SIP. |
| 23:54.43 | bcrisp | if i want to exchange information between * and another server in lala land I should use AGI correct? |
| 23:55.07 | Chainsaw | HaMF: I don't believe 4960 is correct. Let me confirm that. |
| 23:55.18 | HaMF | Chainsaw, nvm |
| 23:55.29 | Chainsaw | HaMF: 7960G. |
| 23:55.47 | HaMF | Chainsaw, thanks. We use TE110-Wildcards here so at least it should not be connected to the card I guess. |
| 23:55.57 | Chainsaw | HaMF: It's not. I had it often on 1.2 |
| 23:56.12 | Chainsaw | /var/log/asterisk/full.0:Jul 25 10:50:08 WARNING[26371] channel.c: Avoided initial deadlock for '0x135eab6f620', 9 retries! |
| 23:56.43 | HaMF | I'll set up an asterisk 1.6 system tomorrow... Thats my best guess I suppose. |
| 23:57.08 | Chainsaw | HaMF: I hope the jump isn't as big as 1.2 -> 1.6 |
| 23:57.12 | Chainsaw | HaMF: Because that was rather epic. |
| 23:57.16 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 23:57.53 | doolittlework | can some please help me recover wav files of asterisk server, i use winscp but it keeps on crashing |
| 23:58.03 | HaMF | Chainsaw, I guess not. even though zaptel was renamed to dahdi the config-files should be compatible in most cases. |
| 23:58.28 | doolittlework | i have tried moving to usb drive to no avail |
| 23:58.35 | Chainsaw | HaMF: Zaptel->DAHDI wasn't my problem ;) |
| 23:58.41 | Chainsaw | HaMF: This was about the dial plan. |
| 23:58.58 | HaMF | oh this won't be the problem. my dialplan is rather simple... |
| 23:59.21 | HaMF | Chainsaw, no macros, no agi-bin-stuff just simple pattern matching in only 14 extensions |
