| 00:04.02 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:13.42 | bman | bak |
| 00:13.53 | bman | ok got the addons package and the asterisk for 1.4 |
| 00:14.05 | Katty | (= |
| 00:14.13 | Katty | my onions are almost done caramalizing |
| 00:14.17 | bman | i got coffee |
| 00:14.21 | bman | :P |
| 00:14.38 | Katty | i have bison burgers, with parsnip chips |
| 00:14.47 | Katty | and of course caramalized onions and portabello mushrooms |
| 00:14.49 | Katty | soon, anyway |
| 00:14.56 | bman | can i marry you |
| 00:15.06 | bman | plz |
| 00:15.17 | bman | heh |
| 00:15.23 | bman | that sounds awesome really |
| 00:15.23 | Katty | ^_- |
| 00:15.26 | Tech_Travis | Jerry87: I've got the queues setup similar to the post, people can log in and out, and when a call comes through it goes to their extension which is cool. I'm thinking of an On-Call queue for people after hours but don't want them tied to their softphone. I'd like the call to route to their cell phone while they are logged into the queue. |
| 00:15.29 | Katty | so uhh..i'm going to go check on dinner |
| 00:15.34 | Katty | disappears before things get REALLY weird |
| 00:15.57 | bman | ill do dishes |
| 00:16.11 | bman | lies |
| 00:16.15 | bman | all lies |
| 00:16.54 | Katty | pouts, not done caramalizing yet |
| 00:17.29 | Katty | jblack: ping! |
| 00:17.50 | jblack | katty: pong! |
| 00:18.11 | Katty | :> |
| 00:18.17 | jblack | wassup |
| 00:18.18 | Katty | jblack: do you like meatloaf, mister black |
| 00:18.30 | jblack | So much, that I like both types. |
| 00:18.41 | Katty | ...both, types? |
| 00:18.46 | Katty | there's more than 1 kind? |
| 00:18.53 | Katty | must be missing out |
| 00:18.57 | jblack | the fatty dinner kind, and the fatty rockstar kind |
| 00:19.03 | Katty | oh. that. |
| 00:19.15 | jblack | which one did you mean? |
| 00:19.16 | Katty | do you have a recipe for meatloaf? |
| 00:19.31 | jblack | nah. I'm not much of a cook |
| 00:19.41 | Katty | ever since i found that pizza crust (batter) that allowed you to make smaller, single portion pizzas...i've been having this idea to make baby meatloaves. |
| 00:19.47 | Katty | hmm. bummer :< |
| 00:19.52 | bman | chan_ooh323.c:2188: error |
| 00:19.53 | Katty | i shall keep asking around. |
| 00:20.02 | bman | i get pages of those when i try to make on addons |
| 00:20.15 | jblack | My ex-wife has my mother's recipes. |
| 00:20.32 | jblack | and it can be scaled down to make several mini-loaves. |
| 00:20.44 | jblack | I know, cause my kid made it |
| 00:20.47 | bman | chan_ooh323.c:3390: error: expected ')' before string constant |
| 00:20.53 | bman | ls |
| 00:21.05 | *** part/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
| 00:21.36 | bman | what is chan_ooh324? |
| 00:21.40 | bman | 323? |
| 00:21.45 | *** join/#asterisk z0k3b3r (n=Zokeber@unaffiliated/zokeber) |
| 00:21.47 | jblack | I emailed her for the recipe. |
| 00:22.04 | bman | Objective Systems H323 Channel |
| 00:22.07 | jblack | It's pretty basic. I know there's beef and bread crumbs in it, typicall a ketchup glaze |
| 00:22.16 | bman | is that required for any 323 usage? |
| 00:23.23 | bman | yeah |
| 00:23.39 | bman | this is why i wanted to use the deb package instead of compiling |
| 00:23.47 | bman | i dont know what any of this means |
| 00:25.35 | bman | any help? |
| 00:25.48 | bman | i want sip, jabber, and mysql support only |
| 00:25.54 | bman | no telephony hardware |
| 00:26.08 | *** join/#asterisk ming_zym (n=ming_zym@114.251.86.0) |
| 00:29.22 | Katty | omnomnomnomnoms dinner |
| 00:29.39 | Katty | jblack: oooh, thank you :> |
| 00:30.30 | Katty | hmm. |
| 00:30.35 | Katty | parsnips are interesting |
| 00:30.44 | Katty | they're like a cross between a carrot, potato, and a turnip |
| 00:30.54 | bman | any asterisk help? |
| 00:31.24 | jblack | bman: Which verison of ubuntu? |
| 00:31.43 | bman | debian |
| 00:32.01 | jblack | I don't know what versions are in debian. |
| 00:32.11 | jblack | apt-get install asterisk should be a great start for you though. |
| 00:33.16 | bman | i had it installed, and i had the setup for realtime done but it wouldnt show sip module as loaded |
| 00:33.25 | bman | so someone in here told me to buiild from source |
| 00:33.30 | bman | which is just confusing me |
| 00:33.43 | jblack | You know better than to listen to them, I'm sure. |
| 00:33.58 | Katty | jblack: you really are missing a fabulious dinner |
| 00:34.02 | jblack | You're in a channel. The first rule of any channel is "must say build from sources" |
| 00:34.22 | jblack | Katty: I can't help that. :( |
| 00:34.37 | Katty | i'll eat some extra for you ;) |
| 00:34.40 | Katty | not to fear! |
| 00:34.45 | bman | let me get back to where i was |
| 00:34.46 | bman | hold on |
| 00:34.48 | Katty | it's a real shame you don't cook. i'd share these recipes with you |
| 00:34.58 | jblack | I do kinda cook. |
| 00:35.06 | jblack | I had tomatos and cucumbers for breakfast |
| 00:35.16 | Katty | for breakfast? ^_- |
| 00:35.20 | Katty | seems kinda odd |
| 00:35.21 | jblack | it's easy. |
| 00:35.41 | Katty | tomato, cucumber, and? |
| 00:35.48 | Katty | i'm assuming some sort of salad. |
| 00:35.53 | jblack | I was outa protein drink, didn't wanna make fake egg, so I sliced up a cucumber and a tomato. |
| 00:36.08 | Katty | and just ate it raw? |
| 00:36.23 | jblack | Yeah. what's weird about that? |
| 00:36.31 | Katty | i don't think i've ever had it that way |
| 00:36.36 | jblack | You've never had sliced tomatoes sprinkled with salt? |
| 00:36.39 | Katty | no |
| 00:36.44 | jblack | Or just eaten cucumber slices? |
| 00:36.47 | Katty | no |
| 00:36.50 | jblack | You're missing out |
| 00:36.53 | Katty | hmm |
| 00:36.54 | jblack | They're both great snacks. |
| 00:37.02 | Katty | well. i don't like plain tomato. |
| 00:37.09 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 00:37.11 | Katty | but cucumber is good. i've had it on crackers or party rye |
| 00:37.12 | jblack | suit yourself. |
| 00:37.34 | Katty | giving me all sorts of thoughts you are! |
| 00:37.35 | jblack | Some day, I'll get me one of those mail-away-foreign brides. |
| 00:38.07 | Katty | http://www.city-of-brides.com/ <- get started |
| 00:38.27 | jblack | Oh, no, I don't want one of those. |
| 00:38.44 | jblack | oh, that sounded rather objectifying. |
| 00:38.56 | Katty | it's actually not. |
| 00:39.05 | Katty | you pay dearly for their contact information |
| 00:39.07 | jblack | I'm not looking for a model. I'm looking for a person. |
| 00:39.11 | Katty | you don't really get anything but a chance to talk to them |
| 00:39.25 | jblack | and yeah, a lot of those places are scams. |
| 00:39.26 | Katty | it's all very legitimate |
| 00:39.50 | snadge | a wife.. interesting prospect :P |
| 00:39.56 | jblack | Not saying that all, or even most, of them are scams, or that those aren't legitimate offers... |
| 00:40.35 | jblack | well, it gets complicated. |
| 00:41.46 | jblack | I think winning the lottery and hiring someone to take care of me would just be better. ;) |
| 00:42.14 | *** join/#asterisk spck (n=spck@75-135-75-112.dhcp.mdsn.wi.charter.com) |
| 00:42.14 | bman | ok bak to where i was |
| 00:42.39 | jblack | bman: apt-get install asterisk; profit! |
| 00:42.49 | bman | chan_iax2.c: Unable to open IAX timing interface: No such file or directory |
| 00:42.52 | jblack | Probably come with 1.4 |
| 00:43.02 | bman | still not actually working |
| 00:43.22 | jblack | start off with an empty iax.conf |
| 00:43.31 | bman | moonglum:~/asterisk-1.4.26.3# asterisk -rx "sip show peers" |
| 00:43.32 | bman | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
| 00:43.37 | jblack | add a [general] section, start adding authentication contexts... |
| 00:43.40 | jblack | it's all in the book |
| 00:43.50 | [TK]D-Fender | jblack: Models are people too you know... |
| 00:43.56 | jblack | [TK]D-Fender: You lie |
| 00:44.04 | jblack | They're styrofoam. |
| 00:44.10 | [TK]D-Fender | </joewilson> |
| 00:44.25 | bman | i dont want iax |
| 00:44.30 | bman | i want sip |
| 00:44.37 | jblack | bman: Then disable the iax module. Again, that's in the book. |
| 00:44.48 | jblack | I'm starting to think that you didn't read the book. |
| 00:44.54 | bman | no i didnt |
| 00:44.56 | Katty | jblack: hmm, yeah, i don't think that'd be very fun. |
| 00:45.06 | bman | im just trying to restore a previously working service |
| 00:45.15 | Katty | jblack: i mean winning the lottery would be fine and all, but then you have unknown relatives and New Friends coming out of the woodwork. |
| 00:45.16 | jblack | You're not one of those morons that sits in an irc channel asking basic questions 5 hours a day because you didn't take the time to read the book, are you? |
| 00:45.29 | jblack | I don't think you are. I think you're probably a smart guy. |
| 00:45.30 | bman | not really trying to become a asterisk expert when its a deb package |
| 00:45.51 | jblack | OHhhh. Then hire someone. I recommend [TK]D-Fender , even though he won't be a facebook friend to me |
| 00:45.55 | Katty | i'd recommend finding a consultant then |
| 00:46.06 | bman | go back to talking food |
| 00:46.17 | jblack | too late. Now I care about you. |
| 00:46.20 | Katty | scowls. |
| 00:46.30 | jblack | I'm like, all stressed out about your situation. |
| 00:46.33 | bman | i explained all this not 2 hours ago |
| 00:46.36 | bman | its cool |
| 00:46.42 | Katty | what are you trying to say there, bman |
| 00:46.51 | jblack | Oh man. How dare I be at the store 2 hours ago. Thatt is so inconsiderate of me! |
| 00:46.57 | jblack | I'm really sorry, man. I should have been here for you |
| 00:47.17 | Katty | that sounded EXTREMELY sexist to me. |
| 00:47.20 | bman | you dont know would have been a acceptable answer |
| 00:47.38 | jblack | ?!? Me being at the store is sexist? |
| 00:47.46 | Katty | jblack: yes, of course. naturally |
| 00:47.55 | Katty | shares parsnip chips and avocado chunks with jblack |
| 00:48.00 | jblack | Hrmmm. I guess it is woman's work... |
| 00:48.19 | jblack | runs and hides behind [TK]D-Fender |
| 00:48.30 | Corydon76-dig | gets his bitch ass back in the kitchen and makes himself some PIE |
| 00:48.37 | bman | using a support channel for chat and babble helps so many |
| 00:48.39 | jblack | rofl |
| 00:48.39 | *** part/#asterisk bman (n=bman@emsn-02-053.dsl.netins.net) |
| 00:49.28 | *** join/#asterisk keith4_ (n=keith@unaffiliated/keith4) |
| 00:49.29 | jblack | Just for the record, the woman's work crack was just a bad joke. As a single father, I did my fair share of grocery trips and cooking |
| 00:49.46 | Katty | jblack: i wasn't actually directing that comment towards you |
| 00:49.50 | keith4_ | can i forcefully lower the volume of another participant in a meetme channel? |
| 00:49.52 | jblack | I know. |
| 00:49.54 | Katty | k |
| 00:50.04 | jblack | keith4_: Yeah. Scream "STFU you noisy bastard!" |
| 00:50.08 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 00:50.13 | keith4_ | tried that. just got "huh? i'm not shouting" |
| 00:50.22 | jblack | word for word? |
| 00:50.34 | keith4_ | well, no. it was more like 'WHY ARE YOU SHOUTING?" |
| 00:50.41 | jblack | Ok. That's not forcefully enough. |
| 00:50.54 | jblack | Try "If you don't talk more quietly, I'm going to rape your father" |
| 00:51.05 | jblack | That's pretty forceful. |
| 00:51.06 | ppc | heh |
| 00:51.14 | Katty | i seem to have a splinter :< |
| 00:51.16 | *** join/#asterisk galeras (n=galeras@190.146.13.27) |
| 00:51.24 | Katty | bbl, must visit the Expert Splinter Remover |
| 00:51.32 | jblack | keith4_: Seriously though... I don't think there's anything built in to the "ui" of sorts. |
| 00:51.41 | *** join/#asterisk manxpower (n=ewieling@82.sub-70-222-194.myvzw.com) |
| 00:51.43 | jblack | There _might_ be something in the console or ami, but I doubt it. |
| 00:52.03 | manxpower | ~answers |
| 00:52.04 | infobot | methinks 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 |
| 00:52.19 | keith4_ | console would be fine |
| 00:52.23 | jblack | if he's coming in under his own context, you might be able to play with rxgain or txgain, somehow, but that may just be zap that has that, and would require a reload anyways |
| 00:52.58 | manxpower | what is the problem/goal? |
| 00:53.02 | Katty | returns. |
| 00:53.04 | Katty | hi manx! |
| 00:53.21 | manxpower | hello Katty |
| 00:53.21 | keith4_ | ah. MeetMeAdmin() can do it |
| 00:53.22 | jblack | manxpower: ONE OF HIS MEETME PARTICIPANTS SHOUTS |
| 00:53.34 | Katty | that's awful news. |
| 00:53.37 | manxpower | jblack: Ah. |
| 00:53.38 | jblack | Really? I know there's a mute, but a volume adjust? |
| 00:53.51 | keith4_ | yep. that's what i just said, to myself |
| 00:54.24 | manxpower | jblack: Meetme uses zaptel for audio mixing. I suspect that it would me reasonably easy to add a gain option to MeetMe. A bounty might get the code written. |
| 00:54.29 | jblack | there it is, t and T. |
| 00:54.36 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:54.41 | Katty | manxpower: looks like someone already paid a bounty |
| 00:54.47 | jblack | He's right. There's meetmeadmin options. t and T. |
| 00:54.57 | manxpower | also be sure to consider 1.6, the MeetMe in 1.6 might even have the option |
| 00:54.58 | Katty | which is lovely news. |
| 00:55.27 | keith4_ | now, how do I fake a call to an extension via the console? |
| 00:55.28 | jblack | I still think the better option is to give him doctored photos of his mummi nekked.. except for the BDSM attire... |
| 00:55.36 | jblack | That'll make him speechless, and the problem is solved. |
| 00:55.37 | Katty | i'm excited about the audio buffer for SendFax() in 1.6 |
| 00:56.01 | Amorsen | keith4: originate? |
| 00:56.27 | jblack | keith4_: You could control with two lines. One phone for talking, the other one, press commands. Im sure that meetme has a way for an admin to do controls while in-band too |
| 00:56.32 | jblack | perhaps features. |
| 00:56.50 | Katty | there is a little web appy that works with meetme |
| 00:56.55 | Katty | i believe it has audio controls. |
| 00:57.08 | Katty | tries to remember name |
| 00:57.21 | keith4_ | oooh |
| 00:57.22 | Katty | infobot: Web Meetme? |
| 00:57.23 | infobot | ACTION forces Meetme to muster up a db backed, web fronted application for keeping track of lawns mowed |
| 00:57.27 | Katty | :< |
| 00:57.49 | Katty | http://areski.net/Web-MeetMe/about.php |
| 00:58.03 | keith4_ | yes! |
| 00:58.13 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 00:58.28 | Katty | hmm. no volume control. |
| 00:58.42 | Katty | bummer. |
| 00:58.50 | Katty | but! it clearly works with asterisk. |
| 00:58.54 | *** join/#asterisk bman (n=bman@emsn-02-053.dsl.netins.net) |
| 00:58.58 | Katty | you could probably edit the dialplan a bit, and edit the php |
| 00:59.10 | keith4_ | yes yes |
| 00:59.14 | keith4_ | rubs his hands together |
| 00:59.49 | bman | any help installing asterisk realtime would be appreciated |
| 01:00.03 | Katty | hello again. |
| 01:00.47 | bman | anyone at all |
| 01:01.15 | Katty | keith4_: hmm. that doesn't look like the most recent version |
| 01:01.30 | Katty | keith4_: either that or i'm looking at two completely differen't products |
| 01:01.51 | keith4_ | for now, i just wrote a quick MeetMeAdmin loop to lower this one jerk |
| 01:01.58 | keith4_ | i'll investigate a longer-term solution tomorrow ;-) |
| 01:02.03 | keith4_ | but this looks like a good start |
| 01:02.03 | Katty | hmm. okay. more recent version. just not posted on his website. |
| 01:02.12 | Katty | http://areski.net/Web-MeetMe/about.php <- 4.0 |
| 01:02.29 | Katty | ^-- err 1.2 |
| 01:02.52 | Katty | http://sourceforge.net/projects/web-meetme/ <- 4.0 |
| 01:03.00 | keith4_ | oh |
| 01:03.07 | Katty | there were major changes between 1.2 and 1.6, especially in regards to api |
| 01:03.14 | Katty | well, especially everything, really |
| 01:03.22 | Katty | thinks |
| 01:03.27 | Katty | no, this probably just uses manager.conf |
| 01:03.32 | Katty | i doubt it matters. |
| 01:03.36 | galeras | weird, in a PRI incoming calls are fine, outgoing calls are failing, please take a look off PRI DEBUG at http://pastebin.ca/1669167 |
| 01:03.40 | Katty | but i'd still recommend using the most current version |
| 01:03.49 | keith4_ | "t" doesn't seem to lower the volume much, though |
| 01:04.16 | Katty | galeras: i would call your telco and ask them to tell you if they are recieving anything from your pri |
| 01:04.29 | Katty | galeras: if they are receving something, they can usually tell you exactly what's snickerdoodled up |
| 01:04.34 | manxpower | galeras: what country are you in? |
| 01:04.48 | galeras | Colombia |
| 01:04.53 | Katty | that is also a lot of macros. |
| 01:05.17 | galeras | mm, give me a sec, i'll past a short debug |
| 01:06.22 | galeras | there is a shorted debug at http://pastebin.ca/1669171 |
| 01:06.26 | galeras | *shorter |
| 01:06.27 | *** join/#asterisk sahafeez (n=sahafeez@12.180.45.140) |
| 01:06.34 | manxpower | galeras: Ext: 1Â Cause: Unallocated (unassigned) number (1), |
| 01:06.43 | manxpower | Try calling a working number |
| 01:06.54 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:58a5:7f49:1128:cdc4) |
| 01:07.25 | manxpower | galeras: also make sure you have either no setting for pridialplan or have pridialplan=unknown |
| 01:07.40 | manxpower | with no other pridialplan options. |
| 01:08.13 | manxpower | I don't know the specific requirements for your country, but for most places that is the correct way. |
| 01:09.19 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-31-240.dsl.lsan03.sbcglobal.net) |
| 01:09.26 | manxpower | The fact that your dialplan had to take 129 lines of output to get to that point is an astounding example of a overly complex dialplan |
| 01:09.41 | galeras | manxpower: Thanks verrrryyyy much, u r the man |
| 01:10.08 | galeras | pridialplan=unknown worked |
| 01:11.27 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 01:13.02 | jblack | too much foreplay |
| 01:13.26 | jblack | or, as we say in #asterisk... "fourplay" |
| 01:13.52 | leifmadsen | snap yo |
| 01:15.04 | russellb | leifmadsen: YOU! |
| 01:15.04 | *** join/#asterisk Chesther (n=cam2@cpe-67-241-8-206.twcny.res.rr.com) |
| 01:15.11 | leifmadsen | russellb: ME! |
| 01:15.11 | Sargun | haha |
| 01:15.17 | leifmadsen | russellb: how goes?! |
| 01:15.34 | russellb | goooooood. |
| 01:15.34 | jblack | mean... I know the chances of this are nill... but to get allison to record "press 4 for play" , along with a... suggestive recording... |
| 01:15.49 | russellb | she'd do it |
| 01:15.55 | leifmadsen | jblack: the chances are quite high actually |
| 01:15.58 | russellb | there are just a few things that she won't say |
| 01:15.59 | leifmadsen | you just have to purchase a prompt |
| 01:16.21 | jblack | That would be worth the money... |
| 01:16.21 | leifmadsen | allison is pretty liberal with what she'll say |
| 01:16.23 | Katty | jblack: http://42ndrecipestreet.blogspot.com/2009/11/parsnip-oven-fries.html |
| 01:16.24 | russellb | and for those you just use cepstral :-) |
| 01:16.37 | leifmadsen | you can make allison say the things you she won't with cepstral :D |
| 01:16.45 | russellb | leifmadsen: i win |
| 01:16.48 | jblack | "Just gimme a good 30 or 40 words suitable for exciting males". =) |
| 01:16.49 | leifmadsen | russellb: you owe me a beer |
| 01:16.56 | jblack | grins |
| 01:16.58 | russellb | lies |
| 01:17.03 | leifmadsen | I never lie |
| 01:17.20 | jblack | Katty: Yumm. |
| 01:17.20 | leifmadsen | well there was that one time, but I was trying to save the world |
| 01:18.49 | snadge | will asterisknow and freepbx be a suitable SIP exchange or proxy.. for a small office that uses sip phones? |
| 01:19.07 | snadge | i know asterisk will do it.. and i can see books dedicated to the subject |
| 01:19.19 | leifmadsen | asterisknow uses asterisk... |
| 01:19.27 | leifmadsen | freepbx is a gui on top of asterisk as well |
| 01:19.37 | leifmadsen | although I think it runs on other platforms as well now |
| 01:19.46 | leifmadsen | asterisk doesn't make a very good proxy |
| 01:19.57 | leifmadsen | (since it's a back2back user agent) |
| 01:21.09 | snadge | the scenario is that we have a voip account from our provider.. which happens to use SIP.. now i've added a trunk, and asterisk has established a connection with the trunk.. i can see from the status.. i've added an inbound and outbound route |
| 01:21.34 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 01:21.36 | leifmadsen | uh huh... |
| 01:21.46 | jblack | snadge: what's your goal? Phones can call each other? |
| 01:22.16 | snadge | to begin with.. just to place calls via the asterisk server .. from a sip client on the local network |
| 01:22.19 | jblack | If you were really talking proxy, you'be be talking about phones that directly register with upstream. Which is unnecessarily difficult. |
| 01:22.35 | jblack | Ok. So, don't proxy. Get asterisk to hook up to the provider, and get the phones to hook up with asterisk. |
| 01:22.36 | snadge | eventually.. to have the asterisk server answer incoming calls.. play an IVR message, and then route the incoming calls based on the extension number ? |
| 01:22.38 | bman | hah |
| 01:22.44 | bman | my problem was permissions |
| 01:22.49 | bman | didnt need asterisk book |
| 01:23.02 | leifmadsen | snadge: right, that isn't really a proxy, that's what asterisk does though |
| 01:23.06 | leifmadsen | a proxy wouldn't do IVR |
| 01:23.15 | jblack | Good job You beat the man, you illiterate fu.. um, friend. |
| 01:23.27 | jblack | snadge: A proxy does something special that isn't what you want anyways. =) |
| 01:23.34 | leifmadsen | and you probably don't mean IVR, you likely mean auto-attendant |
| 01:24.18 | snadge | right.. ok so for now.. how do i connect to this freepbx/asterisknow server i have setup.. and place a call through it? |
| 01:24.39 | leifmadsen | that's a GUI issue that probably is more appropriate for those forums |
| 01:24.39 | jblack | snadge: Asterisk is designed to connect to other servers via sip and/or iax. And to route calls between the two, |
| 01:24.43 | snadge | i've kind of being going from voip providers howtos on how to setup asterisknow which basically only really gives you the basic settings |
| 01:25.07 | Katty | jblack: http://42ndrecipestreet.blogspot.com/2009/11/bison-burgers.html |
| 01:25.09 | jblack | snadge: You get the phone to register with asterisk and to place calls to it. and yu get asterisk to route the call through the provider. It's all in the book, really. |
| 01:25.14 | leifmadsen | ok, I'm outta here, time to hang out with the feemalien |
| 01:25.26 | jblack | katty: I'm a fat diabetic. Why are you being so mean to me? |
| 01:25.30 | snadge | ok.. so i'll buy or download the book :( |
| 01:25.37 | jblack | Do you go around to AA meetings handing out beers? |
| 01:25.43 | leifmadsen | that won't really help as it doesn't cover GUI based systems |
| 01:25.44 | Katty | )_= |
| 01:25.50 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 01:25.54 | leifmadsen | it'll help if you're planning on building from vanilla asterisk though |
| 01:26.00 | jblack | sorry. That was a little harsh of me |
| 01:26.22 | jblack | Oh wow, look at that thing. That looks great. |
| 01:26.24 | bman | for future reference, to use the debian packages for asterisk you must chown -R asterisk.asterisk /etc/asterisk |
| 01:26.28 | snadge | leifmadsen: my plan was to try using a GUI based system.. and then failing that.. do it the hard way.. i figured i should at least try an easier way first :) |
| 01:26.29 | Katty | hugs jblack |
| 01:26.40 | bman | even if your config is perfect it wont work without it |
| 01:26.57 | jblack | bman: No, you don't... Go ahead. That's fine for you. |
| 01:27.01 | Katty | jblack: i am sorry. here: http://serendip.brynmawr.edu/sci_cult/evolit/s07/tomato.jpg |
| 01:27.20 | jblack | heh |
| 01:27.26 | bman | pz |
| 01:27.32 | *** part/#asterisk bman (n=bman@emsn-02-053.dsl.netins.net) |
| 01:27.53 | jblack | That is obviously a guy that solves problems which chmod a+rwx -R |
| 01:28.08 | russellb | i do that! |
| 01:28.12 | hardwire | bad! |
| 01:28.12 | jblack | "Oh, apache can't read the files. Let's let everyone rwx, and we'll have a great webserver!" |
| 01:28.19 | jblack | russellb: You. Lie. |
| 01:28.38 | Katty | wibbles |
| 01:28.44 | russellb | i do that for all of our servers |
| 01:28.45 | russellb | is that bad? |
| 01:29.03 | jblack | wtf? |
| 01:29.05 | hardwire | do they have nice wholesale accounts on them? |
| 01:29.17 | russellb | in the spirit of openness, man |
| 01:29.18 | jblack | russellb trollin' on one side, katty teasing me with food on the other... |
| 01:29.29 | jblack | russellb: Oh, the inner RMS in you snuck out, eh? |
| 01:30.44 | Katty | i think he needs a hug. |
| 01:30.50 | Katty | hugs russellb |
| 01:32.22 | russellb | <3 |
| 01:32.37 | jblack | I'm gonna get one of those bouncy titt apps for my android phone |
| 01:32.49 | Katty | TWEETTWEET TWEETTWEET |
| 01:33.17 | jblack | Oh, for my alarms, I play dennis leary's "Life's gonna suck when you grow up". |
| 01:33.23 | Katty | :< |
| 01:33.41 | jblack | wonders if the breast physics thingies are "games" or "apps" |
| 01:33.48 | Katty | probably app. |
| 01:33.57 | Katty | i don't think you do anything but shake the phone |
| 01:34.56 | jblack | wonders what a lite version would be... small breasts? Only 1 breast? with pasties on? |
| 01:35.38 | jblack | keeps an eye out for a swingin di.. well, for a lady app for the ladies. |
| 01:36.16 | Katty | do me a favor and don't link it to me if you find it |
| 01:36.27 | Katty | ;> |
| 01:38.30 | jblack | I am going to try BigJapBusts and Jack&Jiggle |
| 01:38.54 | jblack | Oh, and definitely botboobs |
| 01:38.57 | Katty | facepalm |
| 01:39.06 | jblack | In honor of Asimov.. |
| 01:39.55 | *** join/#asterisk Iamnacho (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 01:40.52 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 01:41.12 | jblack | Ohhh, like jello |
| 01:41.34 | Katty | sugar free? |
| 01:41.45 | Katty | pectin or gelatin effect? |
| 01:41.55 | jblack | No. I'd say more like pudding, becuase it's definitely dairy... |
| 01:41.57 | jblack | Lotsa dairy |
| 01:42.04 | *** join/#asterisk Tech_Travis (n=tech_tra@cpe-76-87-9-130.socal.res.rr.com) |
| 01:43.09 | Katty | http://www.ae.com/Images/laydowns/front/0382_1354_410.jpg <- |
| 01:44.05 | jblack | That's a.. coat? |
| 01:44.13 | Katty | yes'r |
| 01:44.20 | Katty | a very pretty one. |
| 01:44.41 | Katty | did you notice the buttons? |
| 01:44.50 | jblack | looks closer |
| 01:45.01 | Katty | anchor (= |
| 01:45.09 | jblack | Sure. But that's not a real navy peacoat. |
| 01:45.15 | Katty | no, not at all. |
| 01:45.17 | Katty | i wouldn't want a real one |
| 01:45.22 | jblack | Real navy peacoats are ugly as shit. |
| 01:45.29 | Katty | yes indeed. |
| 01:46.22 | jblack | I wonder which one I should get...http://www.androlib.com/android.developer.jet-boi-wireless-iFB.aspx |
| 01:47.04 | Katty | the highest rated? |
| 01:47.24 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 01:47.42 | jblack | I don't lke the face. |
| 01:47.47 | Katty | woah. cashmere coats |
| 01:48.02 | Katty | i don't think you're going to be looking at the face |
| 01:48.08 | jblack | Good point. |
| 01:48.15 | jblack | Two good points, now that I think about it |
| 01:48.22 | *** join/#asterisk jermudgeon (n=jhaustin@216-67-61-242.static.acsalaska.net) |
| 01:48.35 | jblack | falls off his chair in laughter |
| 01:48.46 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 01:48.47 | Katty | oh dear. |
| 01:49.08 | jblack | cmon, that was funny! |
| 01:49.12 | Katty | it was ;) |
| 01:49.14 | Katty | applauds |
| 01:50.39 | jblack | the reviews in the android market are terrible. |
| 01:51.21 | jblack | And not "this is so immoral" terrible. More like "Looks awful". |
| 01:51.43 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:51.43 | Katty | :< |
| 01:52.37 | Katty | http://www.youtube.com/watch?v=NpKvXw4wRvs |
| 01:55.13 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 01:55.34 | jblack | Oh man. girlicious sucks |
| 01:55.49 | Katty | :< |
| 01:55.55 | Katty | screenshot? |
| 01:56.31 | jblack | http://www.youtube.com/watch?v=gncx1z_PKd8 |
| 01:56.59 | jblack | I think this is the result of that pussycat doll reality show. |
| 01:57.57 | Katty | ^_- |
| 01:57.58 | Katty | creepy |
| 01:58.00 | jblack | Same thing, but somehow even worse. http://www.youtube.com/watch?v=V92UZ0PnW1M |
| 01:59.50 | jblack | They're so bad, that they have their names embossed on their micro-skirts. http://www.youtube.com/watch?v=p0Lr1UPvkF0 |
| 02:00.31 | Katty | covers eyes |
| 02:04.50 | snadge | i'd watch it.. but i might then have to kill myself ;) |
| 02:05.20 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 02:06.26 | jblack | http://www.youtube.com/watch?v=35LqQPKylEA&feature=channel |
| 02:07.29 | Katty | watches |
| 02:07.30 | snadge | shite.. i have asterisknow installed in virtualbox.. and i decided to move it onto one of our xen development boxes (off my workstation) |
| 02:08.30 | Katty | giggles |
| 02:08.59 | jblack | This is better. |
| 02:09.03 | jblack | http://www.youtube.com/watch?v=bTT9-YfgeTU&feature=channel |
| 02:10.47 | Katty | watches second one |
| 02:12.52 | Katty | i like the google one better :P |
| 02:13.46 | jblack | a hilarious thing to watch is people griefing on team fortress 2. |
| 02:15.54 | jblack | http://www.youtube.com/watch?v=DOAKOMHZgCM&feature=channel |
| 02:20.48 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:21.30 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 02:22.31 | jblack | Katty: whoah! What is this? http://www.youtube.com/watch?v=KIEQXIkXrPU |
| 02:25.59 | *** join/#asterisk voipmonk (n=voipmonk@69.172.97.20) |
| 02:30.16 | Katty | looks |
| 02:35.22 | Katty | bleh |
| 02:37.22 | snadge | oh yeah anyways.. i just copied the xen kernel onto the asterisknow/centos image.. and booted it up.. and it appears to be working.. but im getting "dahdi" errors.. whatever they are |
| 02:38.29 | *** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru) |
| 02:42.26 | snadge | i dont suppose dahdi has been ported to xen? |
| 02:42.41 | snadge | it doesnt matter.. because i dont need to access with any physical hardware interfaces.. im only using SIP |
| 02:49.23 | Sier | asterisk is hard to configure :( |
| 02:49.42 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 02:49.45 | *** join/#asterisk Caplain (n=shayne@2001:470:5:fb:58a5:7f49:1128:cdc4) |
| 02:50.07 | snadge | it kind of is yes |
| 02:54.35 | *** join/#asterisk canadait (n=canadait@S010600032f2fd7cf.cg.shawcable.net) |
| 03:06.47 | *** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58) |
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| 03:15.36 | jblack | Just takes practice and experience. |
| 03:16.00 | jblack | seems like a nobrainer to say "to setup an asterisk server, one needs to learn how to setup an asterisk server" |
| 03:16.21 | russellb | you must become one with the channelz |
| 03:20.01 | *** join/#asterisk chendy (n=chatzill@119.137.95.53) |
| 03:20.25 | dlynes | Any idea what the notice, 'Failed to authenticate on INVITE' means exactly, and how to correct it? I keep getting it when I'm trying to send a t.38 session from one asterisk 1.6.1.8 box to another asterisk 1.6.1.8 box |
| 03:21.35 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 03:21.41 | dlynes | russellb: btw...please thank Kevin for finally coming out with some fax applications that work great |
| 03:22.09 | dlynes | russellb: asterisk was long overdue for it, and they work just perfectly from what I've experienced so far |
| 03:22.24 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 03:22.49 | Sier | I'm trying to install asterisk.. I'm doing to "make install" @ asterisk-addons, but I'm getting this error: chan_ooh323.c:2842: error: âstruct ast_channelâ has no member named âlockâ |
| 03:22.49 | Sier | make[1]: *** [chan_ooh323.o] Error 1 |
| 03:22.54 | Sier | What can I do to fix this? :) |
| 03:23.13 | dlynes | Sier: have you installed asterisk yet? |
| 03:24.02 | Sier | yes |
| 03:24.14 | dlynes | Sier: did you install it to default locations? |
| 03:24.39 | Sier | yes, all I did was: ./configure, make menuconfig and make install |
| 03:24.55 | dlynes | Sier: which version of asterisk? which version of asterisk-addons? |
| 03:25.16 | Sier | asterisk-addons 1.4.9 and asterisk- 1.6.1.9 |
| 03:25.23 | dlynes | Sier: they're not compatible |
| 03:25.41 | dlynes | Sier: you need to install asterisk 1.6.1.9 and asterisk-addons-1.6.1.1 |
| 03:25.45 | Sier | I need to use addons-1.6. , understood :) |
| 03:25.51 | Sier | sorry about that |
| 03:26.07 | dlynes | Sier: not a problem...you're not the first to make that mistake, and i'm sure you won't be the last |
| 03:27.12 | dlynes | Sier: that being said, you should always try to use the version of libpri, dahdi-linux, dahdi-tools, and addons that were released closest to the time of your asterisk release |
| 03:28.12 | dlynes | Sier: sometimes stuff gets introduced mid-version that might make certain minor versions of supporting libraries/drivers/... incompatible with certain minor versions of asterisk |
| 03:28.29 | Sier | I see.. interesting.. |
| 03:29.20 | mchou | dlynes: did I hear you right, you faxing over internet? |
| 03:30.27 | dlynes | mchou: yes |
| 03:30.40 | Sier | wow, wonderful.. addons compiled just fine! |
| 03:30.42 | dlynes | mchou: working flawlessly too, I might add |
| 03:30.45 | snadge | what does the following mean? There are 1 bad destinations |
| 03:30.52 | snadge | DEST STATUS: EMPTY |
| 03:31.01 | dlynes | snadge: can you put it into context? |
| 03:31.21 | dlynes | mchou: why do you ask? |
| 03:31.24 | snadge | this is in the freepbx admin status page.. of an asterisknow installation |
| 03:31.29 | mchou | dlynes: can yo describe your setup a bit? |
| 03:31.35 | dlynes | snadge: you want to /join #asterisknow |
| 03:31.42 | dlynes | snadge: and /join #freepbx |
| 03:31.45 | dlynes | ~guis |
| 03:31.45 | infobot | guis are too sexy for [TK]D-Fender's shirt.. too sexy .. it hurts! ~ |
| 03:31.48 | dlynes | ~gui |
| 03:31.49 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
| 03:31.54 | snadge | dlynes: already in there :) |
| 03:31.54 | mchou | dlynes: cause I thought faxing over internet reliably was a pipe dream |
| 03:31.56 | dlynes | grrr |
| 03:32.16 | dlynes | mchou: it is, and it isn't |
| 03:32.33 | mchou | dlynes: heisenberg? :) |
| 03:32.35 | dlynes | mchou: trying to get asterisk's t.38 passthrough working properly is a pipe dream |
| 03:32.50 | dlynes | mchou: getting t.38 to work with SendFAX/ReceiveFAX is not |
| 03:33.23 | dlynes | mchou: also, with asterisk 1.6.2, supposedly you're going to be able to use asterisk as a fax gateway |
| 03:33.44 | dlynes | mchou: I've been using call files to send faxes out |
| 03:33.46 | mchou | dlynes: ok, please describe how that's supposed to work |
| 03:34.09 | dlynes | mchou: SendFAX(/var/spool/asterisk/faxes/myfax.tif,z) |
| 03:34.16 | mchou | dlynes: nono |
| 03:34.20 | dlynes | mchou: ? |
| 03:34.52 | mchou | I mean like faxmachine->ata->Asterisk->?? |
| 03:34.59 | Katty | pokes head in |
| 03:35.24 | mchou | dlynes: basically how are you hooking things up |
| 03:36.45 | dlynes | mchou: Oh...for that, I'm doing faxmachine->ata->sip w/ulaw->asterisk/ReceiveFAX, and then doing dial(...,M(macro^${EXTEN}^${LOCALSTATIONID}^/var/spool/asterisk/fax/myfaxfile.tif)) ... SendFAX(/var/spool/asterisk/fax/myfaxfile.tif,z) |
| 03:37.25 | Katty | dlynes: how do your users get the tiff to the server for SendFax()ing |
| 03:37.25 | dlynes | mchou: that's to get around t.38 passthrough not working for me |
| 03:37.37 | dlynes | Katty: a fax machine is one way |
| 03:37.46 | Katty | dlynes: the other? |
| 03:37.53 | dlynes | Katty: the other way I'm going to be doing is a pdf attachment, with the phone number in the subject line |
| 03:38.05 | dlynes | Katty: and i've got a website set up for phone number to email relationship mapping |
| 03:38.11 | Katty | dlynes: are you using a 3rd party thing for this? |
| 03:38.15 | dlynes | Katty: which backends into the asterisk database |
| 03:38.25 | dlynes | Katty: 3rd party thing for what? |
| 03:38.35 | Katty | dlynes: i assumed your Email To Fax bit was 3rd party |
| 03:38.46 | Katty | dlynes: i've been looking for something to do that. |
| 03:38.53 | dlynes | Katty: other than the fax for asterisk drivers from digium, everything else is written by moi |
| 03:39.00 | Katty | applauds dlynes |
| 03:39.19 | Katty | dlynes: so far i have them putting the tiff into a folder (mapped into my computer) with a numeric filename |
| 03:39.21 | dlynes | dialplan written in normal asterisk dialplan code |
| 03:39.35 | Katty | dlynes: they then called the SendFax() on an extension, enter the file name and phone number, which then starts a call file. |
| 03:39.44 | mchou | dlynes: so lemme understand this....you are about to send a fax....you dial a fax number? |
| 03:40.00 | dlynes | website written in php, with the class that's published on voip-info.org, but then modified a bit so that it's not broken, and so that it's fleshed out enough to be useful |
| 03:40.19 | dlynes | and then all the other stuff will be written in perl or bash script or something |
| 03:40.26 | dlynes | mchou: yes |
| 03:40.36 | dlynes | mchou: after it's connected, then i run sendfax |
| 03:41.09 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-10-86.telstraclear.net) |
| 03:41.30 | dlynes | Katty: yeah...for my testing phase i was using a call file |
| 03:41.47 | dlynes | Katty: for my final phase, i'll have two different modes |
| 03:42.02 | dlynes | Katty: the method i'm using now for people with fax machines and an asterisk box on location |
| 03:42.03 | Katty | isn't so great with php or flash. |
| 03:42.11 | Katty | nods |
| 03:42.29 | dlynes | Katty: and another method for people that would rather just email a pdf file, which then gets thrown into a call file |
| 03:42.51 | Katty | what are you using to convert pdf to tiff? |
| 03:42.55 | dlynes | Katty: i need to set up an mx record to handle that, and set the mail user to use maildir, instead of mail spool |
| 03:42.58 | dlynes | ghostscript |
| 03:43.17 | Katty | hmm. neat. i've been looking for something to do that |
| 03:43.22 | Katty | windows has a problem creating a usable tiff. |
| 03:44.24 | Katty | so what do you do with a fax to send it to asterisk? |
| 03:44.28 | Katty | i've never done that before |
| 03:45.30 | dlynes | Katty: I have a fax machine hooked up to a Mediatrix ata, and then I just execute ReceiveFAX(blahblah,z) on it |
| 03:45.56 | dlynes | Katty: then it dumps out a tiff file for me, which I then subsequently send out |
| 03:46.17 | Katty | neat. |
| 03:46.51 | dlynes | Katty: yeah...my coworker was telling me it looks just like a laser printer printout...doesn't even look like a fax apparently |
| 03:47.08 | mchou | dlynes: you using a provider with T.38? |
| 03:47.09 | dlynes | Katty: I can't see the output of the fax machine...the fax machine's in Vancouver, and I'm near Toronto |
| 03:47.15 | Katty | nifty (= |
| 03:47.18 | dlynes | mchou: yes, we support t.38 |
| 03:47.26 | dlynes | mchou: we're our own provider |
| 03:47.29 | mchou | dlynes: who are you :) |
| 03:47.45 | dlynes | mchou: well, and plus we use Navigata (Saskatchewan Telecom) for our upstream |
| 03:48.02 | dlynes | mchou: are you a care home? |
| 03:48.16 | mchou | hmm?? |
| 03:48.22 | mchou | what's a care home? |
| 03:48.24 | dlynes | mchou: all of our customers are care homes :) |
| 03:48.38 | dlynes | mchou: assisted living, respite care, care homes, homes for the aged, ... |
| 03:49.00 | mchou | dlynes: ok, just making sure :) |
| 03:49.03 | dlynes | mchou: about 20 names for more or less the same thing |
| 03:49.24 | dlynes | mchou: it can't possibly be that difficult to find t.38 terminators, is it? |
| 03:49.30 | dlynes | mchou: what country are you in? |
| 03:49.34 | mchou | US |
| 03:50.00 | mchou | dlynes: I havent tried very hard looking |
| 03:50.30 | mchou | I'm sure there are some out there but it's all buried in the very very fine print |
| 03:50.37 | dlynes | beautiful....calltermination.com is down |
| 03:50.45 | dlynes | their database is even down |
| 03:50.45 | mchou | lol |
| 03:50.57 | dlynes | i was gonna look there for your t.38, but i guess not |
| 03:50.58 | mchou | inspires confidence |
| 03:51.30 | dlynes | mchou: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+T.38 |
| 03:51.42 | dlynes | mchou: it was the first hit on a google search |
| 03:51.51 | mchou | dlynes: that's no reliable |
| 03:51.56 | mchou | not* |
| 03:53.17 | dlynes | mchou: http://www.justfuckinggoogleit.com/?q=t.38+termination |
| 03:53.28 | mchou | lol |
| 03:54.08 | mchou | if that's your idea of "reliable" I'd hate see you definition of failure |
| 03:54.13 | mchou | your* |
| 03:55.31 | dlynes | :) |
| 03:59.02 | p3nguin | So does anyone know if Cisco 7940/7960 phones can subscribe to hints to detect InUse status when using the SCCP image? They don't seem to do it using SIP. |
| 04:03.23 | p3nguin | I hate to go through the trouble of changing my phones over to an SCCP image if they still won't detect presence once I do it. |
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| 04:12.28 | *** part/#asterisk canadait (n=canadait@S010600032f2fd7cf.cg.shawcable.net) |
| 04:15.38 | *** join/#asterisk wtca (n=wtca@williamt.vpn.sonic.net) |
| 04:15.42 | *** part/#asterisk wtca (n=wtca@williamt.vpn.sonic.net) |
| 04:16.57 | *** join/#asterisk Trupsalms (n=shawndel@adsl-68-20-35-251.dsl.chcgil.ameritech.net) |
| 04:17.04 | Trupsalms | hello room |
| 04:17.16 | Trupsalms | may i ask for some small assistance |
| 04:19.51 | *** join/#asterisk dan__t (i=vpn@vpn.withparity.net) |
| 04:20.43 | dan__t | Hi. |
| 04:21.14 | florz | no, this channel is exclusively for being silent |
| 04:21.15 | florz | oops |
| 04:21.24 | florz | sorry |
| 04:21.29 | dan__t | oic. |
| 04:21.32 | florz | really, I didn't wanna say anything |
| 04:21.41 | florz | I'm so sorry |
| 04:21.43 | florz | seriously |
| 04:21.50 | florz | hi dan__t ;-) |
| 04:21.59 | florz | that wasn't addressed to you ;-) |
| 04:22.05 | dan__t | Hello. |
| 04:22.39 | keith4_ | wow. open mouth, insert foot |
| 04:22.43 | Trupsalms | i guess that was for me huh |
| 04:22.46 | dan__t | haha |
| 04:23.37 | dan__t | Trying to do some hackery on a bridge.... what I'm looking to do is to, well assuming one of those channels is a moderator/privileged/whatever, I want another channel in there that is (reverse?) muted so that it can only listen for data or whatever from that privileged channel - regardless of how many others are in the bridge. |
| 04:25.09 | dan__t | Loaded question huh |
| 04:26.59 | dan__t | I was hoping I could use AMI to manipulate this. |
| 04:28.03 | dlynes | Trupsalms: instead of asking if you can get help....start by asking your real question |
| 04:29.53 | Trupsalms | don't know if it is supported, been to a few other channels that lead back here |
| 04:30.14 | dlynes | Trupsalms: just state your question |
| 04:32.25 | p3nguin | Asking to ask is not supported, if that's what you meant. |
| 04:34.27 | Trupsalms | i have compiled asterisk so far on ubuntu desktop, i beleive that it is configured correctly, read on a forum about installing a asterisk-gui on it as well, have did that also, rann the makeconfigcheck, and everything is ok, but when trying to coonect to the localhost on port 8088, i get firefox can not open page at localhost:8088 |
| 04:35.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:35.08 | p3nguin | ~asterisk-gui |
| 04:35.09 | infobot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0. For support go to #asterisk-gui |
| 04:36.10 | dan__t | ChanSpy might be interesting... |
| 04:40.02 | dlynes | Trupsalms: you probably missed a configuration step for asterisk-gui...however, that being said...95% of the people in this channel don't use a gui |
| 04:40.15 | Trupsalms | i have been to that channel was pointed back here and it mostly empty |
| 04:40.53 | Trupsalms | i'm crying because the cammand on everything is just to much to remember, lol |
| 04:41.01 | Trupsalms | seroiusly though |
| 04:41.25 | Trupsalms | if you could point me in a very good direction other than asterisk-gui |
| 04:42.20 | snadge | know the feeling bro.. i'm using freepbx myself, as it comes with asterisknow |
| 04:42.56 | snadge | but i managed to get an ip phone to log into it.. and its connected to my isps voip trunk.. whoopeedoo ;) |
| 04:43.15 | snadge | but ekiga is a pile of crap.. it just keep scrashing |
| 04:43.41 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:58a5:7f49:1128:cdc4) |
| 04:44.48 | Trupsalms | any advice on a channel other than asterisk-gui, that may provide good support |
| 04:53.09 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 04:55.18 | snadge | awesome.. i've got it to say.. "All circuits are busy now, please try again later" ;) |
| 04:55.33 | snadge | when i try to place a call.. but thats an excellent start hehe |
| 04:57.48 | *** join/#asterisk jermudgeon (n=jhaustin@216-67-61-242.static.acsalaska.net) |
| 05:09.03 | dlynes | anyone run across the issue of 'Failed to authenticate on INVITE to '"blahblah" <sip:blahblah@domain.com>;tag=ryp9ayr3'? |
| 05:11.47 | dlynes | bleh |
| 05:11.52 | dlynes | finally got that problem solved |
| 05:12.02 | dlynes | and now the stupid thing hangs up as soon as it tries dialing |
| 05:17.05 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 05:21.09 | *** join/#asterisk voipmonk (n=voipmonk@69.172.97.20) |
| 05:23.12 | Trupsalms | voipmonk are you busy |
| 05:24.17 | voipmonk | ? |
| 05:24.40 | Trupsalms | could i pm u |
| 05:24.45 | voipmonk | yes |
| 05:32.12 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 05:33.46 | ChannelZ | hah burn |
| 05:47.29 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:58a5:7f49:1128:cdc4) |
| 05:57.09 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 06:05.49 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:10.02 | hardwire | pm = private massage? |
| 06:10.04 | hardwire | awesome. |
| 06:10.18 | hardwire | bow chika bow wow |
| 06:22.17 | dan__t | hrm... |
| 06:33.07 | *** join/#asterisk Kpcto (n=kvirc@77.239.64.34) |
| 06:33.57 | dan__t | http://pastie.org/696867 - Is that indicative of a peer problem or a host problem? |
| 06:37.50 | dan__t | Started after I rebooted this machine. |
| 06:43.07 | dan__t | nm, got it. |
| 06:46.42 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 06:48.09 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 06:51.25 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:58a5:7f49:1128:cdc4) |
| 06:55.07 | *** join/#asterisk alecdavis (n=alec@202-78-149-14.cable.telstraclear.net) |
| 06:59.27 | *** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 07:01.00 | *** join/#asterisk admin0 (n=admin0@bb220-255-83-167.singnet.com.sg) |
| 07:01.08 | admin0 | hi .. can I register as a IAX user and send to a SIP trunk ? |
| 07:01.17 | admin0 | or does it need some extra transcoding setting in between ? |
| 07:02.47 | [TK]D-Fender | admin0: * does it automatically. |
| 07:02.52 | admin0 | ok |
| 07:02.53 | admin0 | thanks |
| 07:03.30 | admin0 | when sending from iax -> iax i get this error: requested/capability 0x8/0x41c incompatible with our capability 0x101. .. what does this mean ? |
| 07:04.06 | [TK]D-Fender | admin0: codec disagreement |
| 07:04.17 | admin0 | ok |
| 07:04.24 | admin0 | core show codecs does not list what 0x101 mean |
| 07:04.52 | [TK]D-Fender | admin0: Syou should know what the other side supports and have chosen accordingly |
| 07:05.52 | *** join/#asterisk jermudgeon_ (n=jermudge@216.67.61.242) |
| 07:11.49 | admin0 | the other side has g723... my peer has disallow=all |
| 07:11.50 | admin0 | <PROTECTED> |
| 07:12.30 | admin0 | how do I know what 0x101 is |
| 07:12.38 | admin0 | and what 0x8 and 0x41c means |
| 07:13.45 | [TK]D-Fender | admin0: Do you have a TC400? |
| 07:14.40 | admin0 | this device is svg200sp from stephen-tele.com |
| 07:15.37 | kaldemar | admin0: 0x8 is alaw, iirc |
| 07:15.48 | [TK]D-Fender | admin0: The TC400P card is the only legal way tog et G.723 onto *. And for G.729... thats also licensed. |
| 07:15.51 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 07:16.04 | [TK]D-Fender | admin0: So right now you are offering things * doesn't support "stock" |
| 07:16.19 | [TK]D-Fender | admin0: And by agreeing to try it you will be met with failure. |
| 07:16.23 | kaldemar | admin0: actually, you'll see the hex values for codecs in the table that core show codecs prints out |
| 07:18.30 | admin0 | guys .. thanks .. i figured it out |
| 07:18.47 | admin0 | passes pizza to [TK]D-Fender and kaldemar |
| 07:19.04 | [TK]D-Fender | checkout time.... |
| 07:19.06 | [TK]D-Fender | later all |
| 07:26.51 | *** join/#asterisk xrmx__ (n=rm@host59-59-dynamic.14-87-r.retail.telecomitalia.it) |
| 07:29.09 | Tech_Travis | Is it possible on a stock 1.6 install to forward a queue member's calls to their cell phone instead of their soft/hard phone? |
| 07:30.04 | *** join/#asterisk Malkor (n=marco@hlle-d9ba01f5.pool.mediaWays.net) |
| 07:30.22 | kaldemar | Tech_Travis: sure |
| 07:31.32 | Tech_Travis | kaldemar: Is it with a built-in function or is that something I would need to create on my own? |
| 07:32.31 | kaldemar | you could try by setting forwarding on in the phone itself or by using a Local/... as the member and doing it somehow in the dialplan. |
| 07:35.31 | *** join/#asterisk mumtazah (n=anees@203.82.91.103) |
| 07:36.23 | Tech_Travis | Thanks. I don't think our softphones (XLite free) support that feature so I was trying to write it all in the dialplan. I am not familiar with Local, I'll take a look at what it does. |
| 07:37.06 | kaldemar | it's a channel type that calls an extension in your dialplan. then you can do whatever you want with the call. Local/exten@context |
| 07:37.53 | snadge | i cant figure out why if you specify an incorrect registrar line.. asterisk uses up all avialable memory and dies |
| 07:38.48 | snadge | 07316xxxxx@iinetphone.iinet.net.au:xxxxxxxx:0731609026@iinetout/0731609026 |
| 07:39.36 | snadge | if i drop everything after and including the second @.. it will at least start.. but it wont register |
| 07:40.50 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 07:42.29 | |Cybex| | Hi, I am running Asterisk 1.4.25 (and using Polycom phones). I would like to manipulate 1xx responses with Asterisk (the To: field). Is this possible? |
| 07:44.16 | mchou | |Cybex|: I doubt it |
| 07:44.39 | mchou | |Cybex|: for this kind of stuff look at opensips |
| 07:44.54 | mchou | or openser |
| 07:44.55 | |Cybex| | Hmm, because Polycom implemented a new feature in their phones: 49465: Update Destination of outbound call based on display-name in SIP To header responses |
| 07:45.10 | |Cybex| | So I was hoping to FINALLY see who I am calling when I dial a number... |
| 07:45.25 | |Cybex| | I'll take a look at it though, thanks |
| 07:45.29 | mchou | what? |
| 07:45.34 | |Cybex| | opensips |
| 07:45.37 | mchou | I'm not sure I understand |
| 07:45.41 | |Cybex| | I'll explain |
| 07:46.13 | mchou | what exactly does the polycom feature do? |
| 07:47.33 | |Cybex| | one sec, phone |
| 07:48.23 | |Cybex| | sorry, back |
| 07:48.35 | |Cybex| | We have Polycom phones here (with a display). |
| 07:48.47 | mchou | yes, I gathered that :) |
| 07:48.48 | |Cybex| | When I dial a number, for example 700, I see 700 in the display |
| 07:49.11 | |Cybex| | I would like to see the NAME of the person I am calling. So when I dial 700, the phone should display "John Doe" instead of 700 |
| 07:49.16 | kaldemar | openser is no more. it got forked into kamailio and opensips. |
| 07:49.16 | mchou | 700 get mapped to user@somwhere |
| 07:49.22 | |Cybex| | I hope I am explaining it correctly? |
| 07:49.41 | |Cybex| | correct |
| 07:49.47 | mchou | yes, then you see "User" on the polycom display |
| 07:49.49 | mchou | :) |
| 07:50.55 | |Cybex| | Hmm, I just checked. username = 700 :( |
| 07:51.07 | |Cybex| | I can't show "fullname" in the display? |
| 07:51.40 | mchou | full name doesnt even make sense (from sip perspective) |
| 07:52.00 | |Cybex| | I'm starting to understand |
| 07:52.05 | mchou | you're dialing sip:700@somewhere |
| 07:52.09 | |Cybex| | So what I need is this in users.conf: |
| 07:52.15 | |Cybex| | username="John Doe" |
| 07:52.28 | Tech_Travis | kaldemar: If I write this into a dialplan, how would I be able to keep track of which extension the queue "assigned" the call to? I tried to use ${EXTEN} but that didn't keep the extension info. after the call left the queue. |
| 07:52.58 | kaldemar | |Cybex|: there is no such parameter in users.conf afaik |
| 07:53.13 | mchou | |Cybex|: it's now defaultuser :) |
| 07:53.45 | |Cybex| | This is in my users.conf: http://pastebin.com/m7ed219a7 |
| 07:54.39 | |Cybex| | What should I do to get the name "John Doe" in the display? So when I call extension 700 I would like to see the name instead of the number. |
| 07:55.02 | kaldemar | Tech_Travis: what exactly did you do? |
| 07:55.15 | mchou | |Cybex|: this is the wrong way to approach it |
| 07:55.20 | kaldemar | |Cybex|: you can't |
| 07:55.24 | |Cybex| | ouch |
| 07:56.06 | mchou | I dont think that polycom feature is what you think it does |
| 07:56.07 | |Cybex| | mchou, what's wrong with it? Can you point me in the right direction... |
| 07:56.12 | |Cybex| | oh |
| 07:56.54 | |Cybex| | So what I want (actually see the name I am calling), is not possible? |
| 07:57.37 | kaldemar | not the way you're trying to do it. |
| 07:57.45 | mchou | I think it's might be possible, but not in the way you're thinking |
| 07:57.55 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 07:58.05 | *** join/#asterisk blinkiz (n=blinkiz@unaffiliated/blinkiz) |
| 07:58.13 | |Cybex| | Hmm, I am wondering what I am doing wrong... |
| 07:58.34 | mchou | you arent doing anything "wrong" |
| 07:58.50 | blinkiz | Hi. I need help. No sound what so ever is played on my pbx. What can be wrong? Not even the ringing tone is working. |
| 07:58.56 | blinkiz | How can I troubleshoot this? |
| 08:00.04 | blinkiz | I take that back about the ringing tone. But the other sounds are not working |
| 08:00.16 | |Cybex| | mchou, Could you give me some hints on how to accomplish this? |
| 08:00.52 | kaldemar | blinkiz: what technology are you using? where is to call coming from and where is it going? |
| 08:01.14 | snadge | i still dont get why my register string causes the asterisk process to allocate all available memory and crash |
| 08:01.27 | snadge | that seems like a bug to me |
| 08:02.06 | kaldemar | snadge: search https://issues.asterisk.org for a similar bug, and if none exist, report one. |
| 08:02.39 | mchou | |Cybex|: the proper way is youse your phone book :) |
| 08:02.45 | mchou | use* |
| 08:02.52 | mchou | like a cell phone |
| 08:02.53 | |Cybex| | Phonebook on the phone? |
| 08:03.07 | |Cybex| | Okay, I understand |
| 08:03.19 | kaldemar | once upon a time, snoms had the ability to receive text on the screen. you could use sipsak from dialplan to do it. |
| 08:03.20 | mchou | either on the phone or in LDAP |
| 08:03.43 | blinkiz | kaldemar, See my core debug 5 here: http://blinkiz.pastebin.com/d662ce7db |
| 08:03.48 | |Cybex| | I'll go for LDAP then |
| 08:04.06 | snadge | my asterisk box is behind a NAT router.. do i need to forward a port through to my asterisk box? 5060? tcp/udp? |
| 08:04.09 | |Cybex| | Thank you for your assistance mchou |
| 08:04.13 | blinkiz | kaldemar, The line, Playback("SIP/outgoing-b6500040", "custom/female_welcome"), is not played |
| 08:04.18 | snadge | i have set nat=yes in my trunk settings |
| 08:05.04 | blinkiz | female_welcome.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 08:05.26 | *** join/#asterisk bbt (n=sam@180.189.138.39) |
| 08:05.48 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 08:06.12 | Tech_Travis | I created an on-call queue, where the person who is going to be on call dials an internal extension number. I used VMAuthenticate to get their extension number and used AddQueueMember to put them into the queue. I set a 30 second timeout so if the call isn't answered it goes to a Dial(SIP/${EXTEN} + 10000) statement where 10000 plus their extension is equal to their cell-phone number. In the asterisk console the extension that |
| 08:06.13 | kaldemar | blinkiz: you'll probably find better help in #freepbx. |
| 08:06.18 | blinkiz | kaldemar, Using Asterisk 1.6.2 on Ubuntu 9.10 (kernel 2.6.31-14). |
| 08:06.51 | blinkiz | kaldemar, People in #freepbx doesn't know anything.. |
| 08:07.04 | kaldemar | blinkiz: and people in here, well... |
| 08:07.08 | kaldemar | ~freepbx |
| 08:07.08 | 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 |
| 08:07.20 | blinkiz | kaldemar, okay... |
| 08:07.38 | kaldemar | blinkiz: but that cli output of yours tells nothing about the problem |
| 08:07.46 | kaldemar | blinkiz: is nat involved somehow? |
| 08:07.49 | blinkiz | kaldemar, But the problem is not the dialplan. I gave you the Playback() line.. Its more a "what directory" or "what permission" problem |
| 08:08.04 | kaldemar | blinkiz: what makes you think that? |
| 08:08.09 | blinkiz | nat is not involved. |
| 08:08.44 | *** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 08:08.45 | blinkiz | kaldemar, Well, the dialplan clearly tries to play a sound file according to the core debug. So that rules out problem with the dialplan. Right? |
| 08:08.55 | kaldemar | ~sipnat |
| 08:08.56 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 08:09.02 | kaldemar | snadge: ^ look at that |
| 08:09.48 | kaldemar | blinkiz: it seems to try to play it. |
| 08:10.29 | mchou | blinkiz: this is easy |
| 08:10.57 | mchou | blinkiz: copy a sound file that came with asterisk into custom |
| 08:11.14 | mchou | change dialplan accordingly |
| 08:11.41 | mchou | if it plays you know it's a file encoding issue |
| 08:12.03 | mchou | if not, perms or some other issue |
| 08:12.57 | mchou | and copy all formats like pcmu or whatever |
| 08:13.05 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:13.06 | blinkiz | mchou, Nice troubleshooting thing. Gonna try that |
| 08:13.38 | blinkiz | mchou, Does this include the ringing tone? |
| 08:13.45 | mchou | huh?? |
| 08:13.51 | blinkiz | mchou, because that is the only thing I have heard working |
| 08:14.29 | mchou | any file that came with asterisk should playback |
| 08:14.53 | mchou | i.e be properly encoded already |
| 08:15.15 | kaldemar | blinkiz: the ringing tone does not come from asterisk in that case. |
| 08:15.39 | blinkiz | kaldemar, I mean when I call into the asterisk box and get a ringtone. That is from asterisk alright.. |
| 08:16.01 | *** join/#asterisk oej (n=olle@p5099839d.dip0.t-ipconnect.de) |
| 08:16.01 | kaldemar | blinkiz: no it's not unless you force asterisk to send the tone. |
| 08:16.11 | blinkiz | kaldemar, Hmm, okay |
| 08:16.13 | kaldemar | blinkiz: if you'd use an analog phone, then yes. |
| 08:16.31 | blinkiz | kaldemar, So then no files is really playing.. |
| 08:16.33 | kaldemar | unless you mean a DISA |
| 08:16.35 | mchou | not even an analog phone |
| 08:16.36 | blinkiz | Not even asterisk sounds |
| 08:16.44 | mchou | might be his ata :) |
| 08:16.45 | kaldemar | blinkiz: the ring tone has nothing to do with files playing |
| 08:23.02 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 08:25.24 | *** join/#asterisk Zhad (n=tom@host-1.art-it-services.co.uk) |
| 08:34.55 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:44.36 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:65b6:dfe3:d77b:babf) |
| 08:45.33 | mchou | kaldemar: I have an agi that takes about 2-3 secs to complete. Is there a way to play a ringback tone while the agi is in progress? |
| 08:45.56 | mchou | I mean from the dialplan |
| 08:46.29 | mchou | not from within the agi |
| 08:47.41 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:48.17 | kaldemar | mchou: core show application PlayTones |
| 08:49.44 | mchou | so if I call PlayTones beforce AGI gets called, playtones will stop when agi finishes execution? |
| 08:50.13 | kaldemar | you can stop the tone with StopPlayTones if nothing in the AGI stops it. |
| 08:50.41 | mchou | ahh, ok |
| 08:51.29 | mchou | cool |
| 08:57.18 | *** part/#asterisk mumtazah (n=anees@203.82.91.103) |
| 09:10.50 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.104) |
| 09:14.44 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:15.49 | *** join/#asterisk Da-Geek (n=Da-Geek@ip-217-204-101-67.easynet.co.uk) |
| 09:16.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 09:19.42 | *** join/#asterisk Hunner (i=hunner@firefly.cat.pdx.edu) |
| 09:21.09 | Hunner | Hi, will I be able to register to a peer if they show up as "UNREACHABLE", or do they have to show up as "OK" before registration can take place? |
| 09:21.55 | kaldemar | it has nothing to do with being able to register |
| 09:22.05 | Hunner | I'm having a tough time here in india as I am unsure if it's my config that's messed up or if it's airtel blocking me (they block a lot of the popular voip providers and may filter all port 5060 traffic) |
| 09:23.15 | *** join/#asterisk oej (n=olle@p5099839d.dip0.t-ipconnect.de) |
| 09:23.30 | Hunner | k. Registration with every provider I've tried has timed out... but it might also be my nat that's the problem. My grandstream ATA registered to my *-box fine internally |
| 09:24.43 | kaldemar | Hunner: enable sip debug in the asterisk box and you'll see if the box gets any messages |
| 09:24.59 | kaldemar | so the asterisk box is behind a NAT? |
| 09:26.10 | Hunner | yes, and I've given it my local nets and external ip |
| 09:26.18 | Hunner | enables debug to see |
| 09:36.02 | baijum | Can I install two asterisk servers in same network, say 192.168.1.0/24 |
| 09:36.47 | baijum | ? |
| 09:40.21 | kaldemar | of course |
| 09:44.15 | baijum | kaldemar: ok, thanks |
| 09:47.05 | Tech_Travis | kaldemar: Is it possible to do the "on-call" queue to ring their soft/hard phone, then if no answer to ring a cell number with the Local/ that you mentioned earlier, or do I need a different approach? |
| 09:50.30 | kaldemar | Tech_Travis: you can set a timeout for the queue so that the caller jumps out of it and then proceed in the dialplan |
| 09:55.29 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 09:55.34 | Tech_Travis | I've gotten the queue to timeout and go to the next step, but I'm having trouble with writing a dynamic way to assign the next step. I've been trying to use a dialgroup which is defined by QUEUE_MEMBER_LIST. |
| 09:59.33 | *** part/#asterisk mumtazah (n=mumtazah@203.82.91.104) |
| 10:08.54 | *** join/#asterisk sgimeno (n=sgimeno@80.33.64.226) |
| 10:09.11 | *** part/#asterisk alecdavis (n=alec@202-78-149-14.cable.telstraclear.net) |
| 10:10.01 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 10:10.05 | DelphiWorld | hi all |
| 10:10.11 | DelphiWorld | any change to the IAX2 protocol? |
| 10:10.15 | DelphiWorld | i think is iax3;) |
| 10:12.02 | Hunner | how can I give a different port to connect on for a peer in sip.conf? |
| 10:12.47 | Hunner | oh,looks like just port= |
| 10:17.42 | *** join/#asterisk fish9370 (n=fish9370@79.120.100.132) |
| 10:19.54 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 10:23.23 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:27.23 | *** join/#asterisk af_ (n=getsmart@88-149-240-33.dynamic.ngi.it) |
| 10:39.54 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 10:41.28 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 10:41.57 | *** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com) |
| 10:43.31 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 10:43.46 | kaldemar | Hunner: you can't. the port option for a peer is for outgoing. |
| 10:48.00 | TJNII | Hunner: Why would you want to? You're probably trying to solve a problem using the wrong tool. |
| 10:55.57 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 11:08.33 | Hunner | TJNII: I was trying to see if it's my provider filtering port 5060 or if it's my own problem |
| 11:09.00 | Hunner | I know my provider blocks gizmo5 and sipgate, but I think they're not blocking 5060 by now |
| 11:09.09 | Hunner | But I still can't figure out what the problem is :( |
| 11:12.40 | *** join/#asterisk Cara_Magro (i=0@189.125.173.149) |
| 11:16.36 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:16.43 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 11:21.57 | TJNII | Well, you can easily move your server off 5060. However, you will still need to register to 5060 which the ISP will block. You don't have thwe power to change that. |
| 11:22.20 | TJNII | Get a better ISP. The last two I used both used the fact that they offer QoS for SIP as a selling point. |
| 11:22.27 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:814f:8e04:90f6:90d1) |
| 11:26.41 | *** join/#asterisk jawad (n=jawad@nijmegen.rootnet.nl) |
| 11:26.46 | jawad | Hello i need help |
| 11:26.48 | jawad | with asterisk |
| 11:27.07 | jawad | can somebody help me with this |
| 11:27.25 | jawad | I think its a small problem not difficult |
| 11:27.48 | Hunner | TJNII: I just set up a locally forwarded port 5059 through ssh to sipgate.com:5060, then told asterisk to register to 127.0.0.1:5059 |
| 11:28.06 | Hunner | so it comes out on the ssh endpoint... it still didn't work though |
| 11:28.10 | jawad | I`ve put a custom sound for voice-mails audio-label. After the audio-label was finished playing, another audio-sound was triggerd. |
| 11:28.10 | jawad | "if this is correct press" |
| 11:28.10 | jawad | How can I disable this function in Asterisk. |
| 11:28.11 | jawad | -- Executing [vmblast@app-vmblast:2] ExecIf("SIP/01-0243ae80", "1|Background|custom/spreekuwnaamin") in new stack |
| 11:28.13 | jawad | -- <SIP/01-0243ae80> Playing 'custom/spreekuwnaamin' (language 'en') |
| 11:28.15 | jawad | -- Executing [vmblast@app-vmblast:3] BackGround("SIP/01-0243ae80", "if-correct-press&digits/1") in new stack |
| 11:28.17 | Hunner | Yeah. better ISP would be awesome |
| 11:28.18 | jawad | -- <SIP/01-0243ae80> Playing 'if-correct-press' (language 'en') |
| 11:28.20 | jawad | -- <SIP/01-0243ae80> Playing 'digits/1' (language 'en') |
| 11:28.22 | jawad | -- Executing [vmblast@app-vmblast:4] WaitExten("SIP/01-0243ae80", "20|") in new stack |
| 11:28.24 | jawad | == Spawn extension (app-vmblast, vmblast, 4) exited non-zero on 'SIP/01-0243ae80' |
| 11:28.26 | jawad | I bolted the function I want to disable. |
| 11:28.28 | kaldemar | ~pb |
| 11:28.29 | 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 |
| 11:28.29 | jawad | What config file should I need to make this possible. |
| 11:28.41 | kaldemar | jawad: don't spam here |
| 11:28.44 | jawad | sorry |
| 11:28.50 | TJNII | What was ssh connected to on the other end? |
| 11:29.00 | TJNII | Where was the server? |
| 11:29.20 | Hunner | in the US |
| 11:29.26 | jawad | But I need help |
| 11:29.36 | jawad | with my voicemail menu |
| 11:29.49 | kaldemar | jawad: remove the line in the dialplan |
| 11:29.58 | *** join/#asterisk Tim_Toady (n=moi@adsl119-125.kln.forthnet.gr) |
| 11:30.10 | jawad | ok |
| 11:30.38 | Hunner | TJNII: also portscanning 5060 shows 'filtered' which I think is okay. If airtel was blocking it then I think it would show closed |
| 11:31.13 | jawad | uh what dailplan |
| 11:32.00 | kaldemar | Hunner: it says filtered when a firewall is blocking the port. are you scanning the UDP port, btw? |
| 11:32.33 | Hunner | ... right. I was using tcp |
| 11:33.26 | jawad | Klademar: I cant remove the line |
| 11:33.50 | jawad | Kaldemar: I want too disable the built-in sound if this is correct.gsm |
| 11:35.07 | Hunner | kaldemar: 5060/udp open|filtered # how about that from nmap? |
| 11:35.29 | kaldemar | jawad: why can't you remove the line? |
| 11:35.48 | jawad | asterisk -r |
| 11:36.02 | jawad | it only shows what i does i cant change anything |
| 11:36.07 | kaldemar | Hunner: that indicates it's open |
| 11:36.23 | kaldemar | jawad: extensions.conf has your dialplan. edit it. |
| 11:36.28 | jawad | ok |
| 11:36.50 | kaldemar | but don't just remove the line if you have zero knowledge in dialplans |
| 11:37.01 | jawad | ok |
| 11:37.03 | Hunner | sees no response with sip set debug |
| 11:38.05 | kaldemar | jawad: pastebin the [app-vmblast] context and i'll tell you how to edit |
| 11:40.25 | TJNII | Hunner: Your * box is trying to register and getting no response? |
| 11:41.08 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 11:41.16 | jawad | ok |
| 11:41.31 | jawad | so i need to search in app-vmblast in the extension.conf |
| 11:44.08 | kaldemar | what GUI are you using? |
| 11:44.20 | Hunner | TJNII: yep. I'm behind nat, I've added my external ip and local nets, added a register line and a [section] to sip.conf, and it says UNREACHABLE and that registration timed out, and debug shows sending register requests and no respose |
| 11:44.20 | jawad | elastix |
| 11:45.00 | kaldemar | so freepbx. |
| 11:45.19 | jawad | yes |
| 11:45.28 | jawad | freepbx and elastix |
| 11:46.01 | fish9370 | trixbox |
| 11:46.28 | TJNII | Hunner: Hopefully a stupid question, but you can access the public internet from that box, correct? |
| 11:47.03 | kaldemar | jawad: it's likely that you won't find the context in extensions.conf, but some other file. try asking in #freepbx first. |
| 11:47.12 | TJNII | ~freepbx |
| 11:47.13 | 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 |
| 11:57.21 | jawad | found it kaldemar |
| 11:57.22 | jawad | its |
| 11:57.35 | jawad | extensions_additional.conf |
| 11:57.44 | jawad | i found vmblast |
| 11:57.48 | jawad | app-vmblast |
| 11:58.06 | Hunner | TJNII: yes, just fine |
| 11:59.58 | TJNII | Hunner: Well, fire up your wireshark, I guess. |
| 12:00.04 | TJNII | is going to bed |
| 12:00.20 | jawad | so what now kaldermar |
| 12:01.15 | Hunner | TJNII: thanks, night |
| 12:02.32 | kaldemar | jawad: get rid of the line that plays the unwanted sound |
| 12:02.33 | infernix | has anyone used the Skype for Asterisk channel? |
| 12:03.48 | jawad | I did but it stil play |
| 12:04.02 | kaldemar | jawad: you need the reload the dialplan |
| 12:04.20 | infernix | there's no trial or demo and very little info on how it works (eg how do you call a skype username) |
| 12:04.22 | kaldemar | jawad: asterisk -vvvr and dialplan reload |
| 12:04.26 | jawad | ok |
| 12:05.06 | kaldemar | infernix: i'd ask digium directly |
| 12:11.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 12:12.04 | *** join/#asterisk fofware (n=chatzill@host93.190-229-255.telecom.net.ar) |
| 12:14.45 | jawad | hmm doenst work |
| 12:18.05 | *** join/#asterisk Chesther (n=cam2@cpe-67-241-8-206.twcny.res.rr.com) |
| 12:20.20 | infernix | kaldemar: i guess i'll do that when they wake up |
| 12:28.28 | *** part/#asterisk blinkiz (n=blinkiz@unaffiliated/blinkiz) |
| 12:29.16 | *** join/#asterisk tokozedg (n=Tornike@mail.orient-logic.com) |
| 12:34.27 | *** join/#asterisk albertoandrade (n=albertoa@189.4.53.18) |
| 12:59.04 | *** join/#asterisk Chesther (n=cam2@cpe-67-241-8-206.twcny.res.rr.com) |
| 13:02.12 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:02.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:05.34 | jawad | I have one more question |
| 13:09.41 | kaldemar | just ask |
| 13:10.07 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:10.51 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 13:20.57 | jawad | hey |
| 13:21.00 | jawad | i want |
| 13:21.08 | jawad | it works btw thanks alot! |
| 13:21.27 | jawad | Ive deleted the line |
| 13:21.36 | jawad | now you cant hear it anymore |
| 13:21.46 | jawad | but now the phone is is hanging up |
| 13:22.00 | jawad | thats not good |
| 13:22.12 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 13:22.13 | jawad | i want to play the voice-mail anoucement |
| 13:22.19 | *** join/#asterisk superbeef (n=superbee@74.84.194.4) |
| 13:22.24 | jawad | and then trigger the beep |
| 13:22.39 | jawad | and the stop the recording if hangup |
| 13:22.40 | kaldemar | pastebin the context |
| 13:22.54 | jawad | pastebin? |
| 13:22.58 | jawad | show it? |
| 13:23.06 | kaldemar | ~pb |
| 13:23.07 | 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 |
| 13:26.51 | jawad | ok ill install it |
| 13:27.03 | kaldemar | you don't have to install anything. those are web pages. |
| 13:28.14 | jawad | http://pastebin.ca/1669852 |
| 13:28.32 | jawad | ok i did |
| 13:29.02 | jawad | i want after the anoucement a beep and then if ppl are finished with recording the voicemail to hang up |
| 13:31.33 | *** join/#asterisk ManxPower-work (n=EWieling@82.sub-70-222-194.myvzw.com) |
| 13:34.20 | kaldemar | jawad: you can remove this: exten => vmblast,n,Playback() , it does nothing |
| 13:35.11 | kaldemar | looks like you want something that is really different from that. you need to learn about dialplans before you can do that. |
| 13:35.16 | kaldemar | ~book |
| 13:35.17 | 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 |
| 13:35.37 | kaldemar | jawad: grab the pdf above and learn some |
| 13:35.40 | jawad | ok i did |
| 13:38.20 | jawad | do you have no clue? |
| 13:39.32 | *** part/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 13:41.41 | kaldemar | yes i have a clue but i don't want to modify something that should be thrown away and replaced. |
| 13:42.38 | jawad | what do you mean? |
| 13:43.10 | ManxPower-work | jawad: he means that if he does it for you then you'll never learn anything. |
| 13:43.44 | ManxPower-work | We really don't like people that won't try to learn on their own. |
| 13:43.55 | jawad | Im this close |
| 13:44.06 | jawad | I did everything |
| 13:44.21 | jawad | but I cant understand the voice-mail option |
| 13:44.45 | ManxPower-work | kaldemar's comments seem to indicate that you have a basic misunderstanding of dialplan. |
| 13:44.58 | ManxPower-work | what is the URL for your pastebin? |
| 13:45.21 | jawad | http://pastebin.ca/1669852 |
| 13:45.57 | ManxPower-work | ~freepbx |
| 13:45.58 | 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 |
| 13:46.06 | ManxPower-work | Sorry, I don't help people running FreePBX |
| 13:46.35 | jawad | lol u hate freePBX? |
| 13:47.16 | ManxPower-work | jawad: As infobot said, FreePBX's config files are so complex it's firtually impossible to troubleshoot by hand |
| 13:47.23 | [TK]D-Fender | jawad: You should not be playing around in extensions.conf with that |
| 13:47.39 | jawad | ok |
| 13:47.46 | [TK]D-Fender | jawad: the dialplan is probably just fine. Your configuration on the other hand is something else |
| 13:47.56 | [TK]D-Fender | jawad: Do you not see the big print at the top? |
| 13:48.04 | jawad | yes |
| 13:48.08 | [TK]D-Fender | jawad: ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; |
| 13:48.18 | jawad | ok |
| 13:48.26 | ManxPower-work | If you modify that file it will be overwritten. |
| 13:48.27 | jawad | I will change it back to default |
| 13:48.31 | [TK]D-Fender | jawad: So go learn how to configure you system for their channel -.. #freepbx |
| 13:49.20 | *** part/#asterisk jawad (n=jawad@nijmegen.rootnet.nl) |
| 13:50.09 | ManxPower-work | [TK]D-Fender: I needed to set up a extension w/just voicemail the other day. Took me 30 seconds to do by hand,30 mins to figure it out in the FreePBX GUI and I still didn't get it right. |
| 13:50.39 | [TK]D-Fender | ManxPower-work: Know the feeling.... GUI's are just so God-aweful slow... |
| 13:50.41 | ManxPower-work | "But GUIs are easier!" "Stop drinking the Microsoft Kool-Aid" |
| 13:51.06 | Carlos_PHX | Just depends on your priorities and needs. |
| 13:51.16 | Carlos_PHX | And how much of an expert you want to be. |
| 13:51.34 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:55.14 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:55.24 | Pan3D | expert |
| 13:57.01 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:59.38 | kaldemar | heh, you have him the treatment |
| 13:59.46 | kaldemar | s/have/gave/ |
| 13:59.47 | *** join/#asterisk jch2os (n=jch2os@h2os.com) |
| 14:01.47 | Katty | stretches |
| 14:01.58 | leifmadsen | anti-stretches |
| 14:05.02 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 14:05.21 | [TK]D-Fender | slams leifmadsen & Katty together and watches them anihilate each other producing massive amounts of light and energy |
| 14:05.47 | Katty | :< |
| 14:06.18 | leifmadsen | eep |
| 14:06.30 | jch2os | if I have a linksys router with tomato loaded on it doing QOS, where on my network do I put that. Do I have the internet-->cable modem-->router-->linux firewall-->switch? |
| 14:06.37 | Katty | that was not very nice. |
| 14:06.48 | jch2os | or do I put it between the firewall and switch? |
| 14:08.02 | *** join/#asterisk mveq (n=drent@unaffiliated/romani) |
| 14:08.07 | *** part/#asterisk mveq (n=drent@unaffiliated/romani) |
| 14:08.12 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:11.03 | [TK]D-Fender | jch2os: Do I have the internet-->cable modem-->router-->linux firewall-->switch? <- good |
| 14:11.30 | jch2os | ah ok, do I need to put it in a special mode or something then? |
| 14:11.33 | jch2os | the router that it |
| 14:11.36 | jch2os | *is |
| 14:11.40 | dlynes | Is there a way to take a call from a sip device, handle whatever needs to be handled, hang up on it, and then make another call in the same session? |
| 14:11.50 | Katty | http://www.youtube.com/watch?v=28GUU1YbP_E <- 1:50 |
| 14:12.16 | leifmadsen | dlynes: there is probably an option in Dial() to continue in the dialplan after a hangup |
| 14:12.18 | dlynes | I just don't want the sip reinvite information from the first call getting transferred to the second call |
| 14:12.34 | [TK]D-Fender | dlynes: Originate / call-file just prior, or triggered by your hangup. |
| 14:12.54 | dlynes | leifmadsen: in this case, I'm accepting a call, then hanging up, and then making a call |
| 14:12.58 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 14:13.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:13.05 | leifmadsen | dlynes: yes, I know what you're doing |
| 14:13.19 | leifmadsen | likely you're approaching it the wrong way and should be done via a script with separate calls |
| 14:13.24 | dlynes | leifmadsen: so a dial command doesn't exist until after it's hung up |
| 14:13.32 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 14:13.45 | *** join/#asterisk smps (n=maher@193.170.53.51) |
| 14:13.52 | leifmadsen | dlynes: right, but you have to have a Dial() command in the first place no? so you tell it to continue on in the dialplan after the hangup |
| 14:14.02 | dlynes | leifmadsen: i.e. -> ReceiveFAX(...), Hangup(), Dial(...M(...)), SendFAX(...) |
| 14:14.16 | leifmadsen | in that case, no |
| 14:14.20 | [TK]D-Fender | dlynes: Spawn a new call. this previous one is dead |
| 14:14.28 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 14:14.36 | DelphiWorld | hi |
| 14:14.38 | [TK]D-Fender | dlynes: "h,1,Originate()" |
| 14:14.39 | DelphiWorld | anyone use Skype for Asterisk? |
| 14:14.42 | leifmadsen | in the 'h' extension you might be able to trigger Originate() |
| 14:14.49 | leifmadsen | DelphiWorld: many people do |
| 14:14.58 | dlynes | [TK]D-Fender: , leifmadsen Oh...didn't know about that application/function |
| 14:15.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:15.06 | leifmadsen | dlynes: only available in 1.6.x I think |
| 14:15.10 | DelphiWorld | leifmadsen: hehehe here and VUC...;) |
| 14:15.13 | [TK]D-Fender | dlynes: Its been in CLI/AMI since... forever <- |
| 14:15.40 | DelphiWorld | any free trial or sometnhing to download? |
| 14:15.41 | [TK]D-Fender | dlynes: And Call-files. |
| 14:15.52 | [TK]D-Fender | DelphiWorld: No. |
| 14:15.56 | dlynes | [TK]D-Fender: It's just for me right now, it's passing along the sip reinvite information from the original call and refusing to authenticate me on the other server because the local sip device isn't recognized there....fromuser, fromdomain and username don't seem to help |
| 14:16.09 | DelphiWorld | hehehe so i can't try it out;) |
| 14:16.12 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 14:16.26 | jaytee | leifmadsen, looked over your queues documentation last night. looks good to me. everything seemed pretty clear to me. |
| 14:16.35 | leifmadsen | jaytee: great, thanks! |
| 14:16.49 | dlynes | leifmadsen: thanks to you as well...never knew about originate() |
| 14:17.04 | jaytee | only spotted one typo on line 599, you left out the letter e in logged. |
| 14:17.13 | dlynes | [TK]D-Fender: I just started using AMI recently, and never use the cli for calls...only for monitoring |
| 14:17.15 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:17.16 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:17.22 | [TK]D-Fender | dlynes: And call-files? |
| 14:17.51 | dlynes | [TK]D-Fender: yeah...use them...have used them in the past, but it just complicates things for this |
| 14:18.02 | [TK]D-Fender | dlynes: Same thing though |
| 14:18.03 | jch2os | [TK]D-Fender - so putty the router in between the cable modem and the firewall how does that work. Say I have a 1.1.1.1 IP on the cable modem, and my firewall uses 1.1.1.2 with a gateway of .1 and my inside network uses 192.168.0.0. How do I setup the router? Does it just work without me setting IP's? Will it do the QOS? |
| 14:18.17 | dlynes | leifmadsen: Originate() doesn't seem to exist...is there a special module I need to load? |
| 14:18.28 | leifmadsen | jaytee: https://issues.asterisk.org/view.php?id=16237 |
| 14:18.33 | leifmadsen | dlynes: what version of asterisk? |
| 14:18.41 | [TK]D-Fender | jch2os: this isn't #qos , ##networking, or #tomato |
| 14:18.42 | dlynes | leifmadsen: 1.6.1.8 |
| 14:18.50 | jch2os | [TK]D-Fender - I know, sorry! |
| 14:18.51 | leifmadsen | must only exist in 1.6.2.x then |
| 14:18.59 | leifmadsen | or you're missing a module, but I have no idea which one it is in |
| 14:19.01 | kaldemar | app Originate is 1.6.2 only, yes. |
| 14:19.16 | jch2os | [TK]D-Fender - I'll figure it out, just thought I would see if you had a quick answer. thanks though |
| 14:19.27 | [TK]D-Fender | dlynes: So just System() it. |
| 14:19.40 | dlynes | [TK]D-Fender: *sigh* |
| 14:20.02 | dlynes | [TK]D-Fender: just trying to reduce load as much as possible :) |
| 14:20.06 | leifmadsen | prefers SHELL() |
| 14:20.10 | leifmadsen | (function) |
| 14:20.55 | [TK]D-Fender | leifmadsen: Yes, it is very useful.... nothing special for his specific need, but very cool that Backticks() was "adopted" |
| 14:21.09 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:21.29 | Katty | :< |
| 14:21.51 | dlynes | Yeah...both are probably useful...although shell() sounds like it's more useful...just doesn't give you a return code |
| 14:22.24 | dlynes | Is 1.6.2 compatible, communication wise over iax2 with 1.6.1 or 1.4.26.2? |
| 14:23.06 | dlynes | I'm asking because 1.4.22rc2 wasn't compatible communications wise over iax2 with 1.6.1.1 |
| 14:25.03 | dlynes | Yeah...only module with originate in the name in 1.6.1.8 is res_clioriginate.so |
| 14:25.50 | *** join/#asterisk plagerism (n=shipclar@142.214.69.216) |
| 14:25.57 | dlynes | kaldemar: I'm guessing the t.38 gateway functionality in 1.6.2 should obviate the need for what I'm doing though? |
| 14:28.21 | *** join/#asterisk oej (n=olle@p5099839d.dip0.t-ipconnect.de) |
| 14:30.55 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 14:31.00 | eppigy | GUTEN MORGAN |
| 14:31.17 | eppigy | MORGEN |
| 14:31.35 | [TK]D-Fender | eppigy: aye Cap'n! |
| 14:32.43 | Katty | http://farm3.static.flickr.com/2584/4100807282_1cc1b0e790_o.jpg |
| 14:36.14 | Katty | eppigy: still sleepies :< |
| 14:39.11 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 14:39.48 | eppigy | Katty: me2 :< |
| 14:41.35 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 14:45.53 | Katty | http://dengedenge.com/wp-content/uploads/2009/11/Cold_War_Vintage_Ads_5.jpg |
| 14:50.08 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:50.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.02 | Katty | http://1.media.tumblr.com/tumblr_ksnt9wbJ7b1qa1c6bo1_500.jpg <- omnomnomnom |
| 14:52.13 | Katty | ^- appears to be baby rotty |
| 14:52.50 | *** join/#asterisk Skeeter- (i=Skeeter-@24.226.190.141) |
| 14:53.30 | Skeeter- | hi guys |
| 14:54.03 | Katty | hi |
| 14:54.28 | Skeeter- | how is the dialplan called for this function: *(ext.) which calls an ext. voicemail automaticly |
| 14:54.32 | *** join/#asterisk TiCPU (n=TiCPU@c216.218.2-65.clta.globetrotter.net) |
| 14:54.53 | [TK]D-Fender | Skeeter-: huh? |
| 14:55.27 | Skeeter- | [TK]D-Fender: the function works with internal ext. i would it to reach ext. from the IAX2 trunk |
| 14:55.42 | [TK]D-Fender | Skeeter-: What function? No such thing exists unless you invent it |
| 14:56.12 | [TK]D-Fender | Skeeter-: #freepbx <- back to MagicHappyGUILand with you...... |
| 14:56.53 | Skeeter- | [TK]D-Fender: how do you reach someone ext. voicemail directly, is that a MagicHappyGUILand function or astertisk function |
| 14:59.52 | Katty | giggles |
| 15:00.49 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:02.18 | [TK]D-Fender | Skeeter-: What extension? One does not exist unless you create it |
| 15:02.43 | Katty | eppigy: http://i.imgur.com/ANqLR.jpg <- i saw this and thought of you. |
| 15:04.40 | eppigy | lol |
| 15:04.45 | eppigy | im not sure how to take that |
| 15:05.06 | *** join/#asterisk vally (n=vally@62.48.76.175) |
| 15:06.47 | *** join/#asterisk jermudgeon (n=jermudge@216-67-61-242.static.acsalaska.net) |
| 15:07.36 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 15:07.56 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 15:10.16 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 15:15.18 | *** join/#asterisk zdz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 15:16.06 | *** join/#asterisk ManxPower-work (n=EWieling@82.sub-70-222-194.myvzw.com) |
| 15:16.50 | zdz | does anyone know what facility levels SNOM phones use for syslog? |
| 15:16.56 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:21.38 | *** part/#asterisk zdz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 15:23.14 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 15:30.43 | ghento | Hi all. Has anyone experienced the problem with MP3Player having mpg123 hang? I did a `ps | aux` and noticed quite a number of [mpg123] <defunct> processes listed, all stemmed from asterisk. I'm running 1.9.1 of mpg123. In the asterisk console I see "OTICE[19851]: app_mp3.c:136 timed_read: Poll timed out/errored out with 0 , app_mp3.c:214 mp3_exec: No more mp3". |
| 15:30.51 | *** join/#asterisk bahjons (n=robert@140.99.23.26) |
| 15:31.43 | bahjons | has anyone been successful with the state_interface backport for v1.4? |
| 15:32.12 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 15:32.55 | ariel_ | hello all |
| 15:34.25 | jblack | ghento: that would be an asterisk bug. please file a bug at the bts |
| 15:35.26 | jblack | the defunct processes are, i mean |
| 15:36.58 | ghento | jblack: ok tanks. |
| 15:39.43 | Chainsaw | Tanks! With guns on them. |
| 15:40.01 | Chainsaw | has actually switched to using dumbout (instead of mpg123) as his hold music is in S3M format. |
| 15:40.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 15:40.07 | *** join/#asterisk baijum (n=baiju@122.166.149.190) |
| 15:48.16 | *** join/#asterisk spck (n=spck@75-135-75-112.dhcp.mdsn.wi.charter.com) |
| 15:55.17 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 15:57.22 | ghento | Haha Chainsaw. Thanks :) |
| 15:57.39 | *** join/#asterisk telnettech (n=telnette@71.5.32.196) |
| 15:58.35 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:00.17 | *** join/#asterisk jermudgeon (n=jhaustin@216-67-61-242.static.acsalaska.net) |
| 16:06.22 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:09.23 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 16:12.26 | *** join/#asterisk Uhrheber (i=karlklos@ssh.sign.io) |
| 16:13.13 | Uhrheber | Hi. Anybody here that has experience with asterisk on a NSLU2? |
| 16:13.43 | [TK]D-Fender | Uhrheber: What in particular? |
| 16:14.36 | Uhrheber | The xscale processor has a DSP coprocessor. Do the codecs use it or are they only integer? |
| 16:18.24 | Katty | eppigy: well you're always shouting about pork rinds |
| 16:18.45 | Katty | eppigy: what are they called again? |
| 16:19.00 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-70.twcny.res.rr.com) |
| 16:21.03 | Uhrheber | Is OpenWRT a good OS to start with asterisk? I like it because of it's small uClib, but the premade images always have the WLAN things that I don't need |
| 16:21.37 | Uhrheber | And removing things from the default config is a time consuming trial and error procedure. |
| 16:23.00 | Uhrheber | Is there any chance to get an premade image with a working asterisk, that fits into the NSLU2's 8MB flash? |
| 16:24.03 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 16:24.19 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 16:24.25 | *** join/#asterisk nfi|ermes (n=ErMeS@host54-26-static.93-94-b.business.telecomitalia.it) |
| 16:25.52 | eppigy | Katty: CHICHARONES |
| 16:26.00 | eppigy | or somethin like that |
| 16:26.37 | Katty | :> |
| 16:27.00 | eppigy | yesh |
| 16:27.07 | Katty | cleverbot is cute |
| 16:27.09 | ariel_ | chicharones, spanish for fried pork skin |
| 16:27.29 | Uhrheber | _-P |
| 16:27.32 | Katty | http://imgur.com/IO5Od <- |
| 16:28.32 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:29.18 | *** join/#asterisk maskas (n=maskas@90.158.70.202.dynamic.max.com.pk) |
| 16:29.21 | maskas | hello |
| 16:29.44 | *** join/#asterisk zapotek6 (n=edpman@mail.comelit.it) |
| 16:30.21 | Katty | oh this one is hilarious |
| 16:30.22 | maskas | I've set option g in my dial command, but my dialplan still wont proceed to the next step in the dialplan on answered calls |
| 16:30.26 | Katty | ariel_: i will need your translation skills |
| 16:30.33 | maskas | using asterisk 1.4.26 |
| 16:31.33 | ariel_ | really |
| 16:31.44 | Katty | http://pastebin.ca/1670114 <- |
| 16:32.01 | maskas | would appreciate if someone can help? |
| 16:34.13 | ChannelZ | maskas: That's not what g does - it makes it continue if the call hangs up |
| 16:34.14 | Tim_Toady | maskas it proceeds as soon as the call ends |
| 16:34.17 | ariel_ | Katty: I don't know or have used this word: Tramtadadá |
| 16:34.19 | Katty | ariel_: any luck on that translation? |
| 16:34.24 | Katty | ah :< |
| 16:35.02 | Katty | Tramtadadá <- does anyone know what the word is? |
| 16:35.12 | Katty | Abi ko kasabot ka <- or this phrase? |
| 16:35.18 | *** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk) |
| 16:35.23 | ariel_ | not spanish |
| 16:35.31 | ariel_ | at least not that I have heard of |
| 16:35.34 | maskas | tim_toady: Thats what I meant, I want it to proceed once the call hangs up |
| 16:36.16 | maskas | but even once the call hangs up it wont goto the next priority |
| 16:36.17 | Tim_Toady | maskas a dialplan sample and maybe the output of a call with verbose set to 3 might help |
| 16:36.29 | ariel_ | use pastebin |
| 16:36.48 | [TK]D-Fender | maskas: "g" only works if the CALLED party is the one that ends the call |
| 16:37.07 | [TK]D-Fender | maskas: Otherwise the call will jump to the "h" Asterisk Standard Extension. |
| 16:37.37 | *** join/#asterisk Defraz (n=Defraz@192.41.17.69) |
| 16:38.07 | *** join/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:38.13 | *** join/#asterisk ManxPower-work (n=EWieling@234.sub-75-254-221.myvzw.com) |
| 16:38.26 | *** part/#asterisk ryduh (n=ryduh@204.16.143.186) |
| 16:38.40 | *** join/#asterisk wonderworld (n=ww@62.143.22.226) |
| 16:39.39 | maskas | ok actually, my bad it is going to the next priority with option g |
| 16:39.57 | maskas | but for some reason its not writing the variable in the cdr |
| 16:40.43 | *** join/#asterisk Benatonshore (n=ben@DHCP-192-113.onshore.com) |
| 16:41.31 | maskas | Do I need to do something else for it to write the hangup cause code in the cdr |
| 16:42.03 | [TK]D-Fender | maskas: You aren't showing us anything to debug... |
| 16:42.04 | [TK]D-Fender | ~pb |
| 16:42.05 | 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 |
| 16:42.07 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 16:43.19 | Katty | infobot: cleverbot? |
| 16:43.23 | *** join/#asterisk KingDavidNYC (n=Chris123@66.7.86.2) |
| 16:43.31 | KingDavidNYC | hello everybody!!!! |
| 16:43.34 | Katty | infobot: cleverbot is http://cleverbot.com/ |
| 16:43.35 | infobot | Katty: okay |
| 16:46.04 | maskas | just a second, will pastebin |
| 16:46.40 | Katty | http://pastebin.ca/1670133 <- my conversation with cleverbot about asterisk |
| 16:47.12 | *** join/#asterisk xpot-mobile (n=xpot@c-67-177-18-234.hsd1.ut.comcast.net) |
| 16:47.16 | maskas | http://pastebin.ca/1670135 |
| 16:47.34 | *** join/#asterisk xpot-mobile (n=xpot@67.177.18.234) |
| 16:47.41 | *** join/#asterisk andreas-- (n=andy@ppp079166022016.dsl.hol.gr) |
| 16:47.56 | Katty | infobot: What is trixbox? |
| 16:47.57 | infobot | well, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 16:48.03 | Katty | bummer. |
| 16:49.24 | eppigy | NEIN |
| 16:50.15 | ManxPower-work | the lowly infobot us smarter than all the people using FreePBX. |
| 16:50.42 | Katty | ManxPower-work: did you see my pastebin above about the cleverbot conversation? |
| 16:51.05 | ManxPower-work | Katty: I saw the URL, I didn't read it. |
| 16:51.20 | Katty | k |
| 16:51.26 | maskas | I've pasted my dialplan, the output from console and cdr into http://pastebin.ca/1670135 , would appreciate if you can check |
| 16:52.02 | jaytee | taking a half day today, gonna "git da earl chaynged in muh cur" |
| 16:52.51 | Katty | leafs for lunch |
| 16:53.03 | *** join/#asterisk Blackvel (n=blackvel@84.57.87.135) |
| 16:53.21 | Blackvel | hi. anyone with patton 4634 media gateway here? |
| 16:54.53 | Tim_Toady | maskas in this paste the dialplan continues after dial ends, i dnt see the prob you mentioned |
| 16:56.24 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 16:58.06 | *** join/#asterisk aidinb (n=Aidin@24.182.32.138) |
| 16:58.29 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-126-Utah.hfc.comcastbusiness.net) |
| 16:58.41 | maskas | tim_toady: yes it is continuing, but if you see the cdr the hangup cause code is not in the cdr and the last app is dial, not hangup as it should be |
| 17:00.35 | *** join/#asterisk andres833 (n=andres83@166.238.40.243) |
| 17:02.22 | Hunner | Would I have to forward ports through my nat to my asterisk box to register to a sip trunk like sipgate? |
| 17:02.57 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:02.58 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:02.59 | Hunner | or only if I wanted devices registering to my box? |
| 17:03.04 | *** part/#asterisk bahjons (n=robert@140.99.23.26) |
| 17:05.36 | *** join/#asterisk jermudgeon_ (n=jhaustin@69-161-30-140.static.acsalaska.net) |
| 17:05.49 | ManxPower-work | ~answers |
| 17:05.49 | infobot | methinks 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 |
| 17:05.54 | ManxPower-work | Hunner: see the NAt link |
| 17:06.47 | bmoraca | woot...gettin me a AS5400 to play with |
| 17:06.47 | Amorsen | It would be nice if Asterisk registered which end hung up automatically in the CDR |
| 17:07.28 | maskas | Amorsen: thats exactly what I'm trying to do, any idea how I can do it? |
| 17:08.51 | Amorsen | maskas: You have to proceed as you did, with a h extension and the appropriate dial option |
| 17:08.58 | Amorsen | I haven't had much luck with it myself |
| 17:09.32 | *** join/#asterisk jorgeluiso (n=jorgeoli@190.74.121.69) |
| 17:09.47 | maskas | well it shows it is setting the variable on the console |
| 17:09.48 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:09.52 | maskas | but in the cdr its not there |
| 17:10.46 | Amorsen | Ah, cdr on hangup |
| 17:10.54 | Amorsen | There's an option you must set for that to work |
| 17:11.04 | maskas | oh, what option is that? |
| 17:11.08 | *** part/#asterisk andreas-- (n=andy@ppp079166022016.dsl.hol.gr) |
| 17:11.09 | Amorsen | Good question |
| 17:11.34 | Amorsen | Logically named "endbeforehexten" |
| 17:12.59 | maskas | where would I set that? |
| 17:13.28 | Amorsen | cdr.conf |
| 17:13.40 | Amorsen | (That last one google could have told you) |
| 17:14.50 | maskas | sorry, yes I would it on google |
| 17:14.55 | maskas | *found |
| 17:15.46 | maskas | it should be = no right? |
| 17:15.54 | Amorsen | Right |
| 17:15.57 | Amorsen | Err no |
| 17:16.02 | Amorsen | It should be yes |
| 17:16.14 | Amorsen | I'm confusing myself |
| 17:16.19 | Amorsen | Try it out :) |
| 17:16.56 | Hunner | ManxPower-work: okay, I think I'm doing everything right, but I get no answer from my sip provider |
| 17:16.59 | Hunner | just 17:16:13.902630 IP 192.168.1.109.5060 > mail.gsmcall.com.5060: SIP, length: 413 |
| 17:17.44 | ManxPower-work | Hunner: EVERY instruction must be followed. Most issues are because someone didn't put "canreinvite=no" or didn't port forward or whatever. |
| 17:18.29 | maskas | ok thanks amorsen |
| 17:19.07 | jorgeluiso | Hi, I want to replace the charactter |
| 17:19.21 | Qwell | jorgeluiso: the character? |
| 17:19.45 | jorgeluiso | sorry a charactter * for A |
| 17:19.57 | jorgeluiso | within the dial plan |
| 17:20.36 | jorgeluiso | basically to have an appropriate monitor file name when the exten contents a * |
| 17:21.27 | jorgeluiso | something like exten => s,n,Replace(FILENAME,${EXTEN},*,A) |
| 17:22.56 | jorgeluiso | I know the app replace doesnt exist but I'm looking for the replace funcionality. |
| 17:24.34 | maskas | amorsen: I tried with both yes and no, but still no luck |
| 17:31.59 | Amorsen | maskas: And you restarted asterisk between tests? |
| 17:32.16 | maskas | Amorsen: I did reload |
| 17:32.31 | maskas | I cant restart as there is live traffic on it |
| 17:32.55 | *** join/#asterisk jermudgeon (n=jhaustin@69-161-30-140.static.acsalaska.net) |
| 17:36.01 | archiac_ | why can't you just do a reload? |
| 17:38.44 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 17:38.52 | archiac_ | is the context name just a string; meaning I can name a context an IP? [1.2.3.4] |
| 17:40.04 | ruben23 | hi anyone what error is this coming on my CLI--->http://pastebin.com/m59a87501 |
| 17:41.37 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 17:44.14 | Blackvel | anyone knows if patton 4634 does HWEC (for inbound)? |
| 17:47.51 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 17:50.18 | *** join/#asterisk errotan (n=errotan@a1592.adsl.pool.eol.hu) |
| 17:52.27 | *** join/#asterisk clart001 (n=clart@host209-222-dynamic.11-87-r.retail.telecomitalia.it) |
| 17:53.52 | *** join/#asterisk Tim_Toady (n=moi@adsl119-125.kln.forthnet.gr) |
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| 17:56.08 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:56.45 | jblack | I need the names of 4 or 5 companies I've helped with asterisk |
| 17:57.09 | jblack | So, if I've ever helped you with asterisk, and you work at a company with a recognizable name, please msg me. |
| 18:02.07 | Katty | jblack: have you ever helped me with asterisk stuff? |
| 18:02.23 | Katty | jblack: or do you mean Officially |
| 18:02.39 | jblack | ever asked me a question, and got an answer. |
| 18:02.45 | Katty | thinks |
| 18:02.52 | Katty | i don't know :< |
| 18:03.00 | jblack | I can't think of anything. |
| 18:03.13 | Katty | :< |
| 18:03.13 | jblack | So, ask me something you don't know, so that I can say I provided consultation to your company. =) |
| 18:03.26 | Katty | hmmmm |
| 18:03.31 | Katty | pulls up dialplan |
| 18:03.41 | Chainsaw | jblack: My hold-music sometimes skips a little. I'm using dumbout. What can I do about that? |
| 18:04.53 | jblack | chainsaw: Hrmm. You're what, using an external application, and getting skipping? |
| 18:05.02 | Katty | pastebins |
| 18:05.11 | Chainsaw | jblack: Yes. |
| 18:05.16 | jblack | I don't know of anything specifically called "dumbout", but I see examples that provide a musiconhold context called dumbout. |
| 18:05.23 | Katty | http://pastebin.ca/1670219 |
| 18:05.28 | jblack | chainsaw: I'd try setting a bigger buffer. |
| 18:05.39 | Katty | oh. wait. that's the wrong bit. |
| 18:05.53 | jblack | Chainsaw: what program are you using to play mp3s? |
| 18:06.06 | Katty | http://pastebin.ca/1670220 <- let me know when you're ready. |
| 18:06.29 | jblack | Oh, faxing is hard, but I'll try |
| 18:06.30 | jblack | what's up?: |
| 18:06.39 | Katty | well this isn't about faxing, directly |
| 18:06.48 | Katty | windows can't create a usable tiff that linux likes. |
| 18:07.00 | Katty | so i'm looking for a pdf -> tif commandline converter |
| 18:07.30 | jblack | probably have to take a couple steps. |
| 18:07.41 | Katty | yes, probably |
| 18:07.44 | Chainsaw | jblack: application=/usr/bin/dumbout /etc/asterisk/LINX/TheBlueValley.s3m -m -s 8000 -r 2 -v 0.5 -o - |
| 18:08.02 | Chainsaw | jblack: On mode=custom, cachertclasses=yes. |
| 18:08.13 | Katty | you can help me later (= |
| 18:08.32 | jblack | Yeah, it'll take me a moment to get pdf into something useful |
| 18:08.43 | jblack | You actually have an app called dumbout? |
| 18:08.52 | Chainsaw | jblack: That is correct. |
| 18:08.57 | jblack | What is that, a script? |
| 18:09.16 | Chainsaw | jblack: It's an executable. It uses the "DUMB" framework to play the module audio file (in S3M screamtracker format). |
| 18:09.22 | jblack | Those options look suspiciously close to pplay to m3 |
| 18:09.41 | [TK]D-Fender | Katty: i see you still haven't followed yesterday's advice |
| 18:09.57 | jblack | Chainsaw: double check to make sure it's a binary and not a script? |
| 18:09.58 | Katty | [TK]D-Fender: i plan on turning it into a call file. i want to keep it simple for now. |
| 18:10.10 | Katty | [TK]D-Fender: so simmer down, dear. |
| 18:10.37 | [TK]D-Fender | Katty: its 1 line of dialplan.... |
| 18:10.43 | Katty | [TK]D-Fender: yeah i don't really care (= |
| 18:10.45 | Katty | pats [TK]D-Fender |
| 18:10.47 | Katty | thanks though. |
| 18:10.47 | [TK]D-Fender | Katty: so your way IS complicated |
| 18:10.58 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 18:11.24 | Katty | that's nice. |
| 18:11.33 | jblack | Oh, THAT s3m.. |
| 18:11.36 | jblack | cute. ;) |
| 18:11.40 | ruben23 | hi anyone have idea on this error log on my asterisk CLI------->[Nov 13 10:11:31] WARNING[28618]: rtp.c:891 ast_rtcp_read: RTCP Read too short |
| 18:11.51 | jblack | I bet mplayer can handle those. |
| 18:11.52 | ruben23 | i got a lot of it on the CLI screen |
| 18:12.05 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 18:12.40 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 18:13.28 | Chainsaw | jblack: So where do I set this buffering option? |
| 18:14.03 | jblack | I don't have that specific binary, so I'm checking into whether mplayer has s3m support, in which case, we can just switch you over, and easily set up the buffersize easily |
| 18:15.49 | jblack | It's not exactly a common format. |
| 18:15.58 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 18:16.03 | jblack | you ought to consider just converting them to gsm and be done with it |
| 18:16.38 | Katty | sways |
| 18:17.20 | Chainsaw | jblack: It does waste a lot of disk space that way. |
| 18:17.27 | jblack | gsm? Nah. |
| 18:17.45 | jblack | grabs chiptunes sources to build |
| 18:17.54 | Chainsaw | jblack: 553 kilobytes for 12 minutes of audio. |
| 18:17.58 | Chainsaw | jblack: If GSM can do that, colour me impressed. |
| 18:20.50 | jblack | oh christ. |
| 18:20.57 | jblack | I see why you had such a problem |
| 18:21.12 | Katty | http://www.youtube.com/watch?v=LCHrioqR_oo :> |
| 18:22.01 | Chainsaw | jblack: But you're saying the Asterisk end will not buffer. |
| 18:22.10 | Chainsaw | jblack: So I need to discover command switches for this thing and get it to pump more audio. |
| 18:22.27 | jblack | chainsaw: well, asterisk is running out... so it's the other side that needs to buffer more. |
| 18:22.36 | jblack | it's starving. |
| 18:22.40 | Chainsaw | jblack: *nod* So it's a buffer underrun. Understood. |
| 18:22.50 | Chainsaw | jblack: You can add the London Internet Exchange to your consultancy list. |
| 18:22.50 | jblack | worst case scenario, we make a big ass fifo, feed the fifo. |
| 18:23.03 | Amorsen | Sometimes a regex_replace function would be handy. Call it s perhaps... |
| 18:23.05 | Chainsaw | jblack: https://www.linx.net/ |
| 18:23.11 | jblack | Are you really using s3ms there? |
| 18:23.23 | Chainsaw | jblack: For my hold music? Yes. |
| 18:23.31 | jblack | You're old school. |
| 18:23.41 | *** join/#asterisk Tech_Travis (n=tech_tra@mail.techglia.com) |
| 18:24.17 | Chainsaw | jblack: It's the Blue Valley. You'll know it. |
| 18:24.23 | Chainsaw | jblack: (Ever played Uplink?) |
| 18:24.48 | jblack | I think I played everything on the 64 |
| 18:25.04 | jblack | I dont' remember uplink specifically though. |
| 18:25.17 | Chainsaw | jblack: It was a PC game. |
| 18:26.46 | ruben23 | hi anyone have idea on this error log on my asterisk CLI------->[Nov 13 10:11:31] WARNING[28618]: rtp.c:891 ast_rtcp_read: RTCP Read too short |
| 18:26.46 | ruben23 | (10:13:02 AM) ruben23: i got a lot of it on the CLI screen |
| 18:27.22 | jblack | hrmm. xine plays s3m. |
| 18:27.42 | jblack | I missed this game. |
| 18:28.34 | jblack | gasps |
| 18:28.45 | jblack | I wanted this game so badly! |
| 18:28.57 | clart001 | excuse me, where do i configure an iax trunk? |
| 18:28.58 | Chainsaw | It's perfect for hold music. Network techies are likely to recognise it; it doesn't deteriorate on a bad GSM link like a violin does. |
| 18:29.09 | jblack | This isn't bad for hold music, actually. |
| 18:29.37 | kaldemar | clart001: iax.conf |
| 18:29.57 | jblack | So, let's look at making a big fifo. |
| 18:30.36 | jblack | are you familiar with mkfifo ? |
| 18:30.49 | Chainsaw | I've not used it before, no. |
| 18:31.18 | Chainsaw | (Just doublechecked the possible dumbout arguments, it does not support buffering) |
| 18:31.30 | jblack | yeah, it's basically a port. |
| 18:31.39 | jblack | from dos, it looks like. |
| 18:31.41 | clart001 | @kaldemar: link in sip.conf |
| 18:31.44 | clart001 | ? |
| 18:33.33 | *** join/#asterisk xenoterracide (n=xenoterr@c-68-42-198-183.hsd1.mi.comcast.net) |
| 18:33.52 | xenoterracide | anyone know if it's possible to connect asterisk to ventrilo? |
| 18:34.15 | jblack | Chainsaw: still working on this |
| 18:34.45 | Chainsaw | jblack: It uses barely any CPU, which is another reason why I like this approach. |
| 18:35.06 | Chainsaw | jblack: The MP3 file was huge and it took more CPU to decode then it takes to play this teeny tiny file. |
| 18:36.41 | Chainsaw | jblack: Actually... listening to it now it hasn't dropped once. |
| 18:36.51 | Chainsaw | casts a suspicious look at his Cisco handset |
| 18:39.09 | Chainsaw | jblack: Yes, cancel that. This looks to have been an overloaded link elsewhere. |
| 18:39.37 | jblack | Oh, cmon... I just got dumbout to dump into a big fifo |
| 18:39.59 | Chainsaw | Dropping the volume to 0.4 instead of 0.5 was a good move. |
| 18:40.01 | Chainsaw | Sorry :/ |
| 18:40.09 | jblack | echo cancellation. Heh |
| 18:40.12 | [TK]D-Fender | xenoterracide: No. |
| 18:40.26 | Tech_Travis | If a call goes into a queue with timeout set, and a random member is logged in but does not answer the call, is there a way to get that extension so I can pass it to a Dial()? |
| 18:42.56 | xenoterracide | [TK]D-Fender: ok thanks. I was unable to get ventrilo working properly through wine. so I was thinking maybe another voip could talk ot it |
| 18:43.00 | jblack | Katty: Ping |
| 18:43.15 | Katty | is updating her ringtone |
| 18:43.34 | jblack | Ok. pdf to tiff. |
| 18:43.41 | Katty | meh |
| 18:43.42 | jblack | that's right |
| 18:44.05 | Katty | jblack: i'll email you my ringtone |
| 18:44.29 | jblack | Ohh, there's apdf2svg. Not what we need, but nice. ;) |
| 18:45.46 | Katty | where am i sending it? |
| 18:45.52 | Katty | linuxguru? |
| 18:46.12 | jblack | sure. |
| 18:46.36 | jblack | Ahh, here we go. |
| 18:47.27 | Katty | jblack: sent |
| 18:49.56 | Katty | bah, it's too long. |
| 18:50.01 | jblack | I checked an old script. I used ghostscript to convert pdf to text |
| 18:50.08 | jblack | I don't have a max filesize on my email |
| 18:50.19 | jblack | how about filebin.com ? |
| 18:50.20 | Katty | it's too big for my blackberry |
| 18:50.27 | Katty | no i sent it |
| 18:50.34 | jblack | there's a pastebin for files somewhere |
| 18:50.46 | Katty | no, my blackberry doesn't like it |
| 18:50.50 | Katty | it's too long for it to use as a ringtone |
| 18:50.51 | jblack | anyways, we should be able to convert the pfs to ghostscript, suitable for fax printing |
| 18:51.07 | jblack | so, chop it down a bit with audacity. :) |
| 18:51.13 | Katty | is |
| 18:51.15 | jblack | hell, I'll do it for ya if you want |
| 18:51.38 | jblack | filebin.ca, was the site I was thinking about |
| 18:51.53 | jblack | Ok, so, pdf to tiffs. |
| 18:51.59 | Katty | sure, you can snippet my ringtone |
| 18:52.04 | Katty | that sounds more interesting than pdfs anyway |
| 18:52.12 | Katty | take out the micheal jackson bit. |
| 18:52.15 | Katty | but leave the sweet dreams |
| 18:52.29 | jblack | It's the sperm people! |
| 18:52.36 | *** join/#asterisk pzn (n=pzn@189.35.189.104) |
| 18:52.50 | Katty | i have the 8 second intro |
| 18:52.52 | Katty | if that will help |
| 18:52.55 | Katty | it used to be my ringtone |
| 18:52.57 | jblack | I couldn't get that out. I'm not that good at audio manipulation... |
| 18:53.12 | jblack | That said... if you want the spermpeople, could loopback a recording of that youtube video |
| 18:53.33 | jblack | is the billy jean part the only part? |
| 18:53.45 | pzn | is there any way to make an extension work like this: Dial(sip/300) -- and ring during 10 seconds, if not answered, playback()... how to do this? |
| 18:54.44 | [TK]D-Fender | pzn: Dial(sip/300,10) , then do other stuff |
| 18:55.41 | pzn | [TK]D-Fender: wow! that simple! thanks! |
| 18:56.23 | *** join/#asterisk J-zimbam (n=zimbam@c-98-220-112-218.hsd1.in.comcast.net) |
| 18:56.55 | Katty | there |
| 18:57.04 | Katty | sends jblack a copy |
| 18:57.42 | jblack | I just got MJ out here too |
| 18:57.50 | Katty | send it to me |
| 18:58.03 | jblack | sperm people singing nirvana makes me uncomfortable |
| 18:58.13 | J-zimbam | Anyone have any ideas why Asterisk would consume 99+% CPU usage randomly? It runs fine for hours and even days. Then spikes to the 90's. It does it during high and low traffic times as well. |
| 18:58.14 | superbeef | so how sinful is it to use an ethernet patch able for a T1 patch? |
| 18:58.34 | jblack | superbeef: Not sinful at all. |
| 18:58.35 | Katty | pouts |
| 18:58.37 | Katty | still too big |
| 18:58.40 | Katty | kicks blackberry |
| 18:58.47 | jblack | katty: Cheat. Change the bitrate! |
| 18:59.06 | jblack | the sample you emailed me was stereo. Is your phone stero? |
| 18:59.07 | superbeef | jblack: good deal.. I didnt thnk it was, but i'm having to cover all my bases |
| 18:59.14 | jblack | That right there will cut it in half. |
| 18:59.17 | Katty | good point |
| 18:59.27 | ManxPower-work | superbeef: There is no issue with using an ethernet cable as a T-1 cable. HOWEVER, an ethernet crossover cable cannot be used as a T-1 crossover cable. |
| 18:59.58 | superbeef | ManxPower-work: yeah i guess that makes sense... I've made my own T1 crossover for my testbox |
| 19:00.46 | jblack | LOLOLOLOL! Katty: search youtube for "sperm people" |
| 19:01.45 | Katty | what format do you think i should encode this audio file in |
| 19:01.46 | J-zimbam | we upgraded from asterisk 1.2 to 1.4 and ever since we get the random CPU spikes |
| 19:02.10 | jblack | For your phone? I'd stick with mp3 if it can do it. Just cut it to mono, and maybe take the bit rate down. |
| 19:02.24 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp138-119.adsl.forthnet.gr) |
| 19:02.38 | Katty | pokes around cdex for mono setting |
| 19:02.42 | *** part/#asterisk clart001 (n=clart@host209-222-dynamic.11-87-r.retail.telecomitalia.it) |
| 19:02.45 | jblack | J-zimbam: Hrmm. You've gotta figure out what's going on so it doesn't seem random. |
| 19:03.00 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 19:03.06 | jblack | J-zimbam: You're not gonna be able to figure out what's wrong until you discover the pattern. |
| 19:03.19 | J-zimbam | yeah, ive turned up debug and everything to find a pattern but after 2 weeks...notta |
| 19:03.20 | jblack | katty: mencoder or ffmpeg may be of help too |
| 19:03.22 | Katty | the problem is a severe lack of chocolate |
| 19:03.29 | Katty | cdex has a mono setting |
| 19:04.17 | *** part/#asterisk xenoterracide (n=xenoterr@c-68-42-198-183.hsd1.mi.comcast.net) |
| 19:04.30 | jblack | I downconverted stuff last night with ffmpeg. ffmpeg ringtone.mp3 -acodec mp3 -ac 1 -ab 32 smallringtone.mp3 |
| 19:04.32 | J-zimbam | after we restart asterisk it works great for awhile |
| 19:04.45 | jblack | Ok. then, after two weeks, what? |
| 19:04.47 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 19:04.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:04.48 | J-zimbam | then a day or 2 later it starts again. |
| 19:04.51 | Katty | yay 800kb |
| 19:05.13 | J-zimbam | i meant i havent found a pattern after 2 weeks |
| 19:05.16 | jblack | katty: Now add a r option to a dial somewhere, and say you consulted with me. :P |
| 19:05.18 | Katty | alkjsdflkajwofnalksdf |
| 19:05.23 | Katty | throws blackberry |
| 19:05.28 | jblack | you're kidding. |
| 19:05.39 | jblack | there's no way 800kb is too big |
| 19:06.01 | Chainsaw | jblack: Much too big. 64kb ought to be enough for anybody. |
| 19:06.05 | jblack | The error message is lying. IT doesn't like the format. |
| 19:06.11 | Katty | tries 700kb |
| 19:06.19 | jblack | Chainsaw: Heh. Go s3m. :P |
| 19:07.18 | Blackvel | katty: could you find any voip sip application for blackberry (e.g bold 9000) and connecting it to asterisk? |
| 19:07.26 | Katty | tries 400 |
| 19:07.35 | Katty | Blackvel: in theory, gizmo does. |
| 19:07.45 | jblack | katty: Maybe it's not size, but length ? |
| 19:07.46 | Katty | Blackvel: but it's not a pretty sight. |
| 19:08.05 | Blackvel | wasnt it java on blackberry? shouldnt be too hard to write a sip client (not me!) |
| 19:08.05 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1176436941.dsl.bell.ca) |
| 19:08.17 | Katty | jblack: that just sounds wrong. |
| 19:08.21 | darkdrgn2k | any one know where i can find a cisco 7940 configuration instructions |
| 19:08.29 | jblack | That's a pretty long mp3. |
| 19:08.39 | jblack | Try making a 15 second long clip. No way that's too long |
| 19:08.53 | Chainsaw | darkdrgn2k: For SIP or for SCCP? |
| 19:09.01 | Blackvel | darkdrgn2k: do you know if cisco phones compensate inbound echo better than a snom 370? |
| 19:09.02 | darkdrgn2k | Chainsaw: SIP (work with asterisk) |
| 19:09.11 | Chainsaw | darkdrgn2k: Both work with Asterisk. |
| 19:09.14 | darkdrgn2k | Blackvel: no idea, just got my cisco phones in the mail. |
| 19:09.17 | darkdrgn2k | Chainsaw: whats better? |
| 19:09.21 | Chainsaw | darkdrgn2k: Anyhow, try a Google for 7960 SIP. |
| 19:09.26 | Katty | it plays the 400kb one, but it sounds like crap |
| 19:09.29 | J-zimbam | is there a way to clear cache that Asterisk might be building up over time without restarting? |
| 19:09.30 | jblack | drmessano: ping |
| 19:09.30 | Chainsaw | darkdrgn2k: Because a 7940 is just a 7960 missing two line buttons. |
| 19:09.35 | Katty | and 600kb is too large. |
| 19:09.39 | Chainsaw | darkdrgn2k: All other config is identical :) |
| 19:09.47 | Katty | looks like i'mg oign to have to shorten the audio clip if i want to be able to have any quality work speaking of |
| 19:09.49 | jblack | Really, so it is size, and not length? |
| 19:09.58 | jblack | Let me see what I can do here, ok? |
| 19:10.03 | Chainsaw | darkdrgn2k: SIP is the industry standard, go with that. |
| 19:10.15 | darkdrgn2k | well im gussing i need toupdate firmware |
| 19:10.19 | Chainsaw | darkdrgn2k: (Unless you plan on using a lot of the XML features in the menu, because that doesn't work for SIP) |
| 19:10.20 | darkdrgn2k | (copy right date on this virmware is 2005) |
| 19:10.24 | Chainsaw | darkdrgn2k: You'll want 8.11; *not* 8.12 |
| 19:10.29 | *** join/#asterisk muiro (n=muiro@cpe-173-89-177-15.neo.res.rr.com) |
| 19:10.37 | darkdrgn2k | might i ask why |
| 19:10.52 | Chainsaw | darkdrgn2k: Because 8.12 screws up caller ID text. |
| 19:10.57 | Katty | jblack: no, length is fine. |
| 19:11.04 | Chainsaw | darkdrgn2k: (It can't deal with spaces for example) |
| 19:11.05 | darkdrgn2k | Chainsaw: thanx for the heads up |
| 19:11.18 | Katty | jblack: i sent you the original one i had |
| 19:11.24 | ManxPower-work | jblack: This needs to be in a /topic somewhere "jblack: Really, so it is size, and not length?" |
| 19:11.31 | darkdrgn2k | does sip work with xml (basic xml) |
| 19:11.55 | Katty | jblack: except in mono |
| 19:12.01 | jblack | turns purple |
| 19:12.07 | Chainsaw | darkdrgn2k: The SIP firmware has only very basic XML menu support. A lot of the cool things you read about (like changing softkeys) will not work. |
| 19:12.25 | Chainsaw | darkdrgn2k: But the basic phone capabilities, sure. That's all good. |
| 19:12.31 | [TK]D-Fender | [14:09]<Chainsaw>darkdrgn2k: Because a 7940 is just a 7960 missing two line buttons. <- 4 |
| 19:12.32 | darkdrgn2k | LOL ok |
| 19:12.35 | jblack | she said it's not the size that counts, but how you use it |
| 19:12.42 | Chainsaw | Okay Fender. |
| 19:13.01 | jblack | whoops, i used lame, not ffmpeg |
| 19:13.09 | darkdrgn2k | any idea where i can get a new firmware? |
| 19:13.30 | jblack | gives katty a big smile |
| 19:13.40 | darkdrgn2k | ooo 7.2(4.0) |
| 19:13.41 | jblack | katty: I can get good quality at 135k |
| 19:13.48 | Katty | send it |
| 19:13.55 | jblack | can you send me your nonmj version? |
| 19:14.05 | Katty | my omnomnomnom version?! |
| 19:14.11 | Katty | digs through advance doptions |
| 19:14.34 | jblack | non-mj is what you want, right? |
| 19:14.55 | jblack | if you have lame there, you can do it yourself. lame ringtone.mp3 --abr 16 -a ringtones.mp3 |
| 19:15.03 | Katty | oh |
| 19:15.07 | Katty | not micheal jackson? |
| 19:15.15 | jblack | yeah. you wanted him cut out, right? |
| 19:15.18 | Chainsaw | darkdrgn2k: You may find this page helpful, it has download links near the bottom: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 19:15.19 | jblack | and you did so? |
| 19:15.30 | Katty | well yes, i did cut that audio bit out |
| 19:15.38 | Katty | and set it to 64 encoding, mono |
| 19:15.39 | jblack | so put it up at filebin.com, please? |
| 19:15.41 | Katty | but it sounded like crap |
| 19:15.43 | Katty | k |
| 19:16.09 | Katty | you want it in stereo or mono? |
| 19:16.32 | jblack | stereo. I'll downmix it |
| 19:16.39 | Katty | k |
| 19:16.58 | darkdrgn2k | ugh any one know how to flash these dam phones :-S i have ciscos website |
| 19:17.02 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 19:17.27 | ManxPower-work | darkdrgn2k: you contact Cisco, send them money, they send you firmware. |
| 19:17.34 | jblack | here's an idea of what I'm getting : http://filebin.ca/xyvmu/ringtones.mp3 |
| 19:17.36 | Katty | ah well crap. i'd already saved it in mono |
| 19:17.40 | darkdrgn2k | ManxPower-work: HAHAHA yeh they wish :-P |
| 19:17.42 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.42 | jblack | mono's fine then |
| 19:17.47 | jblack | tell me what you think of that. |
| 19:17.51 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:17.57 | darkdrgn2k | ManxPower-work: i spend enough money on their Smarth whatever services for all the cisco routers |
| 19:17.59 | ManxPower-work | This whole stupig thing about SELLING firmware is one of the reasons we did not choose cisco. |
| 19:18.00 | Katty | that sounds nice. |
| 19:18.04 | jblack | it's 117 k |
| 19:18.09 | ManxPower-work | ~phones |
| 19:18.10 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 19:18.20 | darkdrgn2k | ->>> Cisco 7940+ |
| 19:18.24 | darkdrgn2k | thats why i got it :-P |
| 19:18.27 | Katty | k, let me edit number 2, and rip out the MJ bit again so it's in stereo |
| 19:18.46 | jblack | Ok. ANd I'll downmix to 16kbit mono just like what you heard. |
| 19:19.15 | jblack | then, maybe some faxing stuff? |
| 19:19.33 | jblack | the guy doing my resume is gonna get pissy if I put him off too long. :) |
| 19:19.33 | Katty | i suppose :< |
| 19:19.53 | Katty | http://filebin.ca/cpnvfn |
| 19:21.25 | jblack | you were satisfied with 16kbit, or you want 32? (102k vs 203k) |
| 19:21.51 | Katty | 203 should be small enough |
| 19:22.05 | jblack | actually, 32 is a bad idea. I'm getting underruns on the file |
| 19:22.10 | jblack | lemme try 24 |
| 19:22.11 | Katty | k |
| 19:22.22 | jblack | only certain bit rates are legal |
| 19:22.38 | Katty | jblack: ghostscript - The GPL Ghostscript PostScript/PDF interpreter <- this? |
| 19:23.00 | jblack | That's the beast there, yeah. |
| 19:23.05 | Katty | apt-gets |
| 19:23.09 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 19:23.32 | Katty | great googa mooga |
| 19:23.32 | [TK]D-Fender | [14:18]<darkdrgn2k>thats why i got it :-P <- Yeah... and notice that its FOURTH on the list |
| 19:23.40 | jblack | that's a bad cut, katty |
| 19:23.46 | *** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 19:23.47 | darkdrgn2k | [TK]D-Fender: but its ON the list... as opposed to NO on the list |
| 19:23.53 | Katty | i did my best :P |
| 19:23.58 | jblack | I can o better. |
| 19:24.02 | Katty | k |
| 19:24.03 | [TK]D-Fender | darkdrgn2k: SMRT |
| 19:24.20 | darkdrgn2k | [TK]D-Fender: *sigh* never dull with you around it is |
| 19:25.29 | Katty | anyone have a document about the specs of a tif file that SendFax() likes? dpi and what not |
| 19:26.10 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp138-119.adsl.forthnet.gr) |
| 19:28.27 | Katty | darkdrgn2k: throw things at him |
| 19:28.49 | Chainsaw | Katty: I'd go for group III fax specifications: 204Ã98 (normal) or 204Ã196 (fine) dots per square inch |
| 19:32.23 | jblack | This is harder than I thought |
| 19:32.42 | *** join/#asterisk traxx (n=traxx@91-64-130-76-dynip.superkabel.de) |
| 19:32.52 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 19:33.10 | *** join/#asterisk canadait (n=canadait@142.59.93.52) |
| 19:37.16 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:37.16 | Katty | http://www.thinkgeek.com/geektoys/rc/bb98/ <- |
| 19:37.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:37.38 | Katty | ^- can you say, Christmas Present |
| 19:37.52 | leifmadsen | Asterisk release candidates 1.4.27-rc5, 1.6.0.18-rc3, 1.6.1.10-rc3, and 1.6.2.0-rc6 are now available. See the release announcement! http://www.asterisk.org/node/49867 |
| 19:38.08 | Chainsaw | Katty: That's awesome :D |
| 19:38.17 | Chainsaw | They'll never let us fly them at work though :( |
| 19:38.34 | Katty | i hope ryan likes it |
| 19:38.40 | Katty | i got him one last year, but it had no pitch control on it |
| 19:39.06 | Chainsaw | 4 channel is quite decent. 2 channel is indeed hard to fly. |
| 19:39.16 | *** join/#asterisk Uhrheber (n=X@p50990c12.dip0.t-ipconnect.de) |
| 19:39.23 | Katty | he's been talking about one called the Bumblebee |
| 19:39.25 | Chainsaw | (6 channel as well, but that's because they're 3D helis and very twitchy. If you don't pay attention you'll crash those very easily) |
| 19:40.30 | Katty | bumblebee appears to be a 2 channel :< |
| 19:40.59 | Chainsaw | Katty: It says 3 channel on this other link I found. But yes, a bit limited. |
| 19:41.08 | Chainsaw | Katty: Besides... airsoft. I approve of this christmas present. |
| 19:42.22 | *** join/#asterisk CGMChris (n=chris@74.143.228.142) |
| 19:42.48 | Katty | oooh,t hey make battletanks too |
| 19:43.31 | CGMChris | In sip.conf, how can I define multiple ip ranges using deny= and permit= statements? I cannot find any examples that clearly illustrate this. For example, I want to permit traffic from 208.227.* and from 209.224.*. |
| 19:43.45 | Chainsaw | Katty: Yes, but you can't sneak up on people as well. |
| 19:44.42 | Chainsaw | Going home, back later. |
| 19:44.51 | [TK]D-Fender | CGMChris: do your deny-all, then 1 to add one subnet, another for the next |
| 19:44.59 | p3nguin | cgmchris: deny=0.0.0.0/0.0.0.0 should come first, then two allow= lines should be fine. |
| 19:45.16 | CGMChris | I tried that and it didnt seem to work. Let me give it another go. Thanks. |
| 19:45.40 | kaldemar | p3nguin: allow is for codecs, permit for ACL |
| 19:45.50 | p3nguin | dammit! You're right. |
| 19:45.53 | p3nguin | Sorry. |
| 19:46.02 | p3nguin | I always mess that up. |
| 19:48.19 | jblack | Katty: mailing it as we speak. I don't think I did any better on the cut. :( |
| 19:48.24 | jblack | but the size is small |
| 19:48.29 | jblack | I'm gonna go grab a burger. |
| 19:48.52 | jblack | and be back in about 22 minutes and 17 seconds |
| 19:49.28 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 19:49.32 | Katty | :> |
| 19:49.48 | jblack | it'll work as a ringtone. ;) |
| 19:50.15 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 19:50.19 | jblack | or are you sad that I'm eating a burger |
| 19:50.27 | Katty | :>>> |
| 19:50.30 | Katty | i love it! |
| 19:50.31 | Katty | cheers |
| 19:50.33 | jblack | is confused! |
| 19:50.49 | jblack | is :> a happy face, or a sad face? |
| 19:50.50 | Katty | ^- the ringtone |
| 19:50.54 | Katty | :< is a sadface |
| 19:51.07 | jblack | So you were sad about the cut, and triple sad about the burger. |
| 19:51.19 | Katty | :>>> <- triple happy |
| 19:51.34 | jblack | Oh, > vs < |
| 19:51.42 | jblack | see. I need food. I'm going blind. :P |
| 19:51.49 | Katty | kbai |
| 19:52.35 | Katty | calls self |
| 19:52.42 | Katty | :>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> |
| 19:53.52 | hardwire | do you sound good? |
| 19:57.20 | [TK]D-Fender | hardwire: Just imagine the EC load ;) |
| 19:59.25 | donnib | stupid question. how does one pause the output when working in terminal ? i have tried | less but it did not work |
| 20:00.12 | hardwire | ctrl-z |
| 20:00.15 | hardwire | err |
| 20:00.16 | hardwire | ctrl-s |
| 20:00.20 | hardwire | now ask how to unpause it |
| 20:00.22 | [TK]D-Fender | donnib: you don't. there is no pause, and "less" only works when the output ENDS |
| 20:00.25 | hardwire | after.. of course trying it. |
| 20:00.57 | donnib | hmm |
| 20:01.01 | hardwire | donnib: I use 'screen' typically |
| 20:01.07 | donnib | so how do i work with output ? |
| 20:01.10 | hardwire | but pausing the vt works well using ctrl-s/ctrl-q |
| 20:01.14 | hardwire | donnib: check out screen |
| 20:01.43 | hardwire | run screen.. connect to asterisk.. hit ctrl-a esc and scroll around or write to a buffer |
| 20:01.54 | hardwire | err.. right the buffer to file |
| 20:03.25 | donnib | thanx |
| 20:03.27 | donnib | will try |
| 20:04.07 | hardwire | [TK]D-Fender: I rock. |
| 20:04.09 | hardwire | neener. |
| 20:05.23 | *** join/#asterisk p3nguin_ (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 20:06.07 | [TK]D-Fender | takes the rock, puts it in a slings and takes out hardwire |
| 20:06.21 | hardwire | bends time and space |
| 20:10.24 | telnettech | can someone tell me what I will see in a protocol analyzer for a phone registration using mgcp? |
| 20:10.24 | jblack | that was good |
| 20:10.33 | telnettech | looking for the message |
| 20:10.58 | ManxPower-work | telnettech: I would be surprised if anyone on this channel uses MGCP |
| 20:10.58 | Nugget | telnet is eeeeeeevil! |
| 20:11.00 | eppigy | TRABJO |
| 20:11.29 | jblack | telnettech: Asterisk will happily dump the protocol chat to you in the console. |
| 20:11.43 | *** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) |
| 20:11.44 | jblack | telnettech: Just set verbose and debug to 9 |
| 20:12.19 | jblack | oh, and sip debug of course |
| 20:12.26 | telnettech | not using asterisk......but im needing to find out the message that is used in the MGCP protocol so that I can build out a C# script |
| 20:12.38 | jblack | ... |
| 20:13.40 | ManxPower-work | telnettech: there is an RFC to PGCP |
| 20:13.49 | ManxPower-work | and MGCP too |
| 20:14.04 | telnettech | i am looking thru it now but was trying to do shortcut :) |
| 20:18.11 | *** join/#asterisk bmg505 (n=leon@196-209-77-55-rndf-esr-5.dynamic.isadsl.co.za) |
| 20:18.11 | Katty | http://www.hobbytron.com/ESKYBeltCP3D6CHRTFElectricRCHelicopter.html <- Better Christmas Present. |
| 20:18.12 | *** join/#asterisk andres833 (n=andres83@166.210.227.171) |
| 20:20.04 | jblack | Katty: So, wanna fix your faxing? |
| 20:20.32 | Katty | well, it's not really broken. |
| 20:20.39 | jblack | Oh. |
| 20:20.56 | Katty | it's just that windows fails at tiffs. |
| 20:20.56 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 20:21.08 | jblack | so it is broken. :) |
| 20:21.10 | [TK]D-Fender | Katty: You had us a "Windows fails" :0 |
| 20:21.50 | jblack | How's this sound? Users save to pdf, upload to a page and plug in a phone number, and fax away? |
| 20:22.14 | Katty | upload to a Page? |
| 20:22.24 | *** join/#asterisk coldsteal (n=Administ@unaffiliated/coldsteal) |
| 20:22.24 | Katty | right now they print to location, and call Sendfax() |
| 20:22.24 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 20:22.28 | [TK]D-Fender | Katty: web page. |
| 20:23.11 | CGMChris | Using IP based auth with a SIP provider, when calling dial for outbound calls in the dialplan, will Dial immediately fallthru to the next line in the dial plan if that particular provider is unreachable (down) ? |
| 20:23.41 | hesco | anyone know off-hand the default username and password for an aastra's configuration web page? |
| 20:24.14 | [TK]D-Fender | hesco: http://www.google.ca/#hl=en&source=hp&q=aastra+default+password&btnG=Google+Search&meta=&aq=f&oq=aastra+default+password&fp=cf2547b2365d1cd0 |
| 20:24.22 | *** part/#asterisk coldsteal (n=Administ@unaffiliated/coldsteal) |
| 20:24.26 | CGMChris | Rephrased, my SIP provider has servers in Dallas, NYC, and LAX. How can I utilize this to provide outbound call redundancy ? |
| 20:25.02 | Qwell | CGMChris: ask them to use SRV, and it'll Just Work. |
| 20:25.13 | *** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com) |
| 20:25.21 | Qwell | it would take them about 5 minutes... |
| 20:26.03 | CGMChris | My "turnup" instructions say SRV all over the place, such as "lax01-01.fs.broadvox.net". But I'm not sure how this affects me and my configs. |
| 20:26.48 | hesco | thank you sir, that got me right in |
| 20:27.19 | CGMChris | QWell: Doesnt my dialplan have to specify which peer to dial out from? like this: exten => _1NXXNXXXXXX,n,Dial(SIP/broadvox_nyc/${EXTEN}) ? |
| 20:28.00 | CGMChris | @Qwell: Do I need to call dial a different way or stack dial commands to have it fallthru to another provider? |
| 20:28.25 | [TK]D-Fender | CGMChris: Use a HOST that isn't an IP <- |
| 20:28.49 | jblack | katty, where did you get sendfax from? |
| 20:28.53 | Qwell | CGMChris: no, it just happens |
| 20:29.33 | J-zimbam | wouldnt qualify in the peer settings let asterisk know if the link is down or not? then just link multiple out routes in ur dial string |
| 20:31.42 | ManxPower-work | You can do this many different ways. You can use SRV or Dial and check the HANGUPCAUSE, etc. |
| 20:31.56 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 20:34.47 | Katty | jblack: ^_- |
| 20:34.53 | Katty | jblack: what do you mean, where did i get it? |
| 20:35.00 | Katty | jblack: i got it from the flower shop, down the street |
| 20:35.04 | *** join/#asterisk jermudgeon (n=jhaustin@216-67-61-242.static.acsalaska.net) |
| 20:35.08 | jblack | Oh, of course |
| 20:35.14 | Blackvel | what would i do on inbound echo (only 1-2%)? throw away snom 370 or patton 4634 media gateway? |
| 20:35.57 | Katty | jblack: if you're asking how did i install it, from the tarballs on digium's site. |
| 20:36.23 | Katty | jblack: http://42ndgeekstreet.blogspot.com/2009/11/asterisk-faxing.html <- further information. |
| 20:36.28 | jblack | I'll be honest, whenever I've done faxing, I used external perl scripts. I did a check in the console for sendfax, and neither show application nor show function shows it. |
| 20:36.46 | Katty | we're not really going to use Sendfax() |
| 20:36.57 | Katty | not when our primary products are 50k MFPs |
| 20:37.15 | Katty | RecieveFax() is more useful with our DIDs |
| 20:38.27 | ManxPower-work | jblack: sendfax is part of 1.6 and requires spandsp to build |
| 20:38.59 | jblack | ok |
| 20:39.10 | Katty | i'm using sendfax with 1.4 |
| 20:39.18 | Katty | it's quirky, but recievefax is good |
| 20:40.17 | jblack | Yeah, I use receive_fax, which is a perl agi. =) |
| 20:40.42 | jblack | simple, but works over sip and iax |
| 20:41.10 | Katty | 1.4 doesn't support sip |
| 20:41.14 | Katty | not sure about iax. |
| 20:41.27 | jblack | I don't think what you use does iax. |
| 20:41.31 | Katty | at least that's what i've read. |
| 20:41.42 | jblack | receive_fax works with sip and iax though. |
| 20:43.14 | ManxPower-work | Katty: I thought you used rxfax/txfax for 1.4. Maybe the 1.6 SendFax was mackported. |
| 20:44.15 | jblack | looks for where he got asterisk-faxreceive from |
| 20:44.50 | Katty | ManxPower-work: no |
| 20:44.59 | ManxPower-work | Here's my fax2email script http://www.fnords.org/~eric/fax2email.txt |
| 20:45.15 | jblack | http://www.lilalinux.net/e-trolley/page_8677/index.html |
| 20:45.18 | Katty | looks |
| 20:45.27 | jblack | That's german, however. |
| 20:46.42 | ManxPower-work | that script has been in use for several years. |
| 20:46.58 | jblack | Yeah. works well for me on incoming. |
| 20:46.58 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 20:47.09 | ManxPower-work | the biggest issue I've found with fax2email is when you get a 50 page fax, the PDF might be too big for many e-mail servers |
| 20:47.21 | ManxPower-work | jblack: I mean MY script |
| 20:47.32 | jblack | Oh. The one I use would have the same problem. :) |
| 20:47.42 | jblack | As it emails pdfs of faxes as well. |
| 20:47.48 | ManxPower-work | I've been meaning to update it to split big faxes. |
| 20:49.19 | ManxPower-work | jblack: My users would not have known a TIFF it it went up to them and kicked them |
| 20:49.31 | defsdoor | Tiff can do that now ? |
| 20:49.34 | defsdoor | awesome |
| 20:49.55 | defsdoor | returns to slumber |
| 20:54.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:58.10 | Katty | hmm. |
| 20:58.12 | Katty | i do believe i'm hungry |
| 20:58.24 | Katty | how annoying |
| 21:18.13 | Tech_Travis | What are the ways I can extract the extension of a member that is logged into a queue for later use in the dialplan? |
| 21:19.19 | *** part/#asterisk Uhrheber (n=X@p50990c12.dip0.t-ipconnect.de) |
| 21:21.03 | KingDavidNYC | help!! I can't get this liknsys 3102 to receive calls |
| 21:21.35 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 21:21.37 | KingDavidNYC | I dont get it, I can receive the calls with on XLite |
| 21:25.46 | CGMChris | My SIP provider and voip-info.org both state Asterisk only has partial SRV support, that it only reads the first SRV entry and ONLY on initial startup. Has this been changed as of 1.4.26.3 ? |
| 21:28.06 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:46.50 | *** join/#asterisk jayrod422 (n=jayrod42@pool-72-95-140-102.pitbpa.fios.verizon.net) |
| 21:49.57 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 21:52.30 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 21:57.33 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 21:57.47 | jtodd | CGMChris: Sorry, I don't have the specific revision behaviors off the top of my head, but it should be fairly easy to test, right? |
| 21:59.11 | *** join/#asterisk jordanb (n=jordanb@adsl-99-21-161-249.dsl.chcgil.sbcglobal.net) |
| 21:59.40 | jordanb | Is it not possible to buy Allison Smith recordings from Digium anymore? |
| 22:00.14 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 22:00.24 | DelphiWorld | hi |
| 22:00.30 | Qwell | jordanb: http://store.digium.com/productview.php?category_id=8&product_code=8IVRPROMPT |
| 22:00.31 | DelphiWorld | is it realy that 1.8 is out? |
| 22:00.45 | russellb | 1.8 has not been released. |
| 22:00.50 | russellb | no branch for 1.8 has been created. |
| 22:00.58 | russellb | The current rough target is sometime in Q2 2010 |
| 22:01.07 | DelphiWorld | ok |
| 22:01.09 | jordanb | Qwell, Thanks |
| 22:01.18 | DelphiWorld | russellb: i heare about 1.8;) |
| 22:01.20 | jordanb | My google-fu must be horrible right now |
| 22:01.44 | russellb | DelphiWorld: I talked about it a little bit on blogs.asterisk.org |
| 22:01.55 | DelphiWorld | russellb: http://blogs.asterisk.org/2009/11/10/asterisk-project-update-astricon-2009/ |
| 22:02.08 | russellb | yes, that. |
| 22:02.22 | DelphiWorld | ok |
| 22:02.23 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 22:02.31 | russellb | bye. |
| 22:02.32 | KingDavidNYC | can anybody please help me with this linksys 3102 why would not it take incoming calls?? |
| 22:06.43 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:06.48 | jordanb | I have a 3102 that I never got working |
| 22:06.52 | jordanb | Oh |
| 22:06.53 | jordanb | :< |
| 22:07.23 | jordanb | I got the FSX working, but not the FXO |
| 22:07.25 | mchou | jordanb: send it to me :) |
| 22:07.37 | jordanb | :P |
| 22:07.58 | jordanb | I still have plans for the FXS port. |
| 22:08.11 | jordanb | Although it has sat in my desk for a year now. |
| 22:08.12 | mchou | jordanb: that's cool |
| 22:08.30 | jordanb | It's also got horrible latency, when compared to tdm cards. |
| 22:08.51 | mchou | just be aware of the double hook flash issue too |
| 22:09.02 | jordanb | Oh? |
| 22:10.07 | mchou | jordanb: yeah |
| 22:10.30 | mchou | you probably wont ever use that "feature" though |
| 22:11.17 | seanbright | anyone know of something like queuemetrics but free? |
| 22:12.38 | *** join/#asterisk [netman] (n=netman@203.Red-88-23-82.staticIP.rima-tde.net) |
| 22:13.33 | Qwell | seanbright: queues.conf |
| 22:13.38 | Qwell | and... |
| 22:13.43 | Qwell | I've got nothing. Carry on. |
| 22:13.50 | seanbright | Qwell: i can always count on you |
| 22:14.00 | Qwell | <3 |
| 22:14.23 | maskas | I'm trying to get asterisk to log hangup cause on answered calls, have used option g in dial command so it goes to the next extension and sets the cdr field, but when I check my cdr in mysql I dont see the hangupcause code and lastapp is dial, not hangup |
| 22:15.52 | *** join/#asterisk ESCulapio__ (n=ESCulapi@Leased-Lines-207-178.tricom.net) |
| 22:16.13 | ESCulapio__ | help my please with [Nov 13 14:12:58] NOTICE[17381]: chan_dahdi.c:11052 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 22:22.12 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 22:23.43 | *** join/#asterisk manxpower (n=ewieling@234.sub-75-254-221.myvzw.com) |
| 22:24.28 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 22:26.22 | *** join/#asterisk saxa (n=sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 22:26.30 | saxa | hello, channel :) |
| 22:26.49 | saxa | how can I do to check why my iax2 connection doesnt see the other end ? |
| 22:27.06 | saxa | I mean I had 2 boxes working and calling each other |
| 22:27.32 | saxa | and now for a certain reason the one is not registering anymore at the other side |
| 22:27.36 | saxa | any ideas ? |
| 22:27.51 | saxa | should nmap show the 4569 port open ? |
| 22:27.54 | maskas | hmmm, whats the etiqutee here for repeating a question? |
| 22:28.29 | saxa | just wanted to explain it well :) |
| 22:29.18 | saxa | everything was working, but now it for some reason doesnt anymore, when I call 55700 which is the extension of the remote box |
| 22:29.33 | saxa | I get the error everything is congested |
| 22:30.04 | saxa | [Nov 13 23:33:12] WARNING[10674]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 22:30.24 | saxa | <PROTECTED> |
| 22:31.53 | saxa | is there a way I can try to dial from the CLI ? |
| 22:32.10 | saxa | from the remote box to the local one ? |
| 22:32.55 | saxa | ok sorry people if I have interuppted some of your conversations |
| 22:33.20 | saxa | will try to find it out alone, excuse me for the disturb |
| 22:44.12 | TJNII | saxa: iax2 set debug on |
| 22:44.33 | *** join/#asterisk icyValk77 (n=icyValk7@host81-155-31-214.range81-155.btcentralplus.com) |
| 22:45.48 | saxa | TJNII: thx, and then ? |
| 22:46.01 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:46.31 | maskas | I'm trying to get asterisk to log hangup cause on answered calls, have used option g in dial command so it goes to the next extension and sets the cdr field, but when I check my cdr in mysql I dont see the hangupcause code and lastapp is dial, not hangup |
| 22:48.00 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 22:49.31 | *** join/#asterisk vty (n=UserNick@216-110-94-245.static.twtelecom.net) |
| 22:49.49 | vty | Does anyone know what "call encountered error code (84)" refers to? Anyone have an error code list? |
| 22:50.03 | Amorsen | Err maskas, are you even using an adaptive cdr module? |
| 22:53.56 | maskas | Amorsen: sorry what does that mean? |
| 22:53.57 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:53.58 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:54.55 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:55.37 | *** join/#asterisk doubletoker (n=lol@adsl-218-186-155.jax.bellsouth.net) |
| 22:55.55 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-10-86.telstraclear.net) |
| 22:57.00 | ChannelZ | Hmm. Are there rules as to what the CallerID Name and Num can contain? |
| 22:57.42 | doubletoker | if anyone is a carrier can you please pm me |
| 22:59.51 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 23:00.24 | drmessano | Isnt that a bit personal? |
| 23:01.53 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 23:02.05 | Nugget | heh |
| 23:08.34 | *** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 23:10.10 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 23:10.10 | *** part/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 23:10.16 | *** join/#asterisk andres833 (n=andres83@200.26.149.196) |
| 23:10.52 | citywok | I'm building an XML voicemail application (visual voicemail), and when i send the voicemail to the phone ot be played it cuases the XML browser to exit. Has anybody else done anything like this / found a solution? I'm using Aastra 57i phones |
| 23:15.33 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 23:16.35 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 23:16.58 | Tech_Travis | What function(s) in 1.6 can read the output from a QUEUE_MEMBER_LIST query? |
| 23:25.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:26.52 | manxpower | Tech_Travis: "core show function ARRAY" (function names are CASE SENSITIVE) |
| 23:33.25 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 23:33.32 | ruben23 | hi |
| 23:33.50 | ruben23 | anyone have idea on this error log on my asterisk server...http://pastebin.com/m44d2d42b |
| 23:35.01 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:38.02 | manxpower | ChannelZ: are you still here? |
| 23:38.17 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 23:38.19 | DelphiWorld | hi |
| 23:38.31 | DelphiWorld | asterisk added support for bv16/32 codecs? |
| 23:40.07 | [TK]D-Fender | DelphiWorld: Do you see an announcement for it and its presence in any branch? |
| 23:40.19 | DelphiWorld | [TK]D-Fender: no, i just asked |
| 23:42.47 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.113.169) |
| 23:46.49 | *** join/#asterisk TimToady_ (n=moi@adsl321-212.kln.forthnet.gr) |
| 23:48.15 | *** part/#asterisk ruben23 (n=RPL@122.55.48.243) |
