| 00:00.07 | cmdln | evening |
| 00:02.34 | cmdln | Im curious if asterisk has what I think is called ringback on my NEC phone system (i know ringback is something else in asterisk) |
| 00:02.45 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 00:02.58 | cmdln | basically what happens is a call gets escelated from one queue to another |
| 00:03.01 | cmdln | like overflow |
| 00:03.08 | cmdln | after a timeperiod |
| 00:03.30 | TJNII | You can program that in your dialplan. |
| 00:06.05 | cmdln | when the call hits the second queue it continues to "ringback" to the first queue and see if someone has become available |
| 00:06.18 | cmdln | sorry phone call mid sentance |
| 00:06.56 | cmdln | Is there any functionality like that? |
| 00:07.37 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 00:08.28 | *** join/#asterisk coppice (n=chatzill@27.202.17.210.dyn.pacific.net.hk) |
| 00:09.24 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:11.23 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 00:13.28 | *** join/#asterisk ftoledo (n=Ragnarok@190.49.204.212) |
| 00:13.32 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 00:13.32 | ftoledo | hi all, |
| 00:15.53 | TJNII | cmdln: So basically a caller calles into a queue, which calls one group of phones. After a timeout, it continues to call those phones along with a second, higher priority group? |
| 00:16.09 | ftoledo | i try to use a System() function to perform an ivr, but commands like "mount" or "eject" does not work and the mountpoint have a user option, the pbx run as "asterisk" on debian testing, maybe i forgot something? |
| 00:17.13 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 00:23.42 | Nasra | hello crowd |
| 00:25.30 | *** join/#asterisk Zeddo (i=user@185.19.100.97.cfl.res.rr.com) |
| 00:25.34 | Zeddo | mmm |
| 00:25.42 | Zeddo | I have a random question and I do not now if asterisk is capable of this? |
| 00:26.10 | Zeddo | Im just going to post it and try to be as clear as possible about it, and hopefully I'll get some kind of answer |
| 00:26.59 | tzafrir_laptop | Asterisk is mostly not capable of answring questions |
| 00:27.17 | Qwell | depends on the question |
| 00:27.23 | coppice | or sometimes even calls |
| 00:27.26 | tzafrir_laptop | People in the channel might be able to answer them, though |
| 00:27.28 | Qwell | Playback(yes) would probably answer a few questions |
| 00:27.51 | lanning | Answer() |
| 00:28.00 | Zeddo | I have a digital phone service and I was wondering if there was a way to run an asterisk box as such: Incoming Phone Line -> Asterisk Box -> Digital Modem (or maybe Incoming Phone Line -> Modem -> Asterisk box -> Phones?) |
| 00:28.22 | Zeddo | Regardless I want to use it for call screening/call block as the local company here does not offer call blocking services |
| 00:28.22 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-35faa9f3c5c8bd43) |
| 00:28.23 | coppice | "The answer you have reached is not available. Please leave your question after the tone...." |
| 00:29.07 | TJNII | Zeddo: You can use a modem but I hear they arn't good for VoIP. Get a ATA with a fxo and fxs port and you're good. |
| 00:29.17 | Zeddo | Does that sound fesiable at all? I'm not quite familiar with the extent of how asterisk works so I was hoping I could get a general idea of what i needed. |
| 00:29.27 | tzafrir_laptop | Zeddo, what exactly do you eman by "modem"? |
| 00:29.48 | Qwell | presumably, analog, since he previously specified digital modem |
| 00:29.51 | Zeddo | actually now that I think about it its Internet -> Modem -> Phone line -> house |
| 00:30.02 | Zeddo | So itd have to come after the modem |
| 00:30.04 | Zeddo | anyways |
| 00:30.11 | TJNII | Zeddo: The basic idea behind what you want to do is doable with *. |
| 00:30.17 | tzafrir_laptop | If you want to connect two Asterisk boxes, normally the simplest way to do that is SIP/IAX (VoIP) |
| 00:31.22 | Zeddo | in simple terms I want to use asterisk to block incoming calls from certain numbers. Probably a relatively simple feature given all that asterisk can do |
| 00:31.31 | Zeddo | im an idiot when it comes to this PBX stuff :( |
| 00:31.54 | coppice | you sound perfect for developing the SIP specs :-) |
| 00:32.29 | Zeddo | i feel like a star wars fan walking into a star trek convention :( |
| 00:32.36 | Zeddo | just a simple software developer here |
| 00:32.38 | Zeddo | be gentle |
| 00:33.13 | Qwell | at least you aren't a Windows user. |
| 00:33.19 | Qwell | right? |
| 00:33.31 | Zeddo | well |
| 00:33.32 | *** join/#asterisk WindBack (n=jorge@host227.190-30-194.telecom.net.ar) |
| 00:33.34 | Zeddo | i use it primarily for gaming |
| 00:33.42 | Zeddo | each operating system has a purpose |
| 00:33.44 | Qwell | uh huh |
| 00:33.44 | Zeddo | its XChat |
| 00:33.45 | Zeddo | Silverex |
| 00:33.46 | Zeddo | Windows |
| 00:33.47 | *** join/#asterisk mike8901 (n=mike@ool-45781de2.dyn.optonline.net) |
| 00:34.10 | Qwell | You should be ashamed of yourself for using xchat on windows :) |
| 00:34.14 | mike8901 | sorry if this is a n00bish question but when you buy a DID from someplace like flowroute.com does it include unlimited incomming calls like you'd get with a home phone? |
| 00:34.16 | Zeddo | I like software thats multiplatform |
| 00:34.25 | Qwell | it isn't multiplatform |
| 00:34.40 | Zeddo | Its supports windows and linux |
| 00:34.47 | Zeddo | multiplatform enough |
| 00:34.48 | tzanger | how is xchat running on windows or linux not multiplatform? |
| 00:34.55 | Qwell | It isn't the same code :) |
| 00:34.56 | tzanger | sure, it's relying on GTK to do most of hte heavy lifting |
| 00:35.01 | tzanger | oh? |
| 00:35.05 | tzanger | I wasn't aware of that |
| 00:35.07 | Zeddo | im sure most multiplatform software doesnt use the same code |
| 00:35.11 | Zeddo | unless you're google and the picasa binary |
| 00:35.22 | Qwell | Zeddo: sure it does |
| 00:35.39 | Zeddo | Regardless |
| 00:35.52 | Qwell | it's also non-free, in both senses of the word |
| 00:35.56 | Zeddo | haha |
| 00:36.01 | Zeddo | GNU software doesnt have to be free |
| 00:36.09 | Zeddo | its legal to sell it in the license as long as you provide the source |
| 00:36.16 | Zeddo | however Im not quite sure XChat is under GNU/GPL |
| 00:36.16 | Qwell | They don't provide the source |
| 00:36.18 | Zeddo | Ah |
| 00:36.20 | Zeddo | evil bastards |
| 00:36.24 | Zeddo | I like the BSD license |
| 00:36.25 | Zeddo | good times |
| 00:36.52 | coppice | Zeddo: BSD licence is good for parasites |
| 00:36.57 | Zeddo | Am I a parasite? |
| 00:36.59 | Zeddo | possibly. |
| 00:37.12 | Zeddo | however I want to use something like asterisk to screen incoming calls and block numbers before it ever reaches my phones |
| 00:37.16 | coppice | if you can't life with GPL, you ought to question why? |
| 00:37.24 | Zeddo | Each license has a unique philosophy |
| 00:37.28 | Zeddo | like that whole slashdot debate |
| 00:37.34 | Zeddo | BSD software is free, GPL software stays free |
| 00:37.41 | mike8901 | anyone know if it's standard for DIDs to come with unlimited incomming calling? |
| 00:37.43 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 00:37.48 | coppice | for the developer BSD licence == spank me |
| 00:38.17 | tzanger | haha |
| 00:38.28 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 00:38.37 | Zeddo | id rather figure out a solution to my phone providers lack of call-block without having to forcivly change my number |
| 00:38.44 | Zeddo | :( not debate on the philosophies of various licenses |
| 00:39.13 | tzanger | Zeddo: do it on your own * box |
| 00:39.15 | tzanger | that's what I do |
| 00:39.27 | tzanger | any empty or invalid caller id = straight to voicemail |
| 00:39.28 | Zeddo | tzanger: I'd love to but im an idiot and have no idea what'd id need to get started. |
| 00:39.36 | Zeddo | Oh i dont want empty/invalid |
| 00:39.42 | tzanger | any callerid which doesn't match a whitelist during dinner = voicemail |
| 00:39.43 | Zeddo | i wanna block specific numbers = straight to hell |
| 00:39.51 | Zeddo | maybe redirect them |
| 00:39.54 | Zeddo | to the gay sex hotline or something |
| 00:39.55 | Zeddo | i dont care |
| 00:39.57 | Zeddo | as long as they leave me alone |
| 00:40.03 | tzanger | any callerid that matches my list rings my cell if the house doesn't answer |
| 00:40.12 | tzanger | any callerid that matches my wife's list does the same ofr her phone |
| 00:40.28 | tzanger | Zeddo: in your incoming context |
| 00:40.35 | tzanger | s/1234567,1,Hangup |
| 00:40.42 | Zeddo | that works |
| 00:40.43 | tzanger | if your incoming line comes ot 's' |
| 00:40.54 | tzanger | or 5551212/1234567,1,hangup if it comes in to 5551212 |
| 00:40.56 | tzanger | and so on |
| 00:41.05 | tzanger | 5551212/,1,... matches no callerid |
| 00:41.17 | tzanger | you're only an idiot if you're unwilling to try |
| 00:41.23 | Zeddo | im not unwilling to try |
| 00:41.24 | tzafrir_laptop | 5551212,1,...; even |
| 00:41.35 | tzanger | tzafrir_laptop: no, that matches anything |
| 00:41.40 | tzanger | (i.e. no callerid-based routing) |
| 00:41.50 | Zeddo | i have no idea what hardware im going to need |
| 00:41.53 | Zeddo | i mean in my simplistic world |
| 00:42.00 | tzanger | how is the number coming in to your * system |
| 00:42.14 | Zeddo | a phone modem would do the trick since i just want to identify whos calling, hang up, or ring my house |
| 00:42.16 | Zeddo | but |
| 00:42.17 | Zeddo | no clue |
| 00:42.20 | Zeddo | im here for a start |
| 00:42.29 | tzanger | Zeddo: ah, so you have a phsyical phone line |
| 00:42.31 | Zeddo | yes |
| 00:42.37 | Zeddo | well its "Digital Phone" |
| 00:42.43 | tzanger | define "digital phone" |
| 00:42.53 | Zeddo | it actually comes through my internet, goes into some kind of digital modem, and then into my house |
| 00:43.11 | Zeddo | the modem converts it from digital to analog |
| 00:43.14 | tzanger | well if it presents itself like a normal phone jack that you plug a normal phone in to, it's a telephone line :-) |
| 00:43.15 | Zeddo | i guess would be the best way ot put it |
| 00:43.25 | tzanger | but honestly see if you can't get the number as a SIP call or osmething standard |
| 00:43.28 | tzanger | and eliminate that box |
| 00:43.35 | tzanger | all depends |
| 00:43.37 | Zeddo | The phone companies down here think we are retarded |
| 00:43.48 | Zeddo | i mean i could always just skip the digital part and catch the signal once its converted to analog |
| 00:43.50 | [TK]D-Fender | Katty: Mew. |
| 00:43.53 | tzanger | rogers (cable modem isp here) gives you a similar unit, but it's really a modified cable modem |
| 00:43.57 | Zeddo | go from the modem to the asterisk box to the house |
| 00:44.01 | Zeddo | thats what we have here tzanger |
| 00:44.04 | Zeddo | a modified cable modem |
| 00:44.16 | tzanger | since their VOIP is carried over a different "channel" than their internet (excellent separation of regualr internet use and voip) |
| 00:44.24 | Zeddo | I believe thats how its done here as well |
| 00:44.24 | [TK]D-Fender | Dovid: And I've got martial arts on Tuesdays & Thursdays, so yeah I may be "on", I' might not jsut be "here" |
| 00:44.25 | tzanger | then you're pretty much stuck with it for now :-) |
| 00:44.35 | Zeddo | but whatever |
| 00:44.46 | Zeddo | cant i do something once the signal is converted to analog before it reaches my house/phones? |
| 00:44.48 | *** part/#asterisk ftoledo (n=Ragnarok@190.49.204.212) |
| 00:44.51 | Zeddo | like I said a simple voice modem |
| 00:44.51 | tzanger | Zeddo: so you want an x100/x101 type device (really it's a specific winmodem, motorola or intel chipset, check voip-info.org for the compatibiliyt list) |
| 00:44.56 | tzanger | yes |
| 00:45.00 | Zeddo | i like you |
| 00:45.07 | tzanger | plug the line in to your x100/x101 |
| 00:45.11 | [TK]D-Fender | tzanger: Vo Cable is basically over their frame interface VS straight IP making it more "secure" if you will. |
| 00:45.17 | tzanger | use an ATA to convert it out to a regular line again |
| 00:45.37 | tzanger | you can also get a TDM400 with 1FXS and 1FXO module, but tha'ts pricier I would think |
| 00:45.54 | Zeddo | see now you're talking latin |
| 00:46.07 | tzanger | [TK]D-Fender: yeah rogers has the home phone connection on a completely separate "channel" pair so there is zero interference |
| 00:46.31 | tzanger | Zeddo: FXS interface = something you plug phones in to. FXO interface = something you plug phone lines in to |
| 00:46.36 | Zeddo | Ahahahaha |
| 00:46.37 | Zeddo | Simple |
| 00:46.40 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 00:46.42 | Zeddo | Thanks for dumbing it down :( |
| 00:46.43 | tzanger | think S = station or set, and O = central office |
| 00:46.55 | Zeddo | regardless the x100/x101 would be the cheaper solution? |
| 00:46.56 | tzanger | voip-info.org has lots of info to help get you up to speed |
| 00:47.21 | tzanger | well the x100 or x101 cards are cheap, and if you're lucky you may alreayd have a compatible device (as I said it's just a specific-chipset winmodem) |
| 00:47.27 | tzanger | but that is an FXO interface |
| 00:47.28 | Zeddo | I have a bunch of old winmodems |
| 00:47.33 | tzanger | that lets you get the call IN to the system |
| 00:47.40 | Zeddo | then to send it out of the system to my house |
| 00:48.07 | Zeddo | thats what the FXS would be for? |
| 00:48.10 | tzanger | check them -- there is a GREAT DEAL of winmodems that will work. However the actually "front end" circuitry that connects them to the phone line varies GREATLY, so your results may be mixed |
| 00:48.15 | tzafrir_laptop | Zeddo, sadly most modems will not do. Only some very specific ones |
| 00:48.22 | Zeddo | i have a fuckton of older hardware |
| 00:48.24 | mike8901 | FXS would be using an analog phone line for a voip system |
| 00:48.25 | Zeddo | thats where im hoping ill be in luck |
| 00:48.27 | Zeddo | ah |
| 00:48.30 | mike8901 | er sorry |
| 00:48.32 | tzanger | FXS gets you a way to convert a VOIP call into something that will ring regular old phones |
| 00:48.37 | Zeddo | <PROTECTED> |
| 00:48.39 | mike8901 | FXS is what you'd plug a fax machine into |
| 00:48.47 | tzanger | motorola chipsets were the original x100s, and Intel were the x101s IIRC |
| 00:48.49 | mike8901 | FXO is what you'd plug an analog phone line into |
| 00:49.01 | Zeddo | mm |
| 00:49.02 | mike8901 | anyone know if DIDs usually come with unlimited incomming calling? |
| 00:49.10 | mike8901 | or are you usually charged for incomming? |
| 00:49.19 | tzanger | an ATA (analog telephony adapter) can be FXS, FXO or both... tons of different units |
| 00:49.39 | Zeddo | tzanger: so id need one or two of these modems, run the phoneline from the Digital to Analog Converter (Modified Cable Modem) |
| 00:49.47 | Zeddo | to the x100/x101 |
| 00:49.49 | Zeddo | then to my house? |
| 00:49.50 | tzanger | basically they convert SIP (a type of VOIP) calls into an analog call and vice-versa |
| 00:49.59 | tzanger | Zeddo: precisely. |
| 00:50.02 | Zeddo | makes sense |
| 00:50.08 | Zeddo | see |
| 00:50.10 | Zeddo | you idiot proofed it |
| 00:50.14 | Zeddo | now if you cut off all the corners id be home free |
| 00:50.21 | tzanger | [cable feed] -> VOIP adapter from Cable co -> x101 |
| 00:50.28 | Zeddo | x101 -> house ? |
| 00:50.33 | Zeddo | and bam |
| 00:50.34 | tzanger | and then ethernet -> ATA -> one or more analog phone lines -> phones |
| 00:50.35 | Zeddo | things ring |
| 00:50.37 | tzanger | no |
| 00:50.44 | Zeddo | oh wait what ethernet -> ATA ? now you're caring me |
| 00:50.47 | Zeddo | scaring |
| 00:50.51 | tzanger | x101 connects to the "phone line" and that' sit |
| 00:50.51 | Zeddo | ill do some reading however |
| 00:51.04 | tzanger | think of it as a phone for a computer |
| 00:51.06 | Zeddo | i was trying to bypass all that by just doing everything once the signal has been converted to analog by the modem |
| 00:51.13 | tzanger | cable co rings the line, the computer hears it and answers |
| 00:51.20 | Zeddo | but |
| 00:51.26 | Zeddo | again im an idiot so it halfway is making sense |
| 00:51.31 | tzanger | an FXS ATA takes the call from Asterisk and rings the house phone |
| 00:51.37 | Zeddo | mm |
| 00:51.40 | Zeddo | okay that makes sense |
| 00:51.52 | tzanger | you put the computer in the middle to make the decision to ring the inside phones or not |
| 00:52.11 | Zeddo | So aside from the x100/x101 modem what else am i going to need/ |
| 00:52.20 | tzanger | now if you get a multi-port FXS ATA, you can have a "kitchen extension" a "rec room extension" a "garage extension", hell a "door phone" etc |
| 00:52.24 | tzanger | all totally separate |
| 00:52.26 | Zeddo | haha |
| 00:52.29 | Zeddo | good times |
| 00:52.36 | tzanger | with some magic you can use the phones as an intercom |
| 00:52.42 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:52.42 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:52.43 | tzanger | I do that here... wife rings the office to ask if I'm coming to bed :-) |
| 00:52.45 | Zeddo | also one last question would be |
| 00:52.55 | Zeddo | what kind of specs would an asterisk box need |
| 00:52.58 | Zeddo | i would think very minimal |
| 00:53.00 | Zeddo | especially for this |
| 00:53.07 | Zeddo | im just looking to drop/block/redirect incoming calls from numbers i hate |
| 00:53.12 | tzanger | yes very minimal |
| 00:53.15 | Zeddo | mmm |
| 00:53.16 | Zeddo | define minimal |
| 00:53.21 | Zeddo | i have a 233mhz laying under a kitchen sink |
| 00:53.23 | Zeddo | or we could go 1.4 GHZ |
| 00:53.23 | tzanger | I had a fully functioning asterisk system out of an old P90 (no MMX) |
| 00:53.27 | Zeddo | hahaha |
| 00:53.31 | Zeddo | okay so "minimal" |
| 00:53.33 | Zeddo | gotcha |
| 00:53.37 | tzanger | it oculdn't do music on hold or anything fancy, but it worked for a simple multi-line telephone system |
| 00:54.01 | tzanger | yep 233MHz would work just fine for this |
| 00:54.40 | Zeddo | as confusing and new as this is all to me |
| 00:55.02 | Zeddo | as much as i hate these calls |
| 00:55.04 | Zeddo | im going to figure it out |
| 00:55.41 | tzanger | once you get in to it it becomes second nature |
| 00:55.48 | Zeddo | well |
| 00:55.50 | Zeddo | i already have a BSD router |
| 00:55.53 | Zeddo | maybe I can run asterisks on that |
| 00:56.02 | Zeddo | it already sits under the table with modem among everythinge lse |
| 00:56.13 | Zeddo | and the phone line that needs to be routed or whatever |
| 00:56.21 | tzanger | you could call your house and have it unlock your front door :-) |
| 00:56.33 | Zeddo | yeah no thats too much |
| 00:56.36 | Zeddo | haha |
| 00:56.38 | Zeddo | just block calls |
| 00:56.41 | Zeddo | then maybe as i get into it |
| 00:56.43 | Zeddo | who knows |
| 00:57.11 | tzanger | might want to start with debian packages or something so you aren't trying to compile all day on a 233MHz system :-) |
| 00:57.16 | Zeddo | ah |
| 00:57.18 | Zeddo | the BSD box (router) is overkill |
| 00:57.21 | tzanger | oh... and use a UPS for the asteirsk box, voip modem from cable co and ATA :-) |
| 00:57.22 | Zeddo | its around 1200MHZ |
| 00:57.29 | Zeddo | it was the only thing I had at the time |
| 00:57.32 | Zeddo | when my original router blew up |
| 00:57.44 | Zeddo | so i have plenty of power for asterisk to run on the BSD box |
| 00:57.45 | tzanger | funny thing is... new hardware is generally much better on power use, sadly |
| 00:58.10 | tzanger | I absolutely hate throwing perfectly good hardware away, but it is a waste of my money to run it compared to new kit |
| 00:58.19 | Zeddo | Does asterisk exist in the BSD ports collection? |
| 00:58.22 | tzanger | probably |
| 00:58.25 | Zeddo | just curious |
| 00:58.29 | Zeddo | :) |
| 00:58.36 | tzanger | not sure about zaptel though (which is what is needed for the x101p) |
| 00:58.48 | tzanger | or |
| 00:58.52 | Zeddo | ill figure something out |
| 00:58.56 | tzanger | just get a 1FXS, 1FXO ATA (sipura makes these) |
| 00:58.59 | tzanger | and you have it all in one box |
| 00:59.02 | Zeddo | oh boy |
| 00:59.03 | Zeddo | link me |
| 00:59.06 | Zeddo | thatll handle everything? |
| 00:59.24 | tzanger | hell, some of those devices might have enough brains to be able to do the callerid-based-routing on their own,w ith absolutely no phone server |
| 00:59.33 | tzanger | I don't use ATAs myself (not yet anyway) |
| 00:59.36 | tzanger | SPA-941? si that one? |
| 00:59.47 | Zeddo | :( wish I knew |
| 00:59.56 | tzanger | no a spa-941 is a voip phone |
| 00:59.58 | tzanger | hmm |
| 01:00.24 | *** join/#asterisk iamthelostboy (n=chatzill@190.92.6.2) |
| 01:00.50 | tzanger | linksys makes them too |
| 01:00.57 | tzanger | hell linksys makes a wireless router+ata |
| 01:02.13 | tzanger | there |
| 01:02.24 | tzanger | linsys spa3102 |
| 01:02.29 | tzanger | linksys |
| 01:02.33 | Zeddo | mm |
| 01:02.37 | Zeddo | im looking at ports right now |
| 01:02.40 | Zeddo | im going to compile asterisk |
| 01:02.42 | Zeddo | and run with it from there |
| 01:02.43 | tzanger | gives you 1fxo (you plug that in to your cable voip thinger) and 2fxs |
| 01:02.50 | *** join/#asterisk moy (n=moy@189.169.208.231) |
| 01:02.50 | tzanger | you use one of those for your inside phones |
| 01:03.09 | tzanger | it also does routing and shit, but depending on what you do yourself, it may not be enough |
| 01:03.13 | tzanger | (for routing) |
| 01:03.32 | tzanger | I always think sipura makes these because of the spa :-) |
| 01:03.44 | Qwell | they do make them |
| 01:03.54 | Qwell | or...did, before linksys bought them :p |
| 01:03.59 | tzanger | the really, really nice thing about having FXS+FXO in one unit is that they almost all do "power fail direct connect" |
| 01:04.14 | tzanger | so if they lose power, your internal phones are connected directly to the incoming phone line |
| 01:04.24 | tzanger | a lot of them also do 911 direct connect |
| 01:04.38 | tzanger | so you don't lose 911 if your asterisk box dies :-) |
| 01:04.42 | tzanger | Qwell: that would explain it! |
| 01:06.36 | Zeddo | tzanger: your information has been usefu L:) |
| 01:06.37 | tzanger | grandstream (known for cheap but not necessarily good), linksys, dlink.. all make these kinds of things |
| 01:06.50 | Zeddo | i will be around very very shortly once getting asterisk installed on this BSD box |
| 01:06.52 | Zeddo | seeing how |
| 01:06.57 | Zeddo | i wont be able to do much after that :( |
| 01:07.12 | tzanger | just beware of the cheap cheap stuff that is connected to vonage; they may or may not be able to be unlocked |
| 01:07.32 | tzanger | Zeddo: if you're using a SIP ATA (what I've been talking about for hte last half hour) you don't need to get rid of bsd, just install asterisk on your bsd router |
| 01:07.42 | Zeddo | tzanger: thats what i gathered |
| 01:07.48 | Zeddo | thast why im installed asterisk on my BSD router |
| 01:07.50 | Zeddo | thats ^ |
| 01:07.51 | Zeddo | rather |
| 01:08.06 | Zeddo | im updating my packages, and from there im going to install asterisk |
| 01:08.10 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 01:10.24 | tzanger | if you keep it all SIP it makes things real easy for you, but again... UPS on the cable modem thing, ATA and asterisk box/bsd router |
| 01:16.52 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:20.45 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 01:22.06 | *** part/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 01:22.15 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 01:22.24 | *** part/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 01:27.06 | raz | is there a way to embed the callerid (remote number) into the filename of a recorded message? |
| 01:27.21 | *** join/#asterisk xpl` (n=xpl@84.126.207.62.dyn.user.ono.com) |
| 01:28.39 | xpl` | hola |
| 01:29.01 | xpl` | algun espaniol por aqui ? |
| 01:29.06 | [TK]D-Fender | raz: You choose the name, so, yes |
| 01:29.32 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 01:29.59 | raz | [TK]D-Fender, what would be the variable? i tried Record(mailbox-${CALLERID}-%d:wav|5|120) but that gives me a blank string |
| 01:31.06 | xpl` | alguien abla espaniol por aqui ? |
| 01:32.10 | *** join/#asterisk bsaxon (n=bsaxon@182.sub-75-250-84.myvzw.com) |
| 01:33.42 | jaytee | no hablo espanol |
| 01:33.49 | xpl` | :) |
| 01:34.04 | *** part/#asterisk bsaxon (n=bsaxon@182.sub-75-250-84.myvzw.com) |
| 01:34.36 | xpl` | ablas o no ? |
| 01:34.49 | [TK]D-Fender | raz: because that variable no longer exists as of * 1.4 |
| 01:35.06 | [TK]D-Fender | raz: go read through the list of "core sho functions" and I'm sure it will jump out at you. |
| 01:35.38 | [TK]D-Fender | xpl`: do /join #asterisk-br |
| 01:35.50 | [TK]D-Fender | raz: "core show functions" |
| 01:36.09 | bijit | yo hablo espaƱol |
| 01:36.30 | raz | [TK]D-Fender, aaah got it. thx! :) |
| 01:36.44 | bijit | hola |
| 01:36.58 | xpl` | hola |
| 01:37.24 | xpl` | bijit me puedes contestar en privat ? |
| 01:38.15 | bijit | xpl`: yo no se mucho de asterisk. Apenas estoy empezando con est. |
| 01:38.44 | [TK]D-Fender | xpl`: do /join #asterisk-br <--------------- |
| 01:38.54 | xpl` | br ? |
| 01:38.57 | xpl` | why br ? |
| 01:39.00 | xpl` | es maybe |
| 01:39.08 | jaytee | si |
| 01:39.30 | jaytee | #asterisk-br es espanol |
| 01:39.47 | jaytee | or close enough |
| 01:40.28 | iamthelostboy | hi.. i need to connect an asterisk system into an existing panasonic pabx. the panasonic has 4 isdn lines which i want to leave where they are. I have some spare lines on the panasonic, so i could just connect them into a tdm card, though then i cant transfer from the panasonic system to the asterisk system too well.. any ideas? |
| 01:40.39 | [TK]D-Fender | br <- Brazil (spanish) |
| 01:40.45 | xpl` | :)))))))))))))))))))))))))))))))))))))))))) |
| 01:40.59 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.88) |
| 01:41.00 | xpl` | if you say so |
| 01:41.08 | [TK]D-Fender | xpl`: they say so :) |
| 01:41.46 | [TK]D-Fender | iamthelostboy: How would an analog phone on tht port transfer a call? |
| 01:41.49 | tzanger | iamthelostboy: you could put asterisk between the telco and the pbx pretty easily, I did that at a company used to work for |
| 01:41.55 | xpl` | yeah... but the people on brazil dont speak spanish :P |
| 01:42.17 | tzanger | the problem is of course that asterisk looks like the phone system, so without some magic you can't have sip extensions or anything acting like extensions ot the panasonic |
| 01:42.19 | [TK]D-Fender | iamthelostboy: And what role do you want * to fill on it? |
| 01:43.49 | iamthelostboy | [TK]D-Fender: asterisk is going to support a new office in the same building, and slowly merge out to new staff, taking over completely later... |
| 01:44.13 | tzanger | iamthelostboy: have fun; it's not gonna be pretty. I straddled that line with Nortel |
| 01:44.21 | [TK]D-Fender | iamthelostboy: ok, what is * doing in the interim? |
| 01:44.55 | iamthelostboy | i wanted to try to leave the isdn lines where they are on the panasonic for the moment, and move them later when we merge completely |
| 01:45.09 | iamthelostboy | asterisk hasnt been built yet at all.. |
| 01:45.48 | iamthelostboy | im using it at our other office, and at home.. ive linked them by iax2, which i will also link into the 3rd asterisk server which is the one next to the panasonic pabx |
| 01:46.45 | iamthelostboy | id like all sites to be able to dial out through the isdn lines at the head office as getting phone lines is a bit difficult at the second site, and getting SIP trunking is proving to be difficult |
| 01:47.18 | iamthelostboy | its all a bit of a hack at the moment,, slowly will get it tidied |
| 01:47.31 | jaytee | iamthelostboy, I started that way with a Nortel switch. I used extensions off of it into fxo ports on a tdm04b in *. Then I put * in between my telco and the Nortel using a TE212P card. |
| 01:48.10 | jaytee | some tricky dialplan logic and I've got 4 digit extension dialing between the two. |
| 01:57.31 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 02:01.05 | Katty | oh. look. people.! |
| 02:01.41 | jaytee | where? |
| 02:02.15 | Katty | over there! |
| 02:02.38 | BeeBuu | :) |
| 02:03.06 | Katty | sadly, not the one person i'm actually looking for )= |
| 02:04.59 | tzanger | jaytee: that's pretty much exactly what I did as well |
| 02:05.20 | tzanger | Katty: you need to lower your standards. :-) |
| 02:05.29 | jaytee | sadly, and who would that be? |
| 02:05.44 | Katty | tzanger: avast! |
| 02:05.55 | tzanger | me hearty? |
| 02:05.58 | tzanger | arrrrr |
| 02:06.02 | tzanger | oh wait, it's not pirate day yet |
| 02:06.05 | Katty | yoo hoo hoo, and who brought the rum? |
| 02:06.17 | Katty | alas, i'm looking for krice. |
| 02:07.19 | tzanger | never heard of him |
| 02:07.27 | tzanger | rum? bleh, I've got some whisky |
| 02:07.33 | Katty | never heard of him?! |
| 02:07.35 | Katty | that's ken |
| 02:07.36 | Katty | rice |
| 02:07.42 | Katty | he's here sometimes :< |
| 02:07.47 | Katty | maybe that was months ago. hmm. |
| 02:08.35 | _ShrikE | Katty: isnt he SwK? |
| 02:08.49 | Katty | SwK: ping. |
| 02:08.51 | Katty | we'll see. |
| 02:08.57 | _ShrikE | check on #freeswith |
| 02:09.03 | tzanger | I wish I could send audio with ping replies |
| 02:09.05 | _ShrikE | err freeswitch |
| 02:09.10 | Katty | yeah i'm there too. |
| 02:09.11 | tzanger | ping... BARROWWWOOROWWOONNGGGG |
| 02:09.17 | Katty | hehe. |
| 02:09.20 | tzanger | or ping... Ah-WOOOOOOOOOOOO-gah |
| 02:09.55 | Katty | tzanger: i pronounce you king ircnoise. |
| 02:10.03 | tzanger | gezundheit |
| 02:10.08 | Katty | thank you. |
| 02:10.38 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 02:12.10 | alancio | how can I improve the sound quality of my calls? For example, my datetime application sounds really bad on the phone |
| 02:12.37 | Katty | you're not using el cheapo phones are you? |
| 02:12.47 | alancio | I have tried with many phones |
| 02:13.13 | alancio | and if I try the datetime extension with FreeworldDialup (ext 612) it sounds much better |
| 02:13.45 | jaytee | try using a different codec |
| 02:13.50 | alancio | I am using ulaw, I don't see anything to configure in there |
| 02:13.58 | alancio | and FWD uses ulaw too |
| 02:14.53 | [TK]D-Fender | alancio: You'll have to provide real detail as to what you're working with... |
| 02:14.54 | jaytee | if you installed all the sound files then there will be ulaw, wav and gsm versions of each sound file. Switching preferred codecs on the phone will make * match the codec used by the phone to the correct format sound file. |
| 02:15.35 | alancio | ok, so maybe FWD has the files already encoded in ulaw and I don't? |
| 02:16.15 | alancio | and maybe the realtime conversion is not as good as having the files already there |
| 02:16.44 | jaytee | or you're using Grandstream phones |
| 02:17.03 | alancio | right now I'm using a software phone, called twinkle |
| 02:17.10 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
| 02:17.15 | alancio | but I also have grandstream, snom and linksys phones |
| 02:17.36 | jaytee | and does it exhibit the same behavior on all the phones? |
| 02:18.14 | alancio | yes |
| 02:18.27 | jaytee | and all the phones are set to use ulaw? |
| 02:18.28 | alancio | also the voicemail pre-recorded sounds are terrible |
| 02:18.39 | alancio | but when I leave a message and hear it back, it sounds Ok |
| 02:19.02 | alancio | I'll have to change my sound files |
| 02:19.04 | jaytee | ok, check the contents of /var/lib/asterisk/sounds |
| 02:19.19 | [TK]D-Fender | alancio: First guess is that you compiled * with a bad ver of GCC which ahs been known to corrupt GSM transcoding. |
| 02:19.24 | Katty | [TK]D-Fender: ohai |
| 02:19.26 | jaytee | and see if there are multiple files with different extensions |
| 02:19.32 | *** join/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 02:19.36 | [TK]D-Fender | alancio: make sure to compile with GCC 4.1 or lower |
| 02:19.40 | [TK]D-Fender | Katty: Mew. |
| 02:19.48 | Katty | [TK]D-Fender: how are mew? |
| 02:20.20 | [TK]D-Fender | Katty: not bad. Trying to sell my piano and old camera, and shopping for a lens for my new one. |
| 02:20.34 | alancio | mmm I guess thats it, because I always use the latest GCC version |
| 02:20.45 | Katty | [TK]D-Fender: i want a piano... |
| 02:20.49 | Katty | [TK]D-Fender: do you deliver and tune? |
| 02:20.53 | [TK]D-Fender | alancio: Explains why its the prerecorded stuff that sounds like crap |
| 02:21.09 | [TK]D-Fender | Katty: its a Roland HP-137 digital piano... it doesn't NEED tuning :) |
| 02:21.18 | [TK]D-Fender | Katty: Go Google it :) |
| 02:21.23 | Katty | oh. psh. |
| 02:21.26 | Katty | i want a /real/ piano. |
| 02:21.30 | jaytee | full 88 keys? |
| 02:21.31 | Katty | with /real/ defects. |
| 02:21.52 | alancio | [TK]D-Fender: so when one installs asterisk, the sound files are transcoded? I thought they just come in the tarball |
| 02:22.56 | [TK]D-Fender | alancio: in the tarball, they come in GSM. When you compile * the GSM codec that gets compiled gets optimized bad or something which is why when you call using ulaw/alaw/etc, * transcodes it and turns it into crap |
| 02:23.28 | [TK]D-Fender | Katty: If you'll pay up front I can make it defective for you ;) |
| 02:23.30 | alancio | oh I see, thanks |
| 02:24.49 | Katty | [TK]D-Fender: >.< |
| 02:25.21 | jaytee | [TK]D-Fender, in my sounds directory I've got 3 copies of each sound file with ulaw extension, gsm extension and wav. Are the ulaw and wav files created during install or unpacked? I thought when I selected all my options and ran make part of the install process downloaded the 3 different file types as tarballs. |
| 02:25.54 | [TK]D-Fender | jaytee: downloaded on "make" |
| 02:26.10 | [TK]D-Fender | jaytee: they are not transcoded by your system |
| 02:26.44 | jaytee | so he didn't choose all the options and all he's got is gsm then which forces a transcode? |
| 02:26.48 | *** part/#asterisk spokra (n=spokra@74-61-42-127.sea.clearwire-dns.net) |
| 02:27.08 | raz | hrmm.. what does asterisk do when someone hangs up during Record()? i'm using a System()-call right after the recording to send it to myself. this does not happen when the caller hangs up instead of pressing #. how can i make it happen either way? |
| 02:28.28 | [TK]D-Fender | jaytee: something like that most likely. |
| 02:28.40 | [TK]D-Fender | jaytee: the results and his testimony confirm it. |
| 02:28.53 | mog | raz asterisk hops to h exten on hangup |
| 02:29.02 | mog | it wont go to next step in dialplan |
| 02:29.14 | raz | aah |
| 02:29.36 | raz | okay then i just need to put my conditional there, thx :) |
| 02:29.43 | mog | np |
| 02:30.36 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 02:31.14 | Katty | hai mog |
| 02:31.19 | mog | hi katty |
| 02:31.28 | Katty | [TK]D-Fender: so when are you delivering my piano |
| 02:32.02 | [TK]D-Fender | Katty: After you've paide for it and arranged your own shipping :) |
| 02:32.13 | Katty | boo! |
| 02:35.07 | jaytee | I just watched No Country For Old Men for the first time, it kinda reminded me of #trixbox because it was full of dead or dying people and left you with lots of unanswered questions. |
| 02:36.05 | DigitalIrony | lol |
| 02:39.11 | marc7 | hah |
| 02:39.23 | jaytee | damn, I've managed to setup my system so I can use Exchange UM's auto attendant for voice recognition dial-by-name directory but I'm having to front end that with an * IVR and I'm using Lumenvox for speech recog. Just configuring a grammar for the top level menu is giving me a brain tumor. |
| 02:40.02 | *** join/#asterisk tengulre (n=tengulre@125.69.124.131) |
| 02:40.21 | raz | hrm |
| 02:40.21 | jaytee | "Mav, you got the number for that truck driver's school? What's the name of it, Truckmasters or something?" |
| 02:40.32 | Strom_M | thats funny...stress usually causes aneurysms, not tumors |
| 02:40.44 | jaytee | oh, thanks!!!!! |
| 02:40.44 | raz | asterisk is awesome. just sayin. :) |
| 02:41.31 | jaytee | won't argue with you there |
| 02:41.49 | raz | only thing i'm missing would be a skype hook so i can receive skype calls on my sip phone |
| 02:41.59 | raz | but that's a pipe dream i guess (skype being proprietary) |
| 02:42.05 | jaytee | raz, so write one! |
| 02:42.14 | raz | hehe |
| 02:42.26 | raz | that's way outta my skill level |
| 02:43.13 | jaytee | just get six of your best geek/nerd friends over, some extra computers and a case or two of Jolt cola, some Hot Pockets and have at it. |
| 02:43.26 | Katty | diarreah pocket... |
| 02:43.34 | jaytee | eeeewwwwwwww |
| 02:43.35 | raz | hm. something makes me think the skype guys would not like that - and block it. |
| 02:43.47 | seanbright | i can't spell diarreah either |
| 02:43.49 | jaytee | Katty, I thought that was burritos |
| 02:43.57 | Katty | it's actually a commedian skit. |
| 02:43.59 | jaytee | diarrhea |
| 02:43.59 | Katty | youtube it. |
| 02:44.02 | seanbright | diarhea |
| 02:44.09 | seanbright | cha cha cha |
| 02:44.27 | Katty | you so googled that. |
| 02:44.36 | seanbright | not the 'cha cha cha' part |
| 02:44.41 | raz | the two mtv guys |
| 02:44.46 | raz | beavis & butthead |
| 02:44.47 | seanbright | raz: correct |
| 02:44.50 | Katty | no. |
| 02:44.54 | seanbright | and it has two 'r's |
| 02:44.57 | raz | i know the song, i even made a remix way back |
| 02:44.58 | seanbright | diarrhea |
| 02:45.05 | seanbright | jaytee wins |
| 02:45.11 | Katty | Jim Gaffigan |
| 02:45.18 | jaytee | no, I remember from playing Pandemic II last night. I completely wiped out Greenland in less than 3 days with a virus that spread quick and then erupted with hemorraghic fever after mutating to airborne. |
| 02:45.37 | seanbright | katty, gaffigan = diarrhea pockets? |
| 02:45.52 | Katty | yes. |
| 02:46.03 | seanbright | 'diarrhea cha cha cha' = beavis & butthead |
| 02:46.29 | [TK]D-Fender | jaytee: Dammit, the Ausiies and Peru survived mine! |
| 02:46.51 | [TK]D-Fender | jaytee: I went for maximum spread, but they close off their borders... |
| 02:47.12 | [TK]D-Fender | jaytee: Couldn't get it. I waited till I hit a high infection rate before going turbo-lethal |
| 02:47.16 | raz | http://mbox.bz/s/slrp_cocough.ogg <-- my b&b remix from way back when i was young and had a loooot of time to waste xD |
| 02:47.32 | jaytee | [TK]D-Fender, the key is to sell any symptoms and just let it spread benignly till it's hit most major countries while buying resistance to cold, heat, moisture, then when you've spread far and wide it's time to mutate to the next level. |
| 02:48.01 | [TK]D-Fender | jaytee: have you ever killed EVERYBODY? |
| 02:48.21 | jaytee | what makes me laugh is even with everyone in Greenland dead the airports were still open. |
| 02:48.23 | seanbright | Patriot Act Monitor: Enabled |
| 02:48.47 | jaytee | [TK]D-Fender, not yet but I've killed over 5.7 billion in one game. |
| 02:49.02 | jaytee | Madagascar was the only place left uninfected. |
| 02:49.18 | jaytee | seanbright, lol |
| 02:50.03 | [TK]D-Fender | jaytee: I got all but 2 countries on mine... |
| 02:50.11 | [TK]D-Fender | jaytee: Oh yeah, there too :) |
| 02:50.20 | jaytee | seanbright, right now they're getting feeds from their Tarus STA-6 Semantic Traffic Analyzers and printing them out for tomorrow's daily intelligence brief. |
| 02:51.04 | jaytee | Madagascar frikken sucks |
| 02:53.29 | jaytee | [TK]D-Fender, that game is kind of addictive but it scares me. I picture one of the kids in high school that got swirlies all the time in the boys room and wedgies in gym going on to major in microbiology in college and still has all that unresolved anger getting into a Level 4 work area and "playing" with stuff. |
| 02:53.33 | *** join/#asterisk kensukeido (n=Kensuke@unaffiliated/kensukeido) |
| 02:58.18 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 03:00.50 | jaytee | suddenly a hush fell over the room |
| 03:01.51 | Katty | SHUUSH! |
| 03:02.05 | jaytee | and it's all because your momma don't dance and your daddy don't rock n' roll |
| 03:02.28 | *** join/#asterisk Palin (n=jcook@adsl-69-208-65-187.dsl.klmzmi.ameritech.net) |
| 03:02.38 | Katty | rickrolled. |
| 03:03.40 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 03:04.01 | jaytee | transcoding that song and putting it on * as a playback number and then writing the number somewhere people see it would be a funny new way to rickroll people. |
| 03:04.52 | jaytee | I've been thinking of doing that for 800 numbers incoming here at home |
| 03:05.02 | jaytee | instead of Zapateller |
| 03:05.43 | Palin | wondering if someone can help me? |
| 03:06.04 | Katty | sorry i'm not a licensed therapist |
| 03:06.08 | jaytee | I wonder that sometimes too |
| 03:06.17 | Palin | :) |
| 03:06.34 | jaytee | the first step is admitting you have a problem |
| 03:07.55 | Palin | I have a sip provider and i am behind a cheap router, on the problem i having is, when i call comes in it look in the public domain |
| 03:07.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:07.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:07.59 | Palin | Looking for 6165887123 in 6165887123 (domain 69.208.65.187) <<<<< should be 192.168.3.55 |
| 03:08.04 | jaytee | ~sipnat |
| 03:08.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:08.12 | *** join/#asterisk d-k-t (n=dt@60.176.195.165) |
| 03:08.30 | jaytee | hi russellb |
| 03:08.43 | Palin | Thank i will check them out |
| 03:08.44 | russellb | jaytee: hey there |
| 03:09.15 | jaytee | russellb, another late night of coding? |
| 03:09.30 | russellb | heh ... it's not too late, and i'm not really coding :) |
| 03:10.00 | jaytee | oh, is it just after 10 there? |
| 03:10.15 | russellb | yup |
| 03:10.32 | jaytee | you're in Huntsville, right? |
| 03:10.39 | russellb | yes |
| 03:10.54 | jaytee | I'm always thinking that's EDT not CDT |
| 03:11.32 | russellb | yeah, we're close to the time zone border |
| 03:11.35 | jaytee | living here in Indiana when I first moved here in 2000 was hard getting used to cuz they didn't observe Daylight Savings. |
| 03:12.12 | russellb | that would be quite irritating |
| 03:12.31 | jaytee | so in the winter it would be the same time as NYC and in the summer the same time as Chicago. |
| 03:12.46 | jaytee | more irritating to out of state businesses. |
| 03:12.56 | russellb | right |
| 03:13.05 | russellb | just irritating that your time relative to other places changes |
| 03:13.07 | jaytee | which is why they finally caved and adopted DST. |
| 03:16.43 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 03:18.23 | *** part/#asterisk Palin (n=jcook@adsl-69-208-65-187.dsl.klmzmi.ameritech.net) |
| 03:19.12 | *** join/#asterisk bbryan1 (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 03:19.17 | bbryan1 | ~book |
| 03:19.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 03:19.35 | *** part/#asterisk bbryan1 (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 03:20.55 | *** join/#asterisk Entr4nced (n=none@cpe-76-190-141-153.neo.res.rr.com) |
| 03:22.45 | [TK]D-Fender | jaytee: I'm gonna WIN!!! |
| 03:23.15 | Katty | what are you winning? an orange sodie? |
| 03:23.27 | [TK]D-Fender | Katty: Pandemic 2! |
| 03:23.37 | jaytee | [TK]D-Fender, cool!!!! |
| 03:23.41 | Katty | oh boy! |
| 03:24.00 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 03:24.01 | [TK]D-Fender | jaytee: 100% infection worldwide and I just introduced liver failure :) |
| 03:24.15 | jaytee | ooooh, how many days left till a vaccine? |
| 03:24.34 | [TK]D-Fender | jaytee: 1455, but I promise you..... they don't have that long :0 |
| 03:24.41 | jaytee | hahahahaa |
| 03:24.55 | Katty | hmm. |
| 03:24.56 | Katty | i'm hungry. |
| 03:25.45 | jaytee | what transmission vectors did you use? I usually start with airborne and add rodents and insects to cross closed borders and airports. |
| 03:27.14 | jaytee | and just out of curiosity, are the frakken airports in Greenland still open? |
| 03:28.03 | [TK]D-Fender | jaytee: NO LIFE! |
| 03:28.09 | [TK]D-Fender | jaytee: First to fall |
| 03:28.27 | [TK]D-Fender | jaytee: DAYS TO COMPLETION : Inifinty, muahahahah!!!!! |
| 03:28.41 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 03:28.47 | jaytee | [TK]D-Fender, congrats!!!! |
| 03:29.45 | tzanger | hmm pandemic 2 is kind of a slow game |
| 03:29.51 | tzanger | I'm stuck in argentina |
| 03:30.07 | jaytee | tzanger, you can adjust the speed |
| 03:30.28 | tzanger | yeah it's at max |
| 03:30.39 | tzanger | whoa I just jumped in infection |
| 03:30.42 | tzanger | 1000 -> 80000 |
| 03:31.34 | Katty | someone ship me a bagel. |
| 03:31.40 | tzanger | I just took over south america |
| 03:31.49 | [TK]D-Fender | 118K people to go! |
| 03:32.26 | jaytee | ever wonder about the poor slobs on the little boat that keeps going around looking for an open port? |
| 03:32.36 | [TK]D-Fender | 9K :p |
| 03:32.56 | tzanger | wow a half million injections a day? |
| 03:33.24 | jaytee | "I think we have ourselves a winnah!!!" |
| 03:33.49 | [TK]D-Fender | 74! |
| 03:34.08 | [TK]D-Fender | EXTERMINATION! |
| 03:34.11 | [TK]D-Fender | woohoo |
| 03:34.43 | jaytee | \o/ |
| 03:35.13 | [TK]D-Fender | "Humanity Overcome. Mankind's numbers have declined. Your disease has managed to exterminate humans off the face of the Earth." : Score : 80118 |
| 03:36.07 | Katty | well i sur-pose i'm going to go to bed. |
| 03:36.21 | jaytee | nite Katty |
| 03:38.43 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:42.33 | seanbright | ok, i killed 6.2B people |
| 03:42.49 | seanbright | cuba, south africa, and madagascar got away |
| 03:42.49 | jaytee | seanbright, well good for you!!!! |
| 03:42.51 | seanbright | :-/ |
| 03:42.52 | *** join/#asterisk jplank (n=GBove@ool-18bb0206.dyn.optonline.net) |
| 03:43.30 | jaytee | game I'm currently playing the virus started in South Africa but still hasn't managed to make it to Madagascar. |
| 03:45.26 | [TK]D-Fender | jaytee: My approach : airborne bacteria w/ low resist (NO drug to start). let that run for a while. Then added water & upped resists. NO symptoms for a long time. |
| 03:45.47 | jaytee | same strategy here |
| 03:45.49 | [TK]D-Fender | jaytee: trick is invisibility |
| 03:45.54 | jaytee | yep |
| 03:46.12 | jaytee | fly under the radar and keep the airports and shipping open |
| 03:46.48 | jaytee | then when the shit really hits the fan add rodents as a transmission vector cuz they don't get stopped at border crossings. |
| 03:47.20 | jplank | are you guys talking about pandemic 2? |
| 03:47.20 | jplank | ol |
| 03:47.38 | [TK]D-Fender | yup |
| 03:48.45 | jplank | very frustrating |
| 03:49.16 | jplank | and I don't have enough patience to complete the game on realistic |
| 03:49.34 | jplank | I usually consider infecting the whole population of the world as a win |
| 03:50.02 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 03:53.06 | jaytee | well, I'm off to sleep. nite everyone |
| 03:53.09 | [TK]D-Fender | jplank: I just did that. |
| 03:53.24 | [TK]D-Fender | jplank: [23:35]<[TK]D-Fender>"Humanity Overcome. Mankind's numbers have declined. Your disease has managed to exterminate humans off the face of the Earth." : Score : 80118 |
| 03:54.04 | jplank | How many people dead? |
| 03:56.54 | tzanger | ugh, how do you shake the vaccine? |
| 03:57.10 | jplank | cross your fingers and hope it doesn't work |
| 03:57.14 | tzanger | it did work |
| 03:57.20 | tzanger | now I'm just trying to mutuate to shake it |
| 03:57.23 | jplank | then your beat |
| 03:57.34 | tzanger | I took out greenland |
| 03:57.37 | tzanger | almost canada |
| 03:59.52 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 04:01.34 | jameswf-home | wondes if someone can ewrite this to be relivent within irc/asterisk related forums http://www.catb.org/~esr/faqs/smart-questions.html |
| 04:04.39 | TJNII | "7. If you're a programmer, try to find an answer by reading the source code." |
| 04:04.46 | TJNII | I'm going to remember that one. |
| 04:04.49 | TJNII | RTFS! |
| 04:04.56 | *** join/#asterisk kinkin (n=A@64.180.188.233) |
| 04:06.10 | jameswf-home | bad advise depending on the coder...... bad whitespace, no comments, bad habbits, I would rather post to experts-exchange |
| 04:07.04 | jameswf-home | some asterisk stuff is commented in spanish :) |
| 04:07.13 | TJNII | Heh. |
| 04:07.14 | *** join/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi) |
| 04:07.18 | wasabi | Does anybody have any references for setting up QoS on a Linux router, specifically with Asterisk/Switchvox in mind? As in, I don't want to be "confused" with the idea of bittorrents, and sub queues, and the like. I just want VoIP to be "fastest". |
| 04:07.28 | jameswf-home | ~qos |
| 04:07.29 | jbot | methinks qos is Quality of Service, a great source of information is located @ http://www.lartc.org |
| 04:08.51 | jplank | wasabi: if you search the freepbx forums, theres a script to setup iptables to tag outgoing SIP packets with diffserv if thats what you are looking for |
| 04:09.18 | jplank | or by linux router, do you mean an actual linux router? |
| 04:09.26 | jplank | like a machine dedicated to routing |
| 04:09.48 | wasabi | Yeah. |
| 04:09.56 | wasabi | Actual Linux router. With a T1/E1 card no less. |
| 04:10.09 | jameswf-home | -A OUTPUT -p udp -m udp --sport 5060 -j DSCP --set-dscp 0x2e |
| 04:10.24 | jameswf-home | -A OUTPUT -p udp -m udp --sport 5060 -j DSCP --set-dscp 0x2e |
| 04:10.32 | jameswf-home | damnit |
| 04:10.34 | wasabi | So DSCP is what tags packets? One of the very consuing things was what does the trailing number refer to? |
| 04:10.39 | wasabi | Just a value or something meaningful? |
| 04:10.59 | *** join/#asterisk ManxPower (n=manxpowe@35.sub-70-220-26.myvzw.com) |
| 04:11.14 | jplank | well if your basing your qos policy on diffserv dscp is what that packets are tagged *with* |
| 04:11.33 | wasabi | and diffserv = ? |
| 04:11.52 | jplank | 0x2e is hex for the EF value (1110100 or something like that), for the expedited forwarding queue |
| 04:11.56 | wasabi | Also, looking on the freepbx forums |
| 04:12.13 | jplank | wasabi: that question would probably be better answered by google them me |
| 04:12.14 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 04:12.31 | *** join/#asterisk hi365_m (n=hi365@213.151.63.7) |
| 04:12.33 | wasabi | Yeah. That's been my problem. A dozen different things to learn whose scopes far extend what I'm working on right now. |
| 04:12.53 | wasabi | I don't mind learnign these things... but I have a specific problem to solve first. :) |
| 04:12.54 | jplank | but basically diffserv is a way of tagging ip packets inside the TOS field so routers know what type of traffic it is, and be able to queue it |
| 04:13.11 | wasabi | So a SIP phone doesn't tag them automatically? |
| 04:13.12 | jplank | keeping realtime traffic real |
| 04:13.18 | jplank | not all of them |
| 04:13.23 | wasabi | I see. So this is a field in IP. |
| 04:13.31 | wasabi | Which can then be used by devices to assign priority. |
| 04:13.37 | jplank | but you have to make sure you are tagging it with a tag that is part of your qos policy |
| 04:13.45 | jplank | yea |
| 04:13.46 | wasabi | And like all fields in IP whether a given devices sets it or not is open to how much you paid for the device. :) |
| 04:13.54 | jplank | not really |
| 04:14.09 | wasabi | TOS = Type of Service? |
| 04:14.17 | jplank | in this case, most voip equipment made after like 2000 should support it |
| 04:14.20 | jplank | yes |
| 04:14.22 | wasabi | Ahh. Good. |
| 04:14.30 | jplank | (probably even earlier) |
| 04:14.33 | wasabi | So, first task is to property type the packets, if the types are missing. |
| 04:14.36 | wasabi | properly. |
| 04:14.40 | wasabi | And using iptables one can do that easily. |
| 04:14.46 | jplank | yea |
| 04:14.57 | wasabi | Cool... so diffserv = ? I'm guessing some sort of queueing mechanism? |
| 04:15.08 | jplank | i guess |
| 04:15.17 | jplank | its kind of the map on how to queue the packets |
| 04:15.43 | jplank | dscp = diffserv code point |
| 04:15.46 | wasabi | Ahh. I see. Diffserv is one way of doing queuing which is very rough. |
| 04:15.52 | jplank | yea |
| 04:15.52 | wasabi | Just simply vlauing certain packets higher. |
| 04:15.58 | jameswf-home | ~dscp |
| 04:16.01 | jplank | kinda |
| 04:16.05 | jameswf-home | stupid bot |
| 04:16.06 | wasabi | All this other stuff I've been stuck in is about assigning bandwidth limits to various classes of stuff like bittorrent. |
| 04:16.07 | *** join/#asterisk PepOSX (n=angeldav@190.78.221.212) |
| 04:16.13 | wasabi | Which seems very complicated (because it is) |
| 04:16.25 | jplank | well, the concept behind diffserv is easy |
| 04:16.39 | jplank | implementation could be confusing. |
| 04:16.58 | jplank | every piece of equipment needs to be queuing by it for it to be effective |
| 04:17.12 | jameswf-home | Linux... not for the windows admin or those who are pregnant or might become pregnant. If you have heart problems you may wish to consult a Mac |
| 04:17.25 | jplank | lol |
| 04:17.28 | wasabi | I guess. |
| 04:17.31 | wasabi | My router runs great though. :0 |
| 04:17.41 | wasabi | I built my own router to replace a Cisco box. Bought a DSU/CSU. |
| 04:17.57 | jameswf-home | ~cisco |
| 04:17.58 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 04:18.05 | jameswf-home | ~botsnack |
| 04:18.05 | jbot | jameswf-home: aw, gee |
| 04:18.07 | jplank | aha - but can you tell me what a csu/dsu does :) |
| 04:18.15 | wasabi | Sort of. :0 |
| 04:18.32 | jameswf-home | ~csu |
| 04:18.36 | wasabi | In my case, not much. It establishes a serial connection between two connected end points. |
| 04:18.36 | jplank | making a T1 work doesn't count as an answer |
| 04:18.48 | jameswf-home | psh going to take back the botsnack |
| 04:18.49 | wasabi | (multilink PPP) |
| 04:19.10 | wasabi | And does the various line level encoding and normalization required for the electronics to move! |
| 04:19.21 | jplank | better answer :) |
| 04:19.37 | wasabi | T1's are just connected pairs over a long distance, no? |
| 04:19.43 | jplank | yea |
| 04:19.51 | jplank | with equipment on each end |
| 04:20.03 | wasabi | Yeah. So, it speaks some selection of low level protocols. |
| 04:20.05 | jameswf-home | ~ds1 |
| 04:20.06 | jplank | they are really just two (or four) copper pairs with a csu on either side |
| 04:20.11 | jameswf-home | ~t1 |
| 04:20.12 | jbot | [~T1] T1 is the basic digital telephony circuit used in North America. T1 runs at 1.544 Mbps. It can be an unstructured channel for data. It can be channelized, to provide 24 time slots of voice or data, each of 64kbps. Time slot 24 is used for D-Chan when used with PRI signalling. |
| 04:20.30 | jameswf-home | ~E1 |
| 04:20.31 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 04:20.34 | jplank | or it could be mixed of channelized and un channelized |
| 04:20.53 | wasabi | Yeah. I have one card with two T's plugged in. |
| 04:20.58 | d-k-t | Not sure that's 100% correct |
| 04:21.02 | wasabi | Exposed as /dev/ttyW0 and /dev/ttyW1. Simple serial connection. |
| 04:21.04 | d-k-t | had an E1 in Hong Kong |
| 04:21.09 | wasabi | Overwhich pppd runs multilink. |
| 04:21.21 | jplank | you can create more then one tdm group over a ds1 |
| 04:21.30 | jplank | i do it every day |
| 04:21.32 | jameswf-home | has used an E1 in the us but not sure you can get one from a provider |
| 04:21.58 | wasabi | it is nice having a real linux box as a router though. |
| 04:21.59 | jplank | jameswf-home: the closest you can get is a T1 to E1 point to point to overseas |
| 04:22.01 | wasabi | much easier to deal with. |
| 04:22.21 | wasabi | nice to just pop open ngrep on it |
| 04:22.29 | jplank | but those aren't fun |
| 04:22.36 | d-k-t | jameswf-home, in the US I've used T1s with providers but E1s internally |
| 04:22.43 | wasabi | they're fun after dealing with a ios box for 2 years. |
| 04:22.53 | jameswf-home | I saw on southpark the internet is run by a giant linksys |
| 04:23.06 | jplank | roflmao |
| 04:23.08 | d-k-t | jameswf-home, it's true |
| 04:23.21 | jplank | unfortunately it is |
| 04:23.25 | jplank | shakes fist |
| 04:23.27 | jplank | damn cisco |
| 04:23.38 | jameswf-home | we all need to go californi way and get some internet |
| 04:23.44 | jplank | lol |
| 04:23.57 | jplank | and the worst part is 9 times out of 10, a reboot will fix your problem |
| 04:25.22 | wasabi | Never had that. |
| 04:25.31 | wasabi | Mostly the problem was everytime I used it I had forgotten how it worked. |
| 04:26.54 | [TK]D-Fender | tzanger: Think I'm gonna win again. |
| 04:27.32 | *** join/#asterisk d-k-t-2 (n=dt@vpn21.witopia.net) |
| 04:28.41 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 04:29.15 | BBHoss | jameswf-home: lol lol wtfbbq! |
| 04:42.19 | *** join/#asterisk eltech (n=eltech@ool-4576feda.dyn.optonline.net) |
| 04:43.15 | eltech | hi is it possible to reverse the order of how voicemails are played back when retrieved by phone ? |
| 04:45.15 | *** join/#asterisk hardhatpat (n=pat@c-24-21-226-34.hsd1.mn.comcast.net) |
| 04:45.42 | [TK]D-Fender | tzanger: 100% Infection @ 136 Days & 0 Deaths. Now to kill them all :D |
| 04:47.53 | hardhatpat | I have a problem where after a few days of uptime, my box just stops working. if i 'reload' it works fine. |
| 04:47.59 | hardhatpat | here is what happens in the console: http://pastebin.ca/1080364 |
| 04:49.55 | hardhatpat | by 'box' i mean, 'asterisk app' |
| 04:50.56 | [TK]D-Fender | hardhatpat: [Jul 22 21:47:02] NOTICE[5649]: chan_sip.c:13865 handle_request_invite: Call from '' to extension '5033341400' rejected because extension not found. <--- dialplan errors |
| 04:51.22 | hardhatpat | [TK]D-Fender, what kind of error will cause it to work and then stop a few days later? |
| 04:51.43 | [TK]D-Fender | hardhatpat: What you showed us tells absolutely nothing |
| 04:52.25 | hardhatpat | [TK]D-Fender, where should i look? just guess and pray while editing extensions.conf? logs? |
| 04:52.45 | [TK]D-Fender | hardhatpat: got a better description of exactly what it is that "fails" |
| 04:52.54 | hardhatpat | stops making or receiving calls |
| 04:53.00 | [TK]D-Fender | hardhatpat: "stop working" could use a whole lot of details... |
| 04:53.19 | *** join/#asterisk zver (n=zveruga@altlinux/developer/zver) |
| 04:53.24 | [TK]D-Fender | hardhatpat: well right NOW it looks like you're failing to get calls due to a dialplan problem. |
| 04:53.43 | hardhatpat | [TK]D-Fender, if i go into the console and type 'reload' its going to start working fine... |
| 04:54.23 | hardhatpat | [TK]D-Fender, im loathe to do this, because last time i had this problem and came in to ask how to fix it ... you said you cant fix a working box, so now i return with a broken one that will be fixed by issuing 'reload' |
| 04:54.24 | [TK]D-Fender | hardhatpat: Ok, well sorry, but you've basically got nothing to show us. We can't help you in that state |
| 04:56.13 | [TK]D-Fender | hardhatpat: well right now you only have 1 evident flaw. |
| 04:56.27 | [TK]D-Fender | hardhatpat: and no idea if there is a conditionin your configs that "resets" |
| 04:56.34 | zver | hello. i configure sip user and try auth with soft phone, but it doesn't work. in console see it http://pastebin.ca/1080367. Can you help me how fix it? |
| 04:56.54 | hardhatpat | [TK]D-Fender, i think i may have found the problem ... the default contexts in extensions.conf that i just left in could be causing this? can i remove them? (voip-info is unclear about this) |
| 04:58.16 | [TK]D-Fender | hardhatpat: we have no idea what you're doing in there at all. How should we know? |
| 04:58.52 | hardhatpat | [TK]D-Fender, yes i understand |
| 04:59.37 | hardhatpat | [TK]D-Fender, i found this in the messages log, thought its relevant? (asterisk said i should read it) http://pastebin.ca/1080376 |
| 05:00.33 | hardhatpat | [Jul 22 19:36:17] WARNING[5476] pbx.c: Context 'longdistance' tries includes nonexistent context 'local' ... can i just comment out context 'longdistance'? |
| 05:01.42 | TJNII | Perhaps you should read the chapter on dialplans and write your own instead of futally trying to band-aid one from christ knows where? |
| 05:01.55 | jplank | would you guys think it would be too messy to use asterisk as a multi tenant pbx? |
| 05:02.54 | jplank | (without using a distro made specifically for it) |
| 05:03.06 | [TK]D-Fender | hardhatpat: again we have NO IDEA what you've coded, how should we know? |
| 05:03.15 | hardhatpat | [TK]D-Fender, roger. |
| 05:05.29 | [TK]D-Fender | tzanger: \o/ |
| 05:07.05 | jplank | anyone ever try out thirdlane PBX manager? |
| 05:08.01 | [TK]D-Fender | jplank: plenty of multi-tennent * GUI;'s out there. |
| 05:08.19 | [TK]D-Fender | jplank: ScopServ is one I've used myself. Very large, pretty good. |
| 05:08.49 | jplank | its starting to become a pain in the ass to program everything maanually |
| 05:08.55 | jplank | i'm checking out ScopServ right now |
| 05:10.52 | jplank | [TK]D-Fender: is it easy to use? would I be able to have people not too familiar with asterisk do simple changes? |
| 05:11.09 | jplank | (after teaching them how first of course) |
| 05:11.13 | [TK]D-Fender | jplank: Yes, it entirely runs the show |
| 05:11.22 | [TK]D-Fender | jplank: Disstinctly not-free |
| 05:12.20 | jplank | are the CDR's it creates less verbose then the asterisk CDR's? |
| 05:13.37 | [TK]D-Fender | jplank: no much |
| 05:14.16 | jplank | I got to play around with the demo |
| 05:14.26 | jplank | feature wise it looks perfect |
| 05:14.39 | jplank | hell even the smb version seems like it would do the trick |
| 05:15.11 | jplank | I assume it works like freepbx does? Does it allow custom configs? |
| 05:16.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:16.37 | [TK]D-Fender | jplank: rather nominal outside its GUI. |
| 05:16.51 | [TK]D-Fender | jplank: But the GUI is pretty thorough |
| 05:17.38 | [TK]D-Fender | jplank: I was an early adopter over 2 years ago and am in intermittent regular contact with the company |
| 05:17.47 | jplank | I want to make sure I can make custom contexts |
| 05:18.21 | jplank | one of the things I use my asterisk for (the main thing actually) is to route signalling through from my SIP gateway to rewrite SIP headers |
| 05:19.32 | jplank | to add a p-assered-id |
| 05:19.41 | jplank | p-asserted-id |
| 05:20.37 | [TK]D-Fender | jplank: Yay forgery |
| 05:20.48 | jplank | well no |
| 05:20.49 | jplank | lol |
| 05:20.58 | jplank | my sip gateway doesn't support it |
| 05:20.59 | [TK]D-Fender | jplank: I'm sure you can make a custom dial strig to go out the way you need to. |
| 05:21.11 | *** join/#asterisk Levon1 (n=Levon@adsl-76-230-110-133.dsl.lsan03.sbcglobal.net) |
| 05:21.19 | jplank | yea thats basically how I do it right now |
| 05:21.21 | [TK]D-Fender | jplank: then use an intermediary * server to rewrite it at worst |
| 05:21.56 | jplank | all calls from x context that the dial string starts with 3 gets the treatment and shot back out, before it touches anything else in the asterisk |
| 05:23.31 | jplank | problem with an intermediary asterisk is it would need to handle at least 100's calls any given second, so I would have to scale it pretty large just like my main asterisk |
| 05:25.22 | jplank | today I have somewhere around 100 calls up at any given second during business hours (I reinvite the media as to not tax the server) |
| 05:26.43 | [TK]D-Fender | jplank: 100's of calls isn't a problem, its a question of how fast setup/teardown. And again at worst, shove a proxy like SER in there |
| 05:30.21 | jplank | my sip gateway works as a proxy, no issue there. I just applied for the demo license to try it out in my lab. thanks for push in the right direction, this seems to be able to do everything I need it to, without the messy programming of everything. |
| 05:35.03 | jplank | no pricing info on the website, do you know how much the itsp version is w/ 1000/25? |
| 05:36.40 | [TK]D-Fender | jplank: Go contact them |
| 05:36.58 | [TK]D-Fender | jplank: I signed on before they were so well developed or deliniated |
| 05:37.32 | iamthelostboy | what would cause this message? app_dial.c:1111 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 05:37.58 | iamthelostboy | the iax account all looks in order.. i have the same setup on my other system and it will communicate back, i just cant get to it |
| 05:38.29 | [TK]D-Fender | iamthelostboy: Says nothing. Look at the debig, look at the peer |
| 05:38.38 | [TK]D-Fender | debug* |
| 05:39.47 | *** join/#asterisk steliosk (n=Stelios@athedsl-4396818.home.otenet.gr) |
| 05:41.09 | *** join/#asterisk Keypad (n=Keypad@125-238-132-95.broadband-telecom.global-gateway.net.nz) |
| 05:48.41 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 05:52.24 | *** join/#asterisk CunningPike_ (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 06:02.25 | *** join/#asterisk baliktad (i=baliktad@c-24-16-27-4.hsd1.mn.comcast.net) |
| 06:03.35 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 06:04.00 | baliktad | I have some basic voip questions, trying to figure out if I need an asterisk box or not |
| 06:04.56 | jplank | baliktad: shoot |
| 06:05.38 | baliktad | If I have a normal internet connection and I pay for 1 DID, is it possible to hook up 2 voip phones behind the single connection, or will this require asterisk? |
| 06:06.12 | jplank | depends on how exactly you want to use it, but generally yea |
| 06:06.24 | jplank | you need some type of sip registrar |
| 06:06.49 | baliktad | that would be my online voip provider, right? |
| 06:06.59 | C4colo | do they give you an ATA? |
| 06:06.59 | jplank | you might be able to register both phones to the same DID (if your provider allows it) but when a call comes in, it would only ring on the last phone to register |
| 06:07.17 | C4colo | voip phones |
| 06:07.18 | jplank | both phones should be able to make outbound calls though |
| 06:07.19 | C4colo | nevermind |
| 06:07.36 | baliktad | ahh, I see |
| 06:07.57 | jplank | you need something to register the DID & register both phones |
| 06:08.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:08.40 | baliktad | will asterisk allow me to specify different providers for DID and outbound calling? |
| 06:08.41 | jplank | if its only two phones, you can probably get the open-wrt version of asterisk and put it on a linksys router |
| 06:08.45 | jplank | yea |
| 06:09.02 | jplank | your talking about like least cost routing? |
| 06:09.14 | jplank | or are you saying one provider for inbound, one for out |
| 06:09.23 | baliktad | one provider for in, one for out |
| 06:09.26 | jplank | yea |
| 06:09.28 | jplank | no problem |
| 06:10.11 | baliktad | I use dd-wrt on all my routers, but I've not heard of the open-wrt version of asterisk |
| 06:10.23 | baliktad | is that just a version of asterisk designed to run on a soho router? |
| 06:10.48 | jplank | yea |
| 06:10.48 | jplank | http://www.linux.com/articles/52030 |
| 06:11.00 | jplank | quick google of open-wrt asterisk brought up that |
| 06:11.09 | baliktad | I'll have a look, thanks |
| 06:11.21 | C4colo | make sure you get a compatible wrt54g |
| 06:11.23 | jplank | their should be a dd-wrt version also |
| 06:11.38 | jplank | C4colo: he said he's already using dd-wrt |
| 06:11.53 | C4colo | there are some that are not compatible, the newer ones including rev7 and rev8 and the new v2 (wrt54g2) are not |
| 06:11.57 | C4colo | oh |
| 06:12.04 | C4colo | I'm only half paying attention |
| 06:12.08 | jplank | lol |
| 06:12.41 | C4colo | as you probably noticed |
| 06:13.02 | jplank | just looked it up, as I expected, theirs a asterisk version for dd-wrt |
| 06:13.09 | baliktad | I appreciate the no-bs assistance, my boss wants a voip setup for the office and was just gonna dump money into windows + office communications server |
| 06:13.29 | C4colo | eeeew |
| 06:14.19 | C4colo | I would trust microsoft for something as important as a phone system when they start using linux for their servers |
| 06:14.20 | baliktad | I've heard of asterisk but never set it up before, just trying to learn about my options |
| 06:14.40 | jplank | if you expect a decent amount of calls, I'd get a dedicated asterisk server |
| 06:14.45 | jplank | not on a linksys |
| 06:15.19 | jplank | I don't know the full capabilities for asterisk on dd-wrt or open-wrt, but I'm sure its pretty limited |
| 06:15.46 | baliktad | it's not a lot of calls, very infrequent actually, but we need to have 2 phones for the front/back of the office |
| 06:16.06 | Keypad | lol http://www.break.com/index/it-guy-vs-dumb-employees.html |
| 06:16.16 | jplank | there's some slick things about windows office communications live or whatever its called, but I try to move as much off windows as possible personally |
| 06:16.17 | Keypad | wromg room |
| 06:16.18 | Keypad | ;S |
| 06:16.34 | jplank | Keypad: thats hilarious |
| 06:16.44 | Keypad | lol seen it ? |
| 06:16.50 | jplank | about 10 times already |
| 06:17.01 | jplank | everytime I show it to someone, I have to watch it with them |
| 06:17.11 | jplank | did you reboot it? |
| 06:17.16 | *** join/#asterisk gnorbert (n=gnorbert@228-177.static.ew.hu) |
| 06:17.18 | jplank | yea 3 times, just like you always tell me |
| 06:17.50 | jplank | sorry there's no sort by penis option |
| 06:18.13 | jplank | oh wait |
| 06:18.20 | jplank | thats not it-guy vs sales-guy? |
| 06:18.23 | jplank | theres a new one? |
| 06:18.42 | baliktad | wow, if I can squeeze a basic version of asterisk on the router and avoid buying new hardware AND software, I might pull a nice bonus out of this |
| 06:19.33 | jplank | good for you |
| 06:20.10 | jplank | if you want stuff like vm and the like, I don't know how well it would work (unless you have a fast router and a external storage device) |
| 06:20.44 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 06:20.45 | baliktad | we don't need any extra features like vm or conferencing or anything really |
| 06:21.32 | jplank | oh, then that should work perfectly |
| 06:21.33 | baliktad | it's just a satellite office, the only calls are really to the main office and the occasional customer or pizza place |
| 06:21.40 | jplank | but not as much fun :) |
| 06:22.02 | Keypad | haha jplank yeah its good |
| 06:22.44 | jplank | Keypad: fyi the real website for that video is thewebsiteisdown.com |
| 06:22.48 | Keypad | Can any one help me to get calling working on asterisks |
| 06:23.10 | jplank | thats a very general question |
| 06:23.17 | jplank | what exactly is your issue? |
| 06:28.46 | torrikft | morning all |
| 06:30.14 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:38.47 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 06:38.57 | Keypad | jplank: well I have no idea how asterisks works |
| 06:39.31 | Keypad | jplank: I have a Linksys VOIP router connected |
| 06:39.42 | Keypad | and I can check my voice mail etc |
| 06:41.53 | Keypad | but I have no idea on what to do now |
| 06:41.55 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 06:42.02 | creativx | Keypad: did you read TFOT |
| 06:42.58 | Keypad | TFOT ? |
| 06:43.32 | creativx | ~tfot |
| 06:43.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 06:43.53 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 06:44.02 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 06:44.28 | baliktad | don't worry Keypad it's only 600 pages |
| 06:44.40 | creativx | its also indexed |
| 06:44.42 | baliktad | I have been working my way through it |
| 06:44.49 | Keypad | lol !! |
| 06:44.50 | creativx | meaning you dont have to start at the beginning if you dont need |
| 06:45.03 | creativx | but if you have no clue.. and wish to gain such a clue.. start reading :-) |
| 06:45.17 | Keypad | I have a idea :) |
| 06:45.19 | Keypad | but its just that |
| 06:45.49 | Keypad | I just cant get the PSTN line working with asterisks |
| 06:45.57 | creativx | then start reading.. :) |
| 06:49.14 | Keypad | reads Connecting to the PSTN |
| 06:50.18 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 06:51.14 | creativx | anyone here ever used the asterisk.net + fastagi lib? |
| 06:52.07 | *** join/#asterisk Dovid (n=Dovid@bzq-82-81-16-221.red.bezeqint.net) |
| 06:52.34 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 06:54.02 | Keypad | No luck |
| 06:54.38 | Keypad | how am I ment to call out to the PSTN using my phone line anyway |
| 06:54.51 | *** join/#asterisk zydoon2 (n=zydoon@41.224.163.244) |
| 06:55.24 | *** part/#asterisk zydoon2 (n=zydoon@41.224.163.244) |
| 06:55.55 | creativx | well step 1 would be to get the phone line connected to the asterisk server in some manner |
| 06:56.03 | C4colo | many a good asterisk devveloper has had trouble with the fxo on sipura devices |
| 06:56.59 | Keypad | I have the phone thats connected to the linksys router talking to astrisk |
| 06:57.23 | Keypad | but I am not sure on how to call out using the phone |
| 06:57.41 | Keypad | if I unplug the phone from the ethernet it works, but thats not the point |
| 06:57.45 | creativx | what kind of phone? |
| 06:57.49 | EmleyMoor | Keypad: Do you have services available to call out on? |
| 06:57.53 | Keypad | Just some 2$ phone |
| 06:58.12 | Keypad | 1Custom - PSTNCustom Voip |
| 06:58.21 | Keypad | I set that up |
| 06:58.32 | Keypad | but I dont know how to tell if I did it the right way |
| 07:00.30 | EmleyMoor | Keypad: Is asterisk registering to that account? |
| 07:01.45 | Keypad | EmleyMoor: How do I find that out ? I am newb :) |
| 07:02.29 | EmleyMoor | asterisk -r, and check sip show peers and sip show users |
| 07:03.20 | styelz | sips his coffee |
| 07:03.36 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 07:04.32 | EmleyMoor | might finally be making the move to 1.4 at the weekend |
| 07:05.02 | *** join/#asterisk fcois93 (n=fcois93@bagnolet.acropolistelecom.net) |
| 07:05.41 | EmleyMoor | (of course, check similar for the relevant protocol if that is not a SIP account) |
| 07:06.35 | creativx | EmleyMoor: from 1.2? |
| 07:07.00 | creativx | man this fastagi is puzzling me.. why is asterisk receiving garbage data before the command.. gahh |
| 07:07.10 | EmleyMoor | creativx: Yes |
| 07:07.30 | creativx | EmleyMoor: how did you go about and review outdated functions et al? |
| 07:08.12 | EmleyMoor | creativx: Mainly by watching in the CLI and editing out any functions that were deprecated |
| 07:08.21 | creativx | ah so you did a dry run |
| 07:08.40 | EmleyMoor | (even got rid of one that is not deprecated until 1.4 but is unnecessary as the 1.6 way works already) |
| 07:09.19 | EmleyMoor | creativx: No. I have no deprecated applications or functions in use and have checked the book etc. |
| 07:09.35 | EmleyMoor | I have not actually tried my dialplan out in 1.4 |
| 07:09.48 | *** join/#asterisk tengulre11 (n=tengulre@125.71.208.16) |
| 07:10.30 | EmleyMoor | Some may begin to show as deprecated in 1.4, I will convert those to 1.6 compatible at my convenience |
| 07:11.04 | jplank | EmleyMoor: you might want to try your dialplan on a box running 1.4 first, I had a nice long night of troubleshooting when I went from 1.2 to 1.4 |
| 07:11.23 | jplank | would never wish it on anyone else |
| 07:11.36 | Keypad | EmleyMoor: I am using the web gui ? How do I go about doing tha t? |
| 07:11.56 | EmleyMoor | Keypad: Sorry, no idea on that score |
| 07:12.35 | jplank | Keypad: check out asteriskguru.com for beginner examples to asterisk |
| 07:12.57 | jplank | it has a nice walk through |
| 07:15.35 | *** join/#asterisk mrnick (n=basement@kulnet-nat-2.kulnet.kuleuven.be) |
| 07:15.54 | Keypad | jplank: thanks |
| 07:15.57 | Keypad | brb |
| 07:17.18 | EmleyMoor | Still no further along with what I was looking for on the N95 though |
| 07:20.48 | *** join/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 07:20.54 | *** join/#asterisk devhen (n=devhen@businessmentoring.fttp.xmission.com) |
| 07:21.12 | devhen | hey guys |
| 07:21.19 | *** part/#asterisk zydoon (n=zydoon@213.150.170.26) |
| 07:21.24 | devhen | ive got a problem :( |
| 07:21.45 | devhen | i just replaced a linksys WRT54G which was acting as my router with m0n0wall installed on a dual-nic PC |
| 07:22.03 | devhen | now my calls have no sound! |
| 07:22.07 | devhen | what can I do |
| 07:22.26 | devhen | there is sound on internal calls between extensions but nothing on calls going over my bandwidth.com trunks |
| 07:22.47 | EmleyMoor | devhen: SIP calls? You need to configure the firewall to allow for them and perform any translation necessary |
| 07:23.06 | devhen | EmleyMoor: thanks. I have opened port 5060 to my asterisk server, just as I had it on my linksys |
| 07:23.50 | EmleyMoor | I can't be more specific in your case - but you need to allow for the RTP as well |
| 07:26.40 | devhen | EmleyMoor: RTP? are you talking about 10000-20000 ports ? |
| 07:26.49 | devhen | bandwidth.com told me I only needed 5060 though I dont really believe that |
| 07:27.03 | devhen | before ive used UDP 10001 to 20000 |
| 07:27.14 | many | teh phil. |
| 07:29.03 | dominic1 | why do I always get a Peerstatus expired for my iaxmodems in the manager log? |
| 07:31.34 | devhen | EmleyMoor: opening 10001-20000 seemed to solve it |
| 07:32.47 | *** join/#asterisk oilinki (n=oil@ppp-124-120-7-16.revip2.asianet.co.th) |
| 07:34.49 | gnorbert | Hi, does somebody have any idea, what can be the problem? |
| 07:34.49 | gnorbert | http://pastebin.com/d3faba606 |
| 07:35.50 | gnorbert | I guess it's quite like a beginner's trouble. |
| 07:40.51 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 07:43.05 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 07:43.26 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:44.38 | gnorbert | Anybody reading me? :) |
| 07:45.36 | Keypad | man its hard to see whats going on |
| 07:45.42 | Keypad | I just got a phone called |
| 07:45.44 | Keypad | call* |
| 07:45.46 | dominic1 | can anybody tell me how I can adjust hylafax to send more than one job to a destination? |
| 07:45.53 | dominic1 | at the same time |
| 07:46.48 | Keypad | Its hard to see if it went through asterisks or it just went from the PSTN to line1 thats on the same device |
| 07:49.19 | *** join/#asterisk ToTo (n=ToTo@209.8.41.108) |
| 07:52.35 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 07:53.09 | *** join/#asterisk freddyk (n=freddy@host131-140-dynamic.42-79-r.retail.telecomitalia.it) |
| 07:53.14 | freddyk | Good morning |
| 07:53.30 | freddyk | i should ask a question about ZAPTEL codec |
| 07:53.41 | freddyk | can anyone help ? |
| 07:54.01 | *** join/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 07:54.26 | gnorbert | Yout are the fourth one, who wants to ask something, if I remember well. :D |
| 07:55.08 | Keypad | How can I tell what Asterisk is doing when I try and call something ? |
| 07:55.26 | Keypad | I want to find out if its talking to the PSTN line |
| 07:55.29 | Keypad | but I cant tell |
| 07:55.29 | cfh | hi all, i have a ISDN-BRI gateway SIP (patton) connect to a server asterisk (SIP trunk) and i have some problem with dtmf digit on isdn line. |
| 07:55.36 | Keypad | It just makes a busy ringing sound |
| 07:55.41 | freddyk | is there any way to force alaw instead of slin as default codec for zaptel ? |
| 07:55.41 | gnorbert | By now, no answer, but if I can, of course I try (not so relevant that I can. ():-)) |
| 07:56.20 | gnorbert | Did you try writing disallow=all allow=alaw? |
| 07:56.23 | cfh | what can i use to test the correct use of the dtmf signaling |
| 07:56.33 | EmleyMoor | Keypad: Watch the CLI with a verbosity of 3 |
| 07:56.44 | EmleyMoor | A "busy ringing" sound? |
| 07:57.55 | baliktad | is ip possible/recommended to set up asterisk in a virtual machine (voip only, no analog phones)? |
| 07:59.11 | freddyk | gnobert |
| 07:59.27 | freddyk | my problem is that i really don't know |
| 07:59.32 | EmleyMoor | baliktad: Probably not. I found it poor on a low-grade machine - adequate for testing but that was it. On a VM it is likely to be worse |
| 07:59.34 | freddyk | where to set zaptel codec |
| 07:59.48 | baliktad | hmm |
| 08:01.34 | gnorbert | freddyk: I guess if you install zaptel and configure zaptel.conf and zapata.conf, it should work, but I don't know, maybe I misunderstood the question too. :) |
| 08:02.03 | freddyk | i have a cisco connected to a sangoma e1 |
| 08:02.08 | freddyk | my problem is that |
| 08:02.25 | freddyk | cisco accepts a-law and slin from e1 |
| 08:02.48 | freddyk | but it should transcode to alaw when receving slin |
| 08:02.52 | freddyk | and it results in poor audio |
| 08:03.28 | freddyk | so i would just force alaw to the cisco |
| 08:03.35 | freddyk | (where i can't change the configuration) |
| 08:04.02 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 08:04.06 | gnorbert | freddyk: Then I didn't understand your problem. :) This way I don't know, sorry... |
| 08:05.02 | Keypad | EmleyMoor: How do I access the CLI ? |
| 08:05.17 | EmleyMoor | asterisk -r |
| 08:05.32 | Keypad | if I ssh into the box ? |
| 08:05.51 | Keypad | or do I type that into the CLI in the web manager ? |
| 08:06.01 | EmleyMoor | Yes - provided the user you ssh in as has permission |
| 08:06.06 | freddyk | gnobert |
| 08:06.11 | freddyk | can i pm you ? |
| 08:06.17 | EmleyMoor | I don't know anything about the web manager |
| 08:06.22 | Keypad | ah k |
| 08:06.31 | Keypad | whats the SSH pass ? |
| 08:06.38 | gnorbert | freddyk: Of course, but I'm quite beginner, I don't think, I can't help you. :) |
| 08:07.17 | EmleyMoor | Keypad: We wouldn't know your passwords |
| 08:08.41 | freddyk | does any knows how to change zaptel preferred codec ? |
| 08:10.14 | Keypad | Yeah I got it xD |
| 08:10.21 | Keypad | It dident work on my mac |
| 08:10.31 | Keypad | maybe I was failing at typing on the small keyboard |
| 08:10.53 | *** join/#asterisk Dovid (n=Dovid@tony09-118-62.inter.net.il) |
| 08:10.54 | Keypad | huh ?Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 08:11.29 | gnorbert | Does somebody know, what can be the problem? |
| 08:11.32 | gnorbert | http://pastebin.com/d3faba606 |
| 08:11.41 | Dovid | keypad: Either asterfisk is not running or not under the user name that you are trying to to connect to it as. Do you see it when you run ps ax | grep asterisk |
| 08:12.28 | Keypad | is there a paste bin ? |
| 08:12.30 | Keypad | theres 4 lines |
| 08:12.52 | Keypad | 2358 ? S 0:00 /usr/sbin/lighttpd -f /etc/asterisk-gui/lighttpd.conf |
| 08:12.52 | Keypad | <PROTECTED> |
| 08:12.52 | Keypad | <PROTECTED> |
| 08:12.52 | Keypad | <PROTECTED> |
| 08:12.55 | Keypad | Sorry |
| 08:13.09 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 08:13.10 | Dovid | ~pb |
| 08:13.10 | jbot | [~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://www.rafb.net/paste |
| 08:13.22 | Keypad | ty |
| 08:13.23 | Dovid | seems like it is running |
| 08:13.34 | Keypad | I wanna see shit running and stuff |
| 08:13.37 | Dovid | under what uid is it running ? |
| 08:13.43 | Keypad | so I can see whats happening |
| 08:13.44 | Keypad | uid ? |
| 08:13.58 | Dovid | user id |
| 08:14.04 | gnorbert | Anyobody with any idea?:) |
| 08:14.23 | Dovid | gnorbert: I dont know IAX to well |
| 08:15.09 | Keypad | I dont know, this is just a Standard Asterisk Now setup |
| 08:15.14 | Dovid | basicly the receiving server is rejectin the call because of authentication. try sending the call out via a peer and see what happens. or on the receiving box create an entry for your IP |
| 08:15.16 | gnorbert | Dovid: Anyway thanks.:) |
| 08:15.36 | Dovid | Keypad: Then you belong asking in asterisk-gui |
| 08:15.45 | Keypad | oh |
| 08:15.54 | Keypad | I thought it was all the same thing |
| 08:15.56 | Dovid | i hate asterisk now |
| 08:16.18 | Dovid | this is the gen. room so if some one here has an answer they will respond |
| 08:16.25 | EmleyMoor | Keypad: It's "based on Asterisk" - it differs from Asterisk proper |
| 08:16.38 | tzafrir_laptop | I hated it even before |
| 08:16.45 | Dovid | I would say it is asterisk with a gui on top of it |
| 08:16.45 | EmleyMoor | (should we start calling it "Asterisk Later On"? <g>) |
| 08:16.58 | dominic1 | should I run iaxmodem in daemon or non-daemon mode? |
| 08:17.00 | Keypad | They should just rename it |
| 08:17.45 | EmleyMoor | dominic1: I run it from init |
| 08:18.04 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 08:18.32 | dominic1 | do you nee to add it to /etc/inittab then? |
| 08:19.10 | Keypad | It seams like no ones home in that channel |
| 08:19.15 | Keypad | Or every one is sleeping |
| 08:20.20 | tzafrir_laptop | Anybody running Asterisk from upstart? |
| 08:20.29 | EmleyMoor | I did - I followed pretty much what it says here: http://www.the-asterisk-book.com/unstable/faxserver.html and below |
| 08:21.13 | EmleyMoor | Coupled with fax-to-email from my DID provider, I now have a fax system that is reasonably reliable and doesn't crash into phone calls |
| 08:22.27 | EmleyMoor | http://www.tinsleyviaduct.com/phil/asterfax.html tells more |
| 08:25.26 | DarKnesS_WolF | tzafrir_laptop: there? |
| 08:26.10 | tzafrir_laptop | yup |
| 08:26.28 | Keypad | uh shit |
| 08:26.40 | Keypad | no ones home in asterisk gui |
| 08:27.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e873b3c22f607456) |
| 08:27.41 | tzafrir_laptop | Keypad, so you'll have to become the resident guru there :-p |
| 08:28.02 | gnorbert | :) |
| 08:29.47 | Keypad | tzafrir_laptop: I am having enough problems using the GUI let alone the CLI |
| 08:30.03 | Keypad | :P |
| 08:30.39 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:33.24 | gnorbert | Does somebody have an idea, what can be the problem? |
| 08:33.30 | gnorbert | http://pastebin.com/d3faba606 |
| 08:33.45 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 08:37.02 | EmleyMoor | gnorbert: Is 9252 supposed to be one of your phones? If so, why is it in the from-iax context? |
| 08:37.55 | gnorbert | EmleyMoor: I use softphone (linphone), but it should between two *s. |
| 08:38.02 | EmleyMoor | Do you have a context 9252 in extensions.conf? |
| 08:38.26 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 08:38.29 | gnorbert | No, just a default context and a userA. |
| 08:38.48 | gnorbert | And a general.. |
| 08:39.11 | EmleyMoor | gnorbert: From what I can see, you are trying to use a context that you don't have |
| 08:39.26 | Keypad | Does any one have any idea on how to debug whats happening to calling out ? |
| 08:39.40 | Keypad | I dont even know to see if its talking to my PSTN line |
| 08:39.44 | *** join/#asterisk kopite (n=Dave@5ac3891c.bb.sky.com) |
| 08:40.00 | Keypad | I have set up a service provider (SIP) to my PSTN router |
| 08:40.12 | Keypad | but I think the calling rules a broken |
| 08:40.15 | Keypad | but I dont know |
| 08:40.21 | Keypad | I dont know how to test them out |
| 08:40.27 | Keypad | all I know is i can call 6050 |
| 08:40.30 | Keypad | and listen to mailbox |
| 08:40.43 | gnorbert | EmleyMoor: I changed context to default, but it still writes the same. |
| 08:41.03 | kopite | While upgrading to 1.4.21.2 this morning, I got this warning message: ast_register_application: Already have an application 'Pickup' |
| 08:41.09 | kopite | * doesn't load |
| 08:41.39 | kopite | I presume that that means that I need to remove some modules, but how do I find out which one? |
| 08:41.59 | kopite | DOes anyone out there have any info that'll help me out with this one? |
| 08:42.24 | EmleyMoor | gnorbert: Is there an extension 11111 in that context? |
| 08:42.33 | EmleyMoor | (or a pattern match for it) |
| 08:42.47 | *** join/#asterisk implicit (n=bayan@ip68-105-71-13.sd.sd.cox.net) |
| 08:43.14 | gnorbert | EmleyMoor: There is _11XXX exten. |
| 08:43.32 | EmleyMoor | _11XXX. I hope |
| 08:43.38 | EmleyMoor | Ah, hold on |
| 08:43.42 | EmleyMoor | _11XXX would work |
| 08:44.01 | gnorbert | I think so. |
| 08:44.08 | gnorbert | I should, or not? |
| 08:44.20 | gnorbert | I mean it should. |
| 08:44.50 | EmleyMoor | I am not really familiar with initiating calls that way |
| 08:45.36 | gnorbert | EmleyMoor: I do it anyway, if that suits better for you, just I have to call with * a meetme conference of itself... :) |
| 08:46.16 | Keypad | When I pushed the * button and mashes some keys it made a strange sound lol |
| 08:46.27 | Keypad | Is that the Conference button is ? |
| 08:48.20 | Keypad | is confused |
| 08:48.38 | Keypad | Nearly every thing is working besides the 2 important things |
| 08:51.09 | *** join/#asterisk h-idrisi (n=h-idrisi@212.103.170.132) |
| 08:54.36 | gnorbert | EmleyMoor: Maybe any other idea? |
| 08:56.36 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 09:01.56 | gnorbert | Then anyone other idea? :) |
| 09:03.03 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 09:04.10 | Keypad | urh /wrists |
| 09:04.30 | Keypad | hmmm |
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| 09:08.38 | *** join/#asterisk jonavogt (n=jonathan@u51-229.dsl.vianetworks.de) |
| 09:08.42 | Keypad | Is there any good tutorals for now ? |
| 09:09.03 | Keypad | I am not gonna sleep if I cant fix this |
| 09:10.27 | gnorbert | After writing 9252 instead of ip address, now it writes this: |
| 09:10.29 | gnorbert | http://pastebin.com/d689c7db5 |
| 09:14.15 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 09:18.19 | gnorbert | Any idea? |
| 09:19.18 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 09:21.21 | jonavogt | Hi |
| 09:22.46 | jonavogt | I'm trying to find the option for turning the music off on calls from external. |
| 09:24.12 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:24.28 | *** join/#asterisk mythril (n=luke@c-76-115-209-31.hsd1.or.comcast.net) |
| 09:24.51 | mythril | is there any reason a landline would be playing fur elise? |
| 09:25.14 | *** join/#asterisk ToTo (n=ToTo@209.8.41.47) |
| 09:28.25 | jonavogt | @mythril I don't think so |
| 09:32.53 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 09:34.39 | *** join/#asterisk blinky42 (n=sbrown@67.200.59.43) |
| 09:34.53 | mythril | well I left my phone off the hook for a couple of weeks and it is now playing fur elise |
| 09:35.33 | mythril | I didn't know where else to ask about it, figured asterisk people know a bit about phones |
| 09:35.57 | mythril | o well, I guess it is just an oddity |
| 09:36.10 | jonavogt | so it is just an ordanary phone? |
| 09:36.21 | mythril | jonavogt: yeah |
| 09:36.49 | jonavogt | mythril: and its play fĆĀ¼r Elise when you want to dial a number? |
| 09:37.18 | mythril | no, its been off the hook for a couple of weeks and it is playing fur elise now |
| 09:37.31 | mythril | I wish I had a way to record it |
| 09:37.38 | mythril | it sounds absurd but it is happening |
| 09:37.52 | mythril | stranger than fiction |
| 09:38.06 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 09:38.38 | *** join/#asterisk Keypad (n=Keypad@125-238-132-150.broadband-telecom.global-gateway.net.nz) |
| 09:38.41 | C4colo | easter egg in the pstn? |
| 09:38.41 | *** join/#asterisk XnOSX (n=XnOSX@142.156.15.62.static.jazztel.es) |
| 09:38.51 | mythril | C4colo: that is exactly what I thought |
| 09:39.04 | jonavogt | interesting indeed |
| 09:39.29 | mythril | I googled it, but my googlin skills are too weak for this |
| 09:39.31 | C4colo | apparently if the beep beep beep BEEP BEEP BEEP didn't get you to hang it up, they think playing that song on a loop will for sure |
| 09:39.42 | C4colo | lol |
| 09:40.13 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:40.22 | mythril | well I am going to let it play so I don't have to tell the story to my friends |
| 09:40.27 | mythril | they can hear it themselves |
| 09:41.01 | mythril | the other possibility is that someone broke into my house and got me put on hold while I was gone :P |
| 09:41.39 | jonavogt | you'll find out with the next phone bill :-) |
| 09:42.02 | *** part/#asterisk BeeBuu (n=beebuu@59.38.97.88) |
| 09:47.06 | MaliutaLap | I just got an SMC "wireless travel voice gateway", cute. can run off of usb power, does wireless (but only with WEP as a client) and acts as an ATA |
| 09:50.41 | *** join/#asterisk Falle (n=falle@diana.falle.se) |
| 09:51.10 | hi365 | how do you use the /n option in dial? exten => _x.,n,Dial(Local/${EXTEN}@${CONTEXT},300,${DIAL_OPTIONS}/n) <--- doesnt seem to work |
| 09:52.56 | hi365 | Dial(Local/${EXTEN}@${CONTEXT}/n,300,${DIAL_OPTIONS}) dosent hold the channel either |
| 09:58.00 | *** join/#asterisk Cart_ (i=Cart@x148.ip6.netikka.fi) |
| 09:58.02 | Cart_ | hello |
| 09:58.28 | Cart_ | i have a problem with asterisk. some module causes asterisk not to start, but I don't know which one |
| 09:58.36 | Cart_ | if I disable autoload, it will start |
| 09:58.52 | Cart_ | how can I find which module causes the problem? |
| 09:59.25 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 09:59.42 | jonavogt | hi365 don't you mean the 'n' option? |
| 10:00.12 | hi365 | jonavogt: yes, the n option. where does it fit in to the dial stirng? |
| 10:01.05 | jonavogt | cart: start asterisk with 'asterisk -vvvvvv' it should show you some info |
| 10:01.16 | jonavogt | <PROTECTED> |
| 10:01.37 | Cart_ | jonavogt: yes it should, but it doesn't show which module fails |
| 10:01.50 | Cart_ | last 3 lines are: |
| 10:01.58 | Cart_ | func_global.so => (Global variable dialplan functions) == Parsing '/etc/asterisk/voicemail.conf': Found == Parsing '/etc/asterisk/vm_general.inc': Found == Parsing '/etc/asterisk/vm_email.inc': Found |
| 10:02.49 | jonavogt | cart: any lines before indicate an error? |
| 10:02.54 | Cart_ | nope |
| 10:03.07 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-111-122.red.bezeqint.net) |
| 10:03.24 | Cart_ | if I disable autoload, it will start |
| 10:03.44 | Cart_ | but it doesn't work of course, because some required module is missing |
| 10:04.23 | jonavogt | cart: can you copy the output to some pastebin service? |
| 10:05.02 | *** part/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 10:05.32 | hi365 | jonavogt: missunderstanding. i dont want the n option of dial. i want the '/n' of Local channels |
| 10:05.45 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 10:06.27 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 10:08.46 | Cart_ | jonavogt: here is everything I had in scroll back buffer http://pastebin.com/m3c4e47bc |
| 10:09.32 | defsdoor | I've got a system on analog that's started getting high pitched "pip pip pip" on some of the lines |
| 10:10.42 | defsdoor | I think it's bleedover or short in the lines with a redcare alarm monitoring system but just want to double check that nothing in asterisk with sangoma remora would generate the pips ? |
| 10:11.09 | gnorbert | Hi, does somebody have an idea, what can be the problem? |
| 10:11.10 | gnorbert | http://pastebin.com/d3ffac1be |
| 10:11.13 | jonavogt | cart: sorry to say, i didn't find anything either. strange |
| 10:11.18 | Cart_ | yes it is |
| 10:11.52 | Cart_ | I have had similar problems before also, found a module causing it - it was mp3, but this time it is some other |
| 10:11.53 | jonavogt | hi365: should be Local/00531234567@pbx/n |
| 10:12.00 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 10:12.04 | hi365 | k. thanks |
| 10:12.08 | *** part/#asterisk mythril (n=luke@c-76-115-209-31.hsd1.or.comcast.net) |
| 10:12.12 | hi365 | for now ill try to do it with 'g' |
| 10:12.35 | Dr-Linux|home | what's the difference between cisco sip firmware 7.4 and 8.2 ? |
| 10:13.28 | jonavogt | cart: you can try adding the modules one by one to the manual load list... |
| 10:14.01 | *** join/#asterisk Levon1 (n=Levon@adsl-75-62-130-108.dsl.lsan03.sbcglobal.net) |
| 10:14.25 | Cart_ | well.. of course I could, but ... there is 176 modules :p |
| 10:14.56 | jonavogt | cart: i know... just as a last resort |
| 10:16.06 | Cart_ | hmpf.. I would like to try something else before that :) |
| 10:17.26 | gnorbert | Anybody? |
| 10:20.24 | *** join/#asterisk udzinari (n=david@host-88-210-253-165.adsl.caucasus.net) |
| 10:24.09 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 10:31.40 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
| 10:31.43 | PakiPenguin | hello |
| 10:31.50 | PakiPenguin | does asterisk support SIP-I or SIP-T? |
| 10:33.09 | jonavogt | gnobert: sorry no |
| 10:33.45 | gnorbert | jonavogt: Anyway thanks, then maybe I'll try it later. :D |
| 10:34.03 | *** part/#asterisk Segnale007 (n=Segnale0@host92-1-dynamic.9-79-r.retail.telecomitalia.it) |
| 10:45.34 | *** join/#asterisk LT (n=lt@unaffiliated/lt) |
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| 10:50.50 | tzafrir_laptop | gnorbert, generally instead of asking: "can anybody help me with this pastebin trace?" try to be more specific |
| 10:52.10 | gnorbert | tzafrir_laptop: I wrote the .conf files and the message of Asterisk on pastebin, I didn't want to write it in here, it's not so short. |
| 10:52.15 | tzafrir_laptop | e.g.: "can anybody help me debug an IAX problem? http://cutpaste.example.com/123456 |
| 10:52.41 | gnorbert | tzafrir_laptop: O, ok, sorry. |
| 10:52.48 | tzafrir_laptop | The thing is someone needs to bother reading it first :-) |
| 10:52.53 | gnorbert | tzafrir_laptop: I try it again. :) |
| 10:54.17 | gnorbert | I have an IAX problem, I should make * call its own meetme conference, I get the folowwings: http://pastebin.com/d3ffac1be Can somebody help? |
| 10:54.43 | gnorbert | tzafrir_laptop: I hope that looks better. ():-) |
| 10:54.43 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 11:13.20 | gnorbert | So can anybody help? :) As I wrote I have an IAX problem (with .call files), I should call with * its own meetme conference. I have these: |
| 11:13.21 | gnorbert | http://pastebin.com/d3ffac1be |
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| 11:19.18 | *** join/#asterisk oilinki (n=oil@ppp-124-120-6-223.revip2.asianet.co.th) |
| 11:19.36 | dominic1 | can anybody tell me why iaxmodem is not logging to /var/loh |
| 11:19.41 | dominic1 | /var/log |
| 11:23.20 | yang2 | Many IAX problems today |
| 11:23.52 | gnorbert | Conclusion: IAX is problematic. :) |
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| 11:31.55 | Wayhigh | sup all |
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| 11:48.13 | *** join/#asterisk ToTo (n=ToTo@209.8.41.190) |
| 11:48.21 | gnorbert | Somebody awake? :) |
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| 11:59.38 | HameedKhan | Hello everyone, I have never worked with asterisk before and now one of my friends have asked me do he need a dedicated machine for running asterisk? He already have a machine on which he runs a mail server and a webserver. |
| 12:07.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 12:13.09 | gnorbert | I try again, who knows... :) Van somebody help me? I have problem with IAX and .call files. The .conf files and the message of Asterisk is here: |
| 12:13.15 | gnorbert | http://pastebin.com/d3ffac1be |
| 12:13.20 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.55 | creativx | morning [TK]D-Fender |
| 12:15.40 | [TK]D-Fender | creativx: mornin' |
| 12:25.13 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-0ae430f2a5dc1fce) |
| 12:25.17 | dominic1 | hi, can anybody tell me why my iaxmodem. which I run from inittab is not logging to /var/log/iaxmodem/? |
| 12:27.50 | gnorbert | Can somebody help me in the following: I should call with Asterisk its own meetme conference, but I can't. I pasted the .call file, the .conf files and the message of Asterisk at http://pastebin.com/d38282270 . |
| 12:29.07 | [TK]D-Fender | gnorbert: "iax2 show peer 9252" |
| 12:30.26 | gnorbert | [TK]D-Fender: http://pastebin.com/d526fc757 |
| 12:30.59 | [TK]D-Fender | gnorbert: Do the words "Can't call a phone that hasn't registered" ring a bell? |
| 12:33.27 | gnorbert | [TK]D-Fender: I think then I should read more about how to register a phone to iax. I didn't find anything about it in the article that I read. :) |
| 12:34.43 | [TK]D-Fender | gnorbert: "Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)". It says "I should actually look at what I'm calling, and how." |
| 12:36.21 | gnorbert | Actually I was sure, that I call 11111 in default part of extensions.conf and for this I use 9252 in iax.conf |
| 12:36.41 | [TK]D-Fender | gnorbert: Yes, well that device is off-line and it will fail just the same |
| 12:37.40 | *** join/#asterisk mgdm (n=michael@serenity.mgdm.net) |
| 12:37.43 | gnorbert | Ok, so then I have to register, I go and try, thanks. :) |
| 12:37.56 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 12:49.31 | *** join/#asterisk yakimbo64 (n=yakimbo6@193.251.169.129) |
| 12:50.50 | yakimbo64 | hello, i'm trying to dp passthrough calls with g729, my softphone is in g729 >> asterisk g729 in passthrough >> destination softphone in g729 So whene i place a call, my asterisk box is doing passthrough and voice quality is great. but whene orininating a call using AMI the call fails |
| 12:52.08 | [TK]D-Fender | yakimbo64: You should be pastebin-ing the CLI output of the attempt wit SIP DEBUG. |
| 12:52.12 | [TK]D-Fender | ~pb |
| 12:52.13 | jbot | [~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://www.rafb.net/paste |
| 12:52.14 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 12:52.37 | *** join/#asterisk bsaxon (n=bsaxon@68.sub-75-203-189.myvzw.com) |
| 12:52.58 | yakimbo64 | ok |
| 12:53.09 | yakimbo64 | will do a try now |
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| 12:54.09 | *** join/#asterisk ctooley (n=ctooley@209.33.108.119) |
| 12:54.25 | ctooley | There a way to get the number of channels in the dialplan of 1.4? |
| 12:54.43 | ctooley | Something equivalent to the 1.6 or trunk function "SYSINFO(numcalls)" |
| 12:55.01 | *** part/#asterisk bsaxon (n=bsaxon@68.sub-75-203-189.myvzw.com) |
| 12:56.03 | [TK]D-Fender | ctooley: cheating : use "groupcount" on ALL of your calls. better way : external script that will return the real number |
| 12:57.31 | jplank | Is there any other way to keep the asterisk out of the way of the media without a reinvite? |
| 12:58.06 | ctooley | [TK]D-Fender, Reading up on GroupCount and having a rough time wrapping my head around anything this morning. |
| 12:58.36 | [TK]D-Fender | jplank: No. * is a B2BUA, not a proxy |
| 13:02.41 | *** join/#asterisk ManxPower (n=manxpowe@147.sub-75-251-150.myvzw.com) |
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| 13:08.59 | Dr-Linux|home | hi |
| 13:09.01 | Dr-Linux|home | question: |
| 13:09.06 | Dr-Linux|home | what's the difference between cisco sip firmware 7.4 and 8.2 ? |
| 13:09.23 | Dr-Linux|home | why should i upgrade from 7.4 to 8.2? |
| 13:09.36 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.49 | ManxPower | It sure would be nice if they had a changelog for that sort of thing. 8-| |
| 13:10.30 | Dr-Linux|home | ManxPower: so for i'm unable to find changelog ... you know any? |
| 13:10.33 | *** join/#asterisk Segnale007 (n=Segnale0@host92-1-dynamic.9-79-r.retail.telecomitalia.it) |
| 13:10.49 | Dr-Linux|home | ManxPower: however i've upgraded my cisco phones to 8.2 |
| 13:11.00 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:11.24 | ManxPower | Dr-Linux|home: Back before I saw the light and switched to Polycoms I used Cisco phones. The readme in the firmware zip file always pointed to an online changelog |
| 13:11.34 | Dr-Linux|home | but can't see any difference |
| 13:12.02 | Dr-Linux|home | ManxPower: there is no readme in firmware zip ... |
| 13:12.03 | [TK]D-Fender | Dr-Linux|home: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7906g_7911g/firmware/8_2_2/english/release/notes/11822sr4.html#wp112765 |
| 13:12.14 | [TK]D-Fender | Dr-Linux|home: http://www.google.ca/search?hl=en&q=cisco+SIP+8.2+changelog&btnG=Search&meta= |
| 13:13.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.15 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 13:17.32 | Dr-Linux|home | hhm.. looks like there nothing changed but bug fixes |
| 13:18.33 | mikeshank | hi all, i know this is a really noob question but, I have asterisk up and running, a dialplan set to make and receive calls, the only problem is when I test by calling my number the recieving phone rings but I don't hear anything on the phone placing the call |
| 13:18.54 | gnorbert | [TK]D-Fender: I have registered a user like register => userA@172.18.1.142, but how can I authenticate in the .call file? Maybe my google skills are weak, but I can't find it. |
| 13:18.56 | mikeshank | a ring that is, |
| 13:18.57 | *** join/#asterisk bmg505 (n=leon@196-209-8-9-ndn-esr-2.dynamic.isadsl.co.za) |
| 13:19.44 | [TK]D-Fender | gnorbert: What is supposed to be registered to that peer? |
| 13:20.26 | jonavogt | mikeshank: check if you disabled music on hold for the external line and answered the line |
| 13:21.16 | mikeshank | jonavogt: forgive me, just getting my feet wet with this stuff, how would i check that |
| 13:21.24 | gnorbert | [TK]D-Fender: Ok, min I look for authenticating Asterisk, thanks. :) |
| 13:22.00 | jaytee | mikeshank, do you have the r option in your Dial statement? |
| 13:22.22 | jonavogt | in the dial application make sure you leave out the m flag for now. |
| 13:22.23 | mikeshank | jaytee: no |
| 13:23.00 | [TK]D-Fender | mikeshank: Start over and actually tell use what you are USING. the idea of just saying "I call my number and don't hear ringing on my phone" is ridiculous. |
| 13:23.16 | jonavogt | mikeshank: that is for the musik on hold |
| 13:23.19 | [TK]D-Fender | mikeshank: WHAT "number"? How does the call come in? What "phone"? |
| 13:23.50 | jonavogt | mikdeshank: then check if you answered the call in your dialplan using answer() |
| 13:24.24 | jonavogt | mikeshank: but the first should do the trick |
| 13:25.43 | jonavogt | jaytee: you don't need the r option, do you? Without asterisk generates ringtones were appropiate, doesn't it? |
| 13:26.05 | mikeshank | [TK]D-Fender: ok, sorry. grandstream gxp-2000, asterisk, and broadvoice. and a very minimal dialplan http://pastebin.com/m77d5dd87 |
| 13:26.35 | dominic1 | how can send more than one fax to a destination? |
| 13:26.40 | dominic1 | with hyla |
| 13:27.00 | [TK]D-Fender | dominic1: have more than 1 modem, and visit #hylafax :) |
| 13:27.32 | [TK]D-Fender | mikeshank: pastebin a call with SIP DEBUG enabled |
| 13:27.34 | [TK]D-Fender | ~pb |
| 13:27.34 | jbot | [~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://www.rafb.net/paste |
| 13:27.35 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 13:29.18 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 13:29.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:29.25 | mikeshank | [TK]D-Fender: ok dumbass question here, do I do sip debug in cli |
| 13:29.43 | [TK]D-Fender | mikeshank: "sip debug" |
| 13:30.59 | mikeshank | just to clarify, * handles the call, i mean i can pick up and talk, just don't hear the ringtone when calling into |
| 13:31.04 | mrnick | i can connect to my asterisk server behind NAT from a softphone behind a different NAT but i can't register |
| 13:31.15 | [TK]D-Fender | mikeshank: Ok, fine, sure. PASTEBIN |
| 13:31.16 | *** join/#asterisk moy (n=moy@nat/ibm/x-c039e328375eb5ca) |
| 13:31.24 | mrnick | anybody knows a solution for this problem? |
| 13:31.33 | [TK]D-Fender | mrnick: Read the guide : |
| 13:31.35 | [TK]D-Fender | ~sipnat |
| 13:31.36 | jbot | [~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 |
| 13:31.37 | [TK]D-Fender | ^^^^^^^^^^^ |
| 13:32.36 | mrnick | have already been reading a lot, but not that one, thank you |
| 13:34.11 | ManxPower | Lack of ringing tone *can* be caused by not having a /etc/asterisk/indications.conf |
| 13:34.11 | jonavogt | is back |
| 13:34.56 | tzanger | ManxPower: yep, I have run itno that and it was maddenning to figure |
| 13:34.58 | tzanger | *out |
| 13:37.52 | mikeshank | [TK]D-Fender: http://pastebin.com/m568ad5e5 |
| 13:38.40 | [TK]D-Fender | mikeshank: Contact: <sip:s@192.168.1.7> <- your system isn't correctly configured to work behind NAT either, and that isn't a CALL. |
| 13:38.55 | [TK]D-Fender | mikeshank: Fix your SIP NAT setting, and then show me what I asked for. |
| 13:39.25 | tzanger | bah |
| 13:39.31 | tzanger | hit by asterisk's idiotic matching rules again |
| 13:39.33 | tzanger | [foo] |
| 13:39.45 | tzanger | exten => 1234,1,NoOp(context foo, exten 1234) |
| 13:39.46 | tzanger | [bar] |
| 13:39.50 | tzanger | include => foo |
| 13:40.05 | tzanger | exten _X.,1,NoOp(context bar, wildcard match ${EXTEN}) |
| 13:40.11 | tzanger | it should NOT hit the wildcard!! |
| 13:40.27 | tzanger | (if I dial 1234) |
| 13:41.00 | mrnick | the sip solution doesn't seem to work for me |
| 13:41.07 | ManxPower | tzanger: stuff local to the context always overrides included stuff |
| 13:41.31 | mrnick | the connection with my 2pstn provider and DID becomes "unreachable" |
| 13:41.42 | tzanger | ManxPower: which is dead wrong, IMO. wildcards should be dead last |
| 13:41.57 | ManxPower | tzanger: they are within the same context. |
| 13:42.01 | tzanger | i.e. a no-wildcard match in an included context should match before a wildcard match in the current context |
| 13:42.02 | *** join/#asterisk Peri (n=redanti@66.185.87.56) |
| 13:42.04 | tzanger | IMO anyway |
| 13:42.05 | ManxPower | But I agree with you. |
| 13:42.06 | tzanger | it makes the most sense |
| 13:42.14 | *** part/#asterisk Peri (n=redanti@66.185.87.56) |
| 13:42.23 | Qwell | put the wildcard into another context, and include that also |
| 13:42.24 | tzanger | the current method makes for ass-backwards dialplans |
| 13:42.28 | tzanger | but I digress |
| 13:42.29 | tzanger | ugh |
| 13:42.35 | *** join/#asterisk Peri (n=redanti@66.185.87.56) |
| 13:42.44 | gnorbert | [TK]D-Fender: I'm sure, there is something, again, that I missed, but I don't know, what, sorry. :S Now I have http://pastebin.com/d30286a4f and it still writes the same. |
| 13:42.45 | tzanger | I have to put my wildcard matches in their own context and include them last |
| 13:42.46 | mrnick | how can i make the external-behind-nat softphone reachable |
| 13:42.48 | ManxPower | tzanger: you're experienced enough to know there is no such thing as a simple dialplan |
| 13:43.04 | jaytee | tzanger, wanna see an ass-backwards dialplan? I'll show you mine if you show me yours! :-) |
| 13:43.05 | tzanger | ManxPower: true, but it doesn't mean I can't bitch about it :-) |
| 13:43.18 | tzanger | in fact I think Iw rote a patch at one point but it was rejected |
| 13:43.27 | tzanger | haha |
| 13:43.34 | ManxPower | tzanger: Just ask the devs how much I like to complain 8-) |
| 13:43.40 | tzanger | :-) |
| 13:43.52 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-128-171.dsl.teksavvy.com) |
| 13:44.00 | ManxPower | I can be pretty harsh. |
| 13:44.03 | mrnick | it says the softphone is unreachable in the sip show peers |
| 13:44.12 | jaytee | speaking of complaining....I'm so pissed that no matter what so far I've failed to infect a single person on Madagascar. |
| 13:44.16 | ManxPower | mrnick: there are at least 5 things that could cause that. |
| 13:44.18 | [TK]D-Fender | mrnick: I gave you the guide, not show us what you DID. PASTEBIN your sip.conf masking only passwords |
| 13:44.25 | ManxPower | can you ping the IP of the softphone from the Asterisk server. |
| 13:44.46 | ManxPower | mrnick: But none of us are going to help you if you don't provide the requested information |
| 13:44.56 | mrnick | okay, there we go, i'm a firsttime pastebiner |
| 13:45.03 | [TK]D-Fender | gnorbert: You are NOT getting it. 9252 is set us as an account that SOMETHING ELSE has to register to. |
| 13:45.09 | [TK]D-Fender | gnorbert: Like a PHONE. |
| 13:45.17 | [TK]D-Fender | gnorbert: What is supposed to be using that device? |
| 13:45.18 | *** part/#asterisk ctooley (n=ctooley@209.33.108.119) |
| 13:45.36 | [TK]D-Fender | jaytee: I've been lucky a few times and STARTED there :) |
| 13:45.43 | [TK]D-Fender | jaytee: 2 world-kills :) |
| 13:46.01 | jaytee | [TK]D-Fender, haha it's so addictive |
| 13:46.18 | [TK]D-Fender | gnorbert: What are you trying to do with [9252]? |
| 13:46.28 | gnorbert | [TK]D-Fender: That is used by Asterisk i believe... But the only way of authentication I found for Asterisk was in Dial application. |
| 13:46.32 | mrnick | http://pastebin.com/d607e2e2a |
| 13:46.43 | ManxPower | gnorbert: there are many auth places in Asterisk |
| 13:46.54 | [TK]D-Fender | gnorbert: and "register => userA@172.18.1.142" does NOT belong under a peer entry. I belongs AFTER everying else under [general] and before your first peer entry |
| 13:47.19 | ManxPower | mrnick: good. I assume the softphone is [1000]? |
| 13:47.26 | mrnick | exactly |
| 13:47.37 | [TK]D-Fender | mrnick: I linked you to a guide that provided details of the settings required to operate from behind NAT and you haven't done ANY OF IT. |
| 13:47.38 | ManxPower | mrnick: I'm waiting for the answer to my ping question |
| 13:47.39 | gnorbert | [TK]D-Fender: Ok, I make a try again. |
| 13:47.55 | ManxPower | mrnick: I assume your Asterisk is NOT behind NAT? |
| 13:47.58 | [TK]D-Fender | gnorbert: STOP. NOW. |
| 13:48.01 | mrnick | no, i did but it made my connections unreachable |
| 13:48.08 | mrnick | so i put it back the way it worked |
| 13:48.18 | [TK]D-Fender | mrnick: Do what we have advised and SHOW US. |
| 13:48.25 | mrnick | my asterisk is behind nat |
| 13:48.27 | gnorbert | Ok. |
| 13:48.48 | [TK]D-Fender | mrnick: How can we tell you where you went wrong when you kill the evidence? Stop wasting our time. |
| 13:49.01 | ManxPower | mrnick: then I cannot help you. [TK]D-Fender gave you what you needed. You did not addd localnet/externip, nor did you forward your ports. I can't waste my time on you anymore. |
| 13:49.10 | mrnick | the guides leave the possibility that it doenst work the stun procedure and i want to exclude that before continuing to search |
| 13:49.11 | [TK]D-Fender | gnorbert: What the &^%$# is [9252] ? |
| 13:49.14 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 13:49.30 | ManxPower | mrnick: you seldom need STUN |
| 13:49.35 | [TK]D-Fender | mrnYou are running in circles and not showing anything of value. You aren't going to get far like this. |
| 13:49.35 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:49.36 | gnorbert | The name of an iax user? |
| 13:49.50 | ManxPower | [TK]D-Fender: welcome to Wed |
| 13:49.53 | [TK]D-Fender | gnorbert: WHAT IAX user? What exactly is supposed to use this entry? |
| 13:49.54 | mrnick | i'll put it back in there but everything stopped working... |
| 13:50.13 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:50.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:50.41 | ManxPower | gnorbert: We expect to know what you have setup. Without that we really can't help you. |
| 13:50.56 | ManxPower | [TK]D-Fender: the best punishment for people that don't listen to you is to stop talking to them. |
| 13:51.42 | gnorbert | ManxPower: I pastebined iax.conf, isn't that all? I mean so it's not everything. |
| 13:51.43 | [TK]D-Fender | ManxPower: becoming very tempting. |
| 13:52.08 | [TK]D-Fender | gnorbert: tell me exactly WHAT is supposed to connect to [9252]. |
| 13:52.13 | ManxPower | [TK]D-Fender: "there is no tempt. There is only do and do not. |
| 13:52.28 | gnorbert | The Asterisk server? |
| 13:52.44 | [TK]D-Fender | gnorbert: Asterisk isn't supposed to be CONNECTING TO ITSELF |
| 13:52.48 | ManxPower | gnorbert: We don't know. that's why we are asking YOU. |
| 13:53.09 | ManxPower | gnorbert: if you don't know what it's for why did you put it in there? |
| 13:53.21 | ManxPower | [TK]D-Fender: what is gnorbert's actual issue? |
| 13:53.24 | [TK]D-Fender | ManxPower: Apparently not. clue[-1] |
| 13:53.34 | [TK]D-Fender | ManxPower: NO IDEA |
| 13:53.45 | ManxPower | [TK]D-Fender: Ah! I thought it was just me. |
| 13:53.56 | ManxPower | gnorbert: what is your ACTUAL problem? |
| 13:53.58 | [TK]D-Fender | ManxPower: He's trying to call an IAX peer entry he can't even identify |
| 13:54.19 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
| 13:54.24 | ManxPower | [TK]D-Fender: you just told me you have no idea. 8-| |
| 13:54.29 | [TK]D-Fender | ManxPower: He points to an account but can't say what is supposed to be registered to it, and then makes it look like he's trying to have * CALL ITSELF. |
| 13:54.34 | gnorbert | ManxPower: I looked for .call files with google, I found this way. I should connect with Asterisk to one of it's meetme conferences. |
| 13:54.50 | [TK]D-Fender | ManxPower: What he's trying and what he's WANTING are 2 unseeable things |
| 13:55.08 | ManxPower | Now, gnorbert I will only ask this last time. What SPECIFICITY is not working? I don't care what you tried at this point. I want to know what is not working for you. |
| 13:55.09 | [TK]D-Fender | gnorbert: Just stop alreayd. We can't tell what the hell you are trying to DO. |
| 13:55.11 | mrnick | http://pastebin.com/d567fa57d that's the adapted version |
| 13:55.47 | ManxPower | mrnick: and what ports did you forward on your router? |
| 13:55.51 | [TK]D-Fender | mrnick: "register" must come AFTER everything else in [general] or it will fail. |
| 13:55.59 | [TK]D-Fender | ManxPower: leave this one to me... |
| 13:56.16 | ManxPower | [TK]D-Fender: I just want this guy to actually tell me something I asked for. |
| 13:56.17 | mrnick | a lot ... 3000-20000 udp & tcp |
| 13:56.25 | *** join/#asterisk dikdust (n=dik@151.36.207.204) |
| 13:56.28 | dikdust | hi |
| 13:56.35 | [TK]D-Fender | ManxPower: Guess you're welcome to it, but its a clear config issue |
| 13:56.48 | mrnick | thank you |
| 13:56.57 | [TK]D-Fender | mrnick: and [voipcheap] should be "nat=no" as well |
| 13:57.27 | *** join/#asterisk fish-bulb (n=cstewart@216.207.245.1) |
| 13:58.01 | ManxPower | mrnick: nat=yes/nat=no specifies the REMOTE end is behind NAT. It has nothing to do with Asterisk being behind NAT. The localnet=/externip=/port forwarding is what makes Asterisk work behind NAT. |
| 13:58.31 | mrnick | thank you i'm editing the sip.conf file... |
| 14:00.48 | gnorbert | ManxPower: I try to make Asterisk connect to its meetme conference. That's, what I can't do. I don't know why, that's, why I try to ask it. The message of Asterisk is at pastebin, I can write the actual address again, if you want, but I don't think, that's the problem. I guess then I should solve it on my own, just I don't really understand this part as long as I didn't find too much about it with google (sure my fault, google finds everything |
| 14:00.48 | gnorbert | , that is searched well). Sorry, if I don't understand something relevant. |
| 14:00.59 | mrnick | the softphone is working but voipcheap still unreachable |
| 14:03.01 | [TK]D-Fender | gnorbert: Askerisk should not be using IAX to call ITSELF. What is it supposed to do in this conference? |
| 14:03.28 | [TK]D-Fender | mrnick: And APSTEBIN is your friend. SHOW US |
| 14:03.31 | [TK]D-Fender | PASTEBIN* |
| 14:03.33 | gnorbert | [TK]D-Fender: It should play a sound file. |
| 14:03.47 | [TK]D-Fender | gnorbert: First, DON'T USE IAX for this. Or SIP. |
| 14:04.04 | [TK]D-Fender | gnorbert: This entire approach has been a complete waste of time. |
| 14:04.07 | *** join/#asterisk geek_cl (n=geek@190.54.42.62) |
| 14:04.57 | [TK]D-Fender | ManxPower: Finnally... only to find that it was ALL a complete waste |
| 14:05.40 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:05.40 | mrnick | http://pastebin.com/m17e76384 |
| 14:06.30 | mrnick | i can hear my ivr but once it tries to connect me to a pstn asterisk says something i can't understand |
| 14:06.41 | *** join/#asterisk |dennis| (n=Dennis@200.32.217.34) |
| 14:07.06 | [TK]D-Fender | mrnick: Please pastebin the complete CLI output with sip debug for the busted call |
| 14:07.55 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:09.49 | mikeshank | [TK]D-Fender: I'm a bit confused on setting the SIP NAT, read http://www.aocomputing.net/?p=3 but not getting it. Is externip my routers subnet? |
| 14:10.08 | [TK]D-Fender | mikeshank: no, it is the IP you GET from your ITSP |
| 14:10.22 | [TK]D-Fender | ISP rather |
| 14:11.02 | mikeshank | ok, localnet is the routers ip? |
| 14:11.05 | mrnick | http://pastebin.com/m18e905b0 |
| 14:11.10 | mrnick | this is a part of it |
| 14:11.13 | gnorbert | [TK]D-Fender: I guess it's not such a hard question and I even don't know this, but then what, if neither sip and nor aix? |
| 14:11.16 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 14:11.22 | sehh | hey people |
| 14:11.48 | [TK]D-Fender | gnorbert: Go look at the complete list of * channel types. |
| 14:12.06 | mrnick | i'm doing it remotely, he doesn't want to copy the complete debug stuff |
| 14:12.12 | mrnick | can you do something with it |
| 14:12.19 | mrnick | ? |
| 14:12.27 | sehh | q: i'd like to write a small PHP tool which connects to asterisk and runs a few commands that i need and report the results. Is there a HOWTO or some guide which can help me learn how to talk to asterisk from PHP? |
| 14:12.40 | creativx | sehh: look up AMI |
| 14:12.54 | [TK]D-Fender | mrnick: Sorry, but that isn't a call attempt. |
| 14:12.57 | sehh | creativx, AMI.. i'll google for it, thanks man |
| 14:15.22 | mrnick | that's going to be a problem... |
| 14:15.31 | blinky42 | Are there any known issues with the Page() application on 1.6.0-beta9? - I did look but didn't see anything in the bugtracker. I can place the call and phones auto-answer (polycom) but no audio seems to get passed to the other phones. Doesn't matter how many phones I Page(), same result. http://pastebin.com/d61989b6f |
| 14:17.07 | [TK]D-Fender | mrnick: And why is that? |
| 14:18.24 | mrnick | http://pastebin.com/m181bdd56 |
| 14:18.34 | mrnick | problem solved |
| 14:19.26 | mrnick | i hope it's a call attempt this time, i tried |
| 14:19.45 | drfreeze | Hello |
| 14:19.57 | [TK]D-Fender | mrnick: No, it isn't |
| 14:19.59 | drfreeze | Anyone know that magic incantation to get *8 pickup feature to work |
| 14:20.22 | [TK]D-Fender | mrnick: Go check your IP again, and its looking like 1000 is not NAT'd relative to *. It isn't, is it? |
| 14:20.30 | drfreeze | I remember it being a bit tricky when setting up a system a year ago |
| 14:22.19 | mrnick | my ip is right, just checked it |
| 14:24.09 | [TK]D-Fender | mrnick: And 1000? |
| 14:24.16 | [TK]D-Fender | mrnick: Where is it located? |
| 14:25.09 | mrnick | i presume it is nat'ed i'm on a gigantic network with my not "important" pc |
| 14:25.15 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:26.16 | [TK]D-Fender | mrnick: So it is remote as well? like : 1000 -> Router(nat) -> internet -> Router(nat) -> Asterisk ? |
| 14:27.15 | mrnick | i get my ip from a dhcp that's what it says in the configuration |
| 14:27.34 | mrnick | pretty sure it is |
| 14:28.14 | mikeshank | [TK]D-Fender: I followed the instructions for setting up sip nat and here is the call, I think http://pastebin.com/m4908b50c |
| 14:28.48 | [TK]D-Fender | mikeshank: Contact: <sip:s@192.168.1.7> <--- nope, still bad |
| 14:29.20 | [TK]D-Fender | mikeshank: and thats a REGISTER, not a CALL (which would say INVITE) |
| 14:29.31 | [TK]D-Fender | mikeshank: But either way still shows its not right |
| 14:29.49 | [TK]D-Fender | mikeshank: restart * |
| 14:29.49 | mikeshank | the nat you mean? |
| 14:29.54 | [TK]D-Fender | mikesYes |
| 14:29.57 | mikeshank | yeah i did |
| 14:30.13 | mrnick | [Jul 23 16:29:21] WARNING[14491]: app_dial.c:1210 dial_exec_full: Unable to crea |
| 14:30.13 | mrnick | te channel of type 'SIP' (cause 3 - No route to destination) |
| 14:30.36 | mrnick | that's what appears in my terminal |
| 14:31.26 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 14:31.27 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:31.33 | [TK]D-Fender | mrnick: pastebin the entire think. thats useless |
| 14:31.51 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 14:31.52 | [TK]D-Fender | mrnick: with SIP debug. |
| 14:34.28 | mrnick | http://pastebin.com/m2d9b620a |
| 14:34.34 | mrnick | i hope i got it all now |
| 14:35.59 | mikeshank | [TK]D-Fender: here's my sip and extensions confs, any reason the nat would not set correctly? http://pastebin.com/m559d9901 |
| 14:37.00 | [TK]D-Fender | mrnick: All you have are "options" packets in there. NO CALL |
| 14:37.19 | *** part/#asterisk korihor (n=korihor@190.39.163.45) |
| 14:37.52 | mrnick | i put the debug function on, called from [1000] to * until it stalled, and then copy the debug output |
| 14:38.00 | mrnick | how come it's not there? |
| 14:38.28 | [TK]D-Fender | mikeshank: mikeshank>[TK]D-Fender: I'm a bit confused on setting the SIP NAT, read http://www.aocomputing.net/?p=3 but not getting it. Is externip my routers subnet? <--- You have done A SINGLE THING that guide told you to do. |
| 14:38.44 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:38.49 | [TK]D-Fender | mrnick: What do you have forwarded to * again? |
| 14:39.36 | [TK]D-Fender | haven't* |
| 14:39.52 | dominic1 | hello, I have troubles with hylafax and nobody is in #hylafax. I know here are some hfaxgeeks. Hope you can help me. I am only able to send and receive one fax at a time. I have 4 active modems and they are used variantly |
| 14:39.57 | mrnick | forwarded? nothing, i'm calling to my * which after a festival introduction dails out |
| 14:40.22 | [TK]D-Fender | mrnick: lets try this again. WHAT PORTS have you forwarded from your router to *? |
| 14:40.45 | [TK]D-Fender | dominic1: Google <- |
| 14:41.12 | ManxPower | This is not #Hylafax 2nd leven support. |
| 14:41.37 | [TK]D-Fender | dominic1: http://www.cs.bgu.ac.il/~piavka/hylafax-howto/tweaking.html#ss6.5 |
| 14:41.41 | mrnick | ports 3000 to 20000 are open for udp&tcp on my router |
| 14:42.11 | [TK]D-Fender | mrnick: What model is it? |
| 14:42.21 | mrnick | the router? |
| 14:42.23 | [TK]D-Fender | YES |
| 14:42.38 | ManxPower | you don't need to forward TCP. |
| 14:42.47 | ManxPower | Asterisk does not use TCP for calls |
| 14:43.03 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:43.03 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:43.30 | mrnick | philips adsl wifi something |
| 14:44.38 | mrnick | don't know the specific model number |
| 14:45.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:45.52 | mikeshank | [TK]D-Fender: I'm looking it over again, what exactly did i miss |
| 14:45.57 | [TK]D-Fender | mrnick: Ok, restart the remote phone. |
| 14:46.02 | [TK]D-Fender | mikes EVERYTHING. |
| 14:46.20 | [TK]D-Fender | mrnick: And pastebin its register attempt. |
| 14:46.23 | *** join/#asterisk dmz (n=dmz@216.16.220.180.dyn-cm-pool48.pool.hargray.net) |
| 14:46.29 | [TK]D-Fender | mrnick: then try a call immediately. |
| 14:46.52 | [TK]D-Fender | mikeshank: You haven't done ANYTHING that guide told you to. |
| 14:47.41 | mikeshank | [TK]D-Fender: totally confused, i added the nat, externip, localnet, canreinvite settings, what am i missing |
| 14:47.59 | *** join/#asterisk codestr0m (n=codestr0@unaffiliated/codestr0m) |
| 14:48.17 | [TK]D-Fender | mikeshank: ..... you shoved that SIP CONFIG stuff into your DIALPLAN |
| 14:48.28 | mrnick | http://pastebin.com/d7231bf02 |
| 14:48.53 | [TK]D-Fender | mikeshank: You're supposed to be CONFIGURING SIP. |
| 14:48.58 | codestr0m | I'm having a blank right now, but what's the name of the company/site which gives out free 206 numbers? you have to have it forward to your gw |
| 14:49.00 | ManxPower | [TK]D-Fender: do you really think you'll ever manage to solve this guy's problem? |
| 14:49.28 | ManxPower | Point him to ATFOT and be DONE with him. |
| 14:49.33 | *** join/#asterisk vhatz (n=vhatz@194.219.121.194) |
| 14:49.38 | vhatz | hello all |
| 14:49.46 | [TK]D-Fender | ManxPower: mikeshank's, yes |
| 14:49.47 | mikeshank | ManxPower: you guys are tough:) |
| 14:50.26 | [TK]D-Fender | codestr0m: www.ipkall.com |
| 14:50.27 | vhatz | could someone please tell me how to use the character '#' for incoming and outgoing SIP calls? for incoming calls the # is replaced by % as I see |
| 14:50.47 | [TK]D-Fender | vhatz: put "pedantic=yes" under [general] in sip.conf |
| 14:50.51 | codestr0m | [TK]D-Fender: thanks. that's the one |
| 14:51.29 | [TK]D-Fender | mrnick: You have NOT restarted the remote phone in there. There is no register attempt. |
| 14:51.38 | vhatz | [TK]D-Fender: thank you very much, I try it!!! |
| 14:52.37 | ManxPower | mikeshank: just tired of helping people for free that don't help themselves |
| 14:53.26 | mikeshank | ManxPower: I'm trying, sorry |
| 14:53.30 | mrnick | what!? i did, but because there is constant text scrolling i put the phone and immediately started the debug option afterwards |
| 14:53.46 | mrnick | or else there is no way you'll find the register attempt |
| 14:53.46 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:54.30 | vhatz | [TK]D-Fender: I try it & did a sip reload but the sip debug output shows the same replacement of # by %... :( |
| 14:54.35 | ManxPower | vhatz: what version of Asterisk? |
| 14:54.50 | ManxPower | The % stuff is the standard URL encoded # |
| 14:56.17 | *** join/#asterisk kensukeido (i=c829e4f4@gateway/web/ajax/mibbit.com/x-789961301aa9d791) |
| 14:56.23 | codestr0m | [TK]D-Fender: ok. stupid question, but ipkall used to work. all my other inbound DID work, but I don't get inbound audio from only that number.. the server receiving the call isn't behind nat of course, but the client endpoint is.. (I'll double check.. maybe I'm missing something) |
| 14:56.27 | [TK]D-Fender | vhatz: please provide SIP DEBUG for your failed attempt along with your dialplan & sip.conf [general] section. |
| 14:56.51 | [TK]D-Fender | codestr0m: No question in there. |
| 14:57.13 | vhatz | ManxPower: it's 1.4.21.1... |
| 14:57.43 | mrnick | <[TK]D-Fender> i got to go, thanks a lot for your help i'll continue search and let the channel now if i found a solution |
| 14:57.47 | codestr0m | [TK]D-Fender: I need to debug further to get a question, but seems strange it doesn't *just work* |
| 14:58.21 | [TK]D-Fender | codestr0m: Well, we're not psychic.... lets us know when you're ready to be helped |
| 14:59.02 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 14:59.33 | *** join/#asterisk thansen|laptop (n=thansen@160.7.243.19) |
| 15:01.38 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 15:01.40 | b11d` | hello |
| 15:02.13 | b11d` | how exactly can I make Call Waiting work on a polycom 501? I can hear the "beep" when a call comes in, but cant seem to answer it without hanging up on the person im already talking ot. |
| 15:02.21 | b11d` | putting them on hold doesnt seem to work either |
| 15:02.47 | dominic1 | [TK]D-Fender: Checked already the modemgroups, but didn't help. I just want to be able to receive more than one fax from one destination.# |
| 15:03.28 | [TK]D-Fender | dominic1: RECEIVE more than one fax at a time? Same deal. Set up multiple IAXmodems and the rest is your dialplan's job |
| 15:04.02 | [TK]D-Fender | b11d`: Is it coming in on a separate line-key, or under the same as the current call? |
| 15:04.23 | b11d` | same as the current call |
| 15:04.26 | dominic1 | hm.... will I have to check with chanisavail? I simply wrote the modems in a dial statement for each. |
| 15:05.28 | [TK]D-Fender | b11d`: use the cursor arrows, and press [answer] |
| 15:05.44 | b11d` | hrm.. nothing appears on the screen though, all i hear is a beep on the line.. |
| 15:05.56 | [TK]D-Fender | dominic1: IAXmode doe not support multiple calls. you should be able to call them back to back. |
| 15:06.05 | [TK]D-Fender | b11d`: just do it |
| 15:06.08 | b11d` | ok |
| 15:06.14 | b11d` | i will give it a shot.. thanks TK! |
| 15:07.19 | vhatz | [TK]D-Fender & ManxPower: thank you guys, I think I got it! |
| 15:07.34 | mikeshank | [TK]D-Fender: ok, think the nat set now, http://pastebin.com/m307464a2 |
| 15:08.02 | [TK]D-Fender | mikeshank: that part looks better at least. |
| 15:08.15 | Superbartt | Hmmfg, I seem to be having a strange problem with a linksys ATA with a fax connected to it |
| 15:08.20 | Superbartt | It always says it's busy :x |
| 15:08.53 | mikeshank | now when i try to call i get a not available message |
| 15:10.22 | [TK]D-Fender | mikeshank: Your register tells you what BV should be dialing when they call you and that does not exist in your dialplan anywhere, let alone where one would think it should. |
| 15:10.30 | [TK]D-Fender | mikeshank: clue : CONTACT <-------- |
| 15:10.47 | dominic1 | I get errors in the cli for my iaxmodem: No route to destination) |
| 15:10.50 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 15:10.54 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:10.59 | dominic1 | I have no ideia why |
| 15:11.07 | dominic1 | sometimes it works sometimes not |
| 15:11.08 | [TK]D-Fender | dominic1: FFS pastebin and stop wasting our time. |
| 15:11.13 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:11.43 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-168-152-132.socal.res.rr.com) |
| 15:12.11 | *** join/#asterisk HonestWorker (n=Wothanaz@201.87.225.101) |
| 15:12.26 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 15:12.29 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 15:12.33 | HonestWorker | Good afternoon, gentlemen. |
| 15:12.33 | vhatz | bye all! |
| 15:13.38 | mikeshank | [TK]D-Fender: sorry, what's BV |
| 15:14.02 | [TK]D-Fender | mikeshank: .... BroadVoice. |
| 15:15.12 | HonestWorker | We have a t1 link with 30 spanned channels. We also have a few direct dialing numbers. I would like to know how to associate a channel with a particular telephone number so that whenever the person from that extension makes outgoing calls his telephone number is displayed to the receiver(which could be a PSTN endpoint). |
| 15:15.39 | HonestWorker | Does anybody has a tip on that, please? Thank you |
| 15:15.54 | EmleyMoor | HonestWorker: You mean to set the caller ID to a particular DDI based on the phone used? |
| 15:16.17 | HonestWorker | Well, I dont know whether the caller ID is the term. |
| 15:16.44 | HonestWorker | Basically the receiver should see the number that placed the call as an specific number |
| 15:16.51 | EmleyMoor | HonestWorker: You want the recipient's caller display to show the caller's DDI? |
| 15:17.04 | dominic1 | TK: Here ist my pastebin: http://pastebin.org/56373 |
| 15:17.04 | HonestWorker | currently everybody is calling the DIAL application by passing the channel group parameter |
| 15:17.49 | HonestWorker | EmleyMoor, yes, I do. I am not sure how the DDI number is sent to a PSTN telephone. I don't suppose its the caller ID attribute because that attribute could be a string. |
| 15:17.55 | EmleyMoor | HonestWorker: Try setting callerid in the relevant channel definition in zapata.conf (its name may be changing but that's what it used to be called) |
| 15:17.58 | HonestWorker | any string...including letters |
| 15:18.08 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 15:18.11 | EmleyMoor | Include the number between <> marks |
| 15:18.58 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 15:19.02 | [TK]D-Fender | dominic1: Sure doesn't look like they are registered, not does it? |
| 15:19.06 | HonestWorker | so, basically on zapata.conf(or is it zaptel? I am always exchanging those two files. anyway, the one that asterisk consults) I'd remove that channel |
| 15:19.17 | *** join/#asterisk [Akemi] (n=akemi@206-248-128-171.dsl.teksavvy.com) |
| 15:19.19 | HonestWorker | from the catch-all channel group I have currently |
| 15:19.35 | HonestWorker | and add it to a separate channel with its own characteristics |
| 15:19.43 | EmleyMoor | HonestWorker: Define each port individually - you can change just what's needed each time and keep the old characteristics |
| 15:20.24 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 15:20.25 | EmleyMoor | (not sure if what you are connected to makes a difference - I have analog FXS ports for which I do it) |
| 15:20.27 | HonestWorker | then I'd have a GotoIf application on my main dial-plan that would check against the SIP callerid, right? If sip callerid = xyz, Dial(hischannel/extension) |
| 15:20.57 | EmleyMoor | HonestWorker: What would you be trying to achieve with that? |
| 15:20.59 | dominic1 | I found out the following: The first time I dial the modem, everything is okay, if I do that a few times (call, hangup, call, hangup), after the third dial I get the error |
| 15:21.14 | mikeshank | [TK]D-Fender: so in my incoming context of my dialplan i should replace my phone number with the contact ip? sorry I know this is driving you nuts |
| 15:21.21 | dominic1 | is the registertimeout of 60 seconds okay? |
| 15:21.28 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:21.38 | HonestWorker | I'd be trying to associate a specific channel with an extension/callerid and on the zapata.conf that channel would be configured with that person's DDI |
| 15:21.44 | dominic1 | can it be a realtimerproblem? |
| 15:21.51 | [TK]D-Fender | mikeshank: register => user:pass@host/extentodialoritsgonnahitS |
| 15:22.04 | [TK]D-Fender | dominic1: Go prove that its bloody well REGISTERED |
| 15:22.09 | EmleyMoor | HonestWorker: So how does the caller ID on a SIP call make a difference? |
| 15:22.32 | dominic1 | it's registered, I have entries for ip and port in database |
| 15:22.41 | HonestWorker | well, I don't want to write a specific dial-plan excerpt for that single extension |
| 15:22.55 | EmleyMoor | HonestWorker: You wouldn't have to |
| 15:23.16 | HonestWorker | I believe I would because I need to make that person dial-out with the same channel everytime |
| 15:23.25 | HonestWorker | the other people can still use the channel group |
| 15:23.30 | thansen|laptop | is there a detailed how-to for automon?...I can't seem to get it to work |
| 15:23.31 | EmleyMoor | Just put them all in the relevant context. Caller IDs set in zapata.conf, sip.conf etc. would apply unless your dialplan changes them |
| 15:23.45 | EmleyMoor | What do you mean by "the channel group"? |
| 15:24.05 | HonestWorker | my Dial application calls a channel group that has all the 30 spanned channels included |
| 15:24.16 | HonestWorker | I will edit the group removing one of the channels |
| 15:24.18 | EmleyMoor | If you mean some don't have their own numbers, just set that group of ports last |
| 15:24.30 | EmleyMoor | (with an appropriate caller ID set) |
| 15:24.39 | HonestWorker | yes |
| 15:24.45 | EmleyMoor | Is this instance for receiving calls? |
| 15:24.48 | HonestWorker | that is exactly what I was trying to say |
| 15:24.51 | HonestWorker | no |
| 15:24.55 | HonestWorker | it is for dialing-out |
| 15:25.29 | EmleyMoor | ... and you want the caller's direct number to be sent, if any, otherwise the group number? |
| 15:25.37 | HonestWorker | so, basically on the zapata.conf I'd configure all the 29 data channels and the delta channel with their attributes first and them I'd configure the 30 channel with its own attributes |
| 15:25.42 | HonestWorker | including the DDI as its callerid |
| 15:25.48 | HonestWorker | yes |
| 15:25.54 | HonestWorker | precisely |
| 15:26.01 | EmleyMoor | Yes, that would do it. |
| 15:26.14 | HonestWorker | so |
| 15:26.27 | HonestWorker | I have just learnt that the callerid configured on the zapata.conf configuration file |
| 15:26.38 | HonestWorker | defines what is sent to the PSTN telephones receiving the calls |
| 15:26.42 | HonestWorker | is that correct? |
| 15:26.57 | EmleyMoor | Provided you are permitted to set it and send it, yes |
| 15:27.12 | [TK]D-Fender | HonestWorker: No. |
| 15:27.27 | HonestWorker | what do you mean by saying 'permitted to set it and send it' ? I do have permission to edit the configuration files. |
| 15:27.34 | EmleyMoor | Unless I have misunderstood what you were suggesting |
| 15:27.40 | [TK]D-Fender | HonestWorker: You set the CID to send in your DIALPLAN, or its taken for the originating channel |
| 15:27.42 | HonestWorker | [TK]D-Fender, no? So EmleyMoor proposed approach wouldnt work? |
| 15:28.17 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 15:28.26 | EmleyMoor | The one set for the originating channel is sent unless it is changed in the dialplan - as long as your service provider lets you set it, that is. |
| 15:28.39 | HonestWorker | Is the callerID extensions.conf variable equivalent to the number displayed on a PSTN receiver? |
| 15:29.19 | EmleyMoor | HonestWorker: Yes. If not set in extensions.conf, it takes whatever the originating channel has set as its default. |
| 15:29.26 | HonestWorker | It does allow me to set, for at the moment every channel share the same 'callerid' |
| 15:29.47 | EmleyMoor | HonestWorker: That proves nothing. It could be fixed at that callerid |
| 15:30.00 | HonestWorker | That is wierd. Suppose I edit zapata.conf and type in letters instead of numbers |
| 15:30.23 | HonestWorker | that would not be a valid value for many devices that would be receiving or switching that phone call |
| 15:30.37 | EmleyMoor | Invalid values are usually denied by the provider |
| 15:30.47 | HonestWorker | Oh, I see |
| 15:30.53 | HonestWorker | interesting. |
| 15:30.54 | EmleyMoor | Name <Number> format is supported |
| 15:31.43 | HonestWorker | I will carry on with that approach and provide you feedback. thank you for you rhelp EmleyMoor . |
| 15:31.52 | EmleyMoor | With that kind of connection I believe you are usually allowed to set it to any number which is part of your deal... but YMMV |
| 15:32.26 | HonestWorker | I have one more question. Lets say I have sent a call to a queue. Is the DIALSTATUS variable returned/acessible to the next priority of the same context? |
| 15:33.08 | HonestWorker | I would like to generate an event whenever costumers wait on the queue until the timeout(because there were no attendants available). |
| 15:33.27 | putnopvut | HonestWorker: there is a QUEUESTATUS variable for that purpose. |
| 15:33.32 | HonestWorker | I was wondering if I could check the DIALSTATUS after returning from the queue to the enxt priority |
| 15:33.39 | putnopvut | HonestWorker: in the case of a timeout, it will be set to TIMEOUT |
| 15:33.48 | HonestWorker | I see |
| 15:33.53 | HonestWorker | But timeout is generic |
| 15:33.55 | jaytee | costumers? are they wearing clown suits? |
| 15:34.09 | dominic1 | Hi TK I am a idiot |
| 15:34.15 | HonestWorker | hahaha |
| 15:34.21 | dominic1 | I set permit to 127.0.0.1 |
| 15:34.24 | HonestWorker | jaytee, hahaha sorry, bad english |
| 15:34.36 | jaytee | well, at least you're honest! |
| 15:34.37 | HonestWorker | customers |
| 15:34.47 | dominic1 | that caused the problems with iaxmodem |
| 15:34.54 | jaytee | I was just being a pita |
| 15:35.00 | HonestWorker | jaytee, sure, why not?One most to acknowledge his limitations |
| 15:35.00 | mikeshank | [TK]D-Fender: so i would add another register along with the broadvoice one like, register => s@68.37.250.134 |
| 15:35.01 | dominic1 | will I have to set permit to the default ip of the system? |
| 15:35.17 | putnopvut | HonestWorker: I don't see what you mean when you say "timeout is generic." |
| 15:35.26 | [TK]D-Fender | mikeshank: no, I just gave yout he format to use. It should be screamingly obvious what part you left off. |
| 15:35.29 | HonestWorker | Did you guys get the e-mail report from asterisk's security team? |
| 15:35.58 | HonestWorker | putnopvut, nevermind, I think the timeout will do. |
| 15:36.09 | EmleyMoor | Is there a special name for making heavy use of festival within asterisk? |
| 15:36.18 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:36.24 | HonestWorker | festival? |
| 15:36.26 | HonestWorker | what is that? |
| 15:36.38 | EmleyMoor | festival is a text-to-speech program |
| 15:36.47 | HonestWorker | oh, I see. |
| 15:36.54 | C4colo | we call that a very festive asterisk box |
| 15:36.54 | [TK]D-Fender | EmleyMoor: .. STUPID? ;) |
| 15:37.13 | C4colo | you need to put crape paper streamers on the box if you do that |
| 15:37.18 | C4colo | and bows and ribbons |
| 15:37.22 | HonestWorker | ha ha ha ha ha |
| 15:37.22 | EmleyMoor | I actually use it to avoid using my own voice for certain announcements |
| 15:37.28 | C4colo | ack |
| 15:37.29 | HonestWorker | I see |
| 15:37.42 | C4colo | and you wonder why half your customers hang up |
| 15:37.48 | HonestWorker | so basically it is the hability to have your pbx read its playbacks from a text file |
| 15:37.51 | HonestWorker | interesting |
| 15:38.03 | HonestWorker | that is good for dynamic-content messages |
| 15:38.22 | C4colo | "who is the mentally deficient voice talent they got to record their prompts?" |
| 15:38.28 | EmleyMoor | I don't have customers - in fact, most of the people that hit a festival message first off are nuisance callers anyway |
| 15:38.38 | HonestWorker | over here, we paid a professional company to record our playbacks. its rather expensive |
| 15:38.55 | C4colo | you can get Allison Smith to record them for a fee |
| 15:38.59 | EmleyMoor | I use Jay Benham's recordings for the standard asterisk prompts |
| 15:39.03 | C4colo | then all your recordings and prompts are consistant |
| 15:39.11 | HonestWorker | I see |
| 15:39.11 | C4colo | ah |
| 15:39.14 | HonestWorker | that is interesting |
| 15:39.23 | EmleyMoor | C4colo: But Allison Smith sounds like she's from the wrong side of the pond |
| 15:39.24 | C4colo | http://theivrvoice.com I beleive is her site |
| 15:39.28 | C4colo | lol |
| 15:39.34 | HonestWorker | I could prepare an external script that would update a text files with weather forecast or currency exchanges |
| 15:39.40 | C4colo | yea, but it would be consistantly the wrong side of the pond |
| 15:39.44 | C4colo | and who's side is the wrong one again? |
| 15:39.47 | HonestWorker | and people could call that extension to listen to that update info |
| 15:39.54 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:39.57 | HonestWorker | man...I didnt know about this festival application. very useful. |
| 15:40.02 | C4colo | flite |
| 15:40.17 | EmleyMoor | Jay Benham comes from Essex I believe - so he sounds quite good and not at all offputting |
| 15:40.19 | HonestWorker | 'flite' ? |
| 15:40.20 | C4colo | that is faster and less processor intensive |
| 15:40.29 | C4colo | flite is festival light |
| 15:40.52 | thansen|laptop | can someone give me some pointers on automon? I've enabled it in features.conf but can't get it create resulting sound files |
| 15:42.02 | C4colo | you want someone to sound like they are reaaaly from the wrong side of the pond, my wife does very good regionalized US Southern accent sound prompts |
| 15:42.24 | EmleyMoor | I would like a UK Northern accent |
| 15:42.24 | *** join/#asterisk angryuser (n=sldf@78.115.236.230) |
| 15:42.33 | EmleyMoor | ... preferably male |
| 15:42.33 | C4colo | she doesn't do that one |
| 15:42.37 | C4colo | or that |
| 15:42.59 | C4colo | and my british accent would make you cry |
| 15:43.10 | *** join/#asterisk NovceGuru (n=NovceGur@rrcs-70-62-198-142.central.biz.rr.com) |
| 15:43.11 | C4colo | so I don't think I can help you |
| 15:43.15 | C4colo | lol |
| 15:43.37 | EmleyMoor | UK Southern male is OK (Jay Benham) |
| 15:44.59 | HonestWorker | I would like to understand the historical reasons or rationale behind the fact in zapata.conf you define the attributes first and then the channel directive that associates the attributes with the channel(s). its much less intuitive and more difficult to read |
| 15:45.21 | HonestWorker | in my opinion. |
| 15:45.42 | [TK]D-Fender | HonestWorker: "Because" |
| 15:46.10 | C4colo | I honestly don't know why they are that way, in fact I don't really understand zapata.conf at all ... I just poke it with a stick until it works |
| 15:46.16 | C4colo | I think most people do |
| 15:46.23 | HonestWorker | C4colo, hahahahaha |
| 15:46.37 | EmleyMoor | C4colo: My original card supplier was trying to tell me I was doing it wrong... |
| 15:46.46 | HonestWorker | starts poking his zapata.conf file with a stick |
| 15:46.55 | EmleyMoor | In the end we proved the fxo was faulty and they replaced it. |
| 15:47.02 | C4colo | heh, yea |
| 15:47.08 | n3hxs | needs to be a sharp stick! |
| 15:47.10 | C4colo | it is never their fault, you are doing it wrong |
| 15:47.17 | C4colo | viatalk is like that |
| 15:47.29 | C4colo | they can't figure out their DTMF settings across all of their servers |
| 15:47.41 | wasabi | Hey, so... I'm actually using switchvox, which is using asterisk... but maybe my question is just a plain phone question. I have *55 set up to send directly to voicemail. But it's not working. Dialing it while on a call 'does nothing' |
| 15:47.42 | C4colo | half the time dtmf just doesn't work, and tech support blames your server config |
| 15:48.51 | HonestWorker | wasabi, is the dialplan logic you mentioned on the switchvox(I am assuming that is an ata or voip device) or on the asterisk? |
| 15:50.16 | putnopvut | wasabi: Have you tried asking your question in #switchvox? It may be easier to get an answer there. If you've already tried and haven't gotten an answer, then of course, nevermind. |
| 15:50.19 | HonestWorker | so, wrapping out, any characteristics/attributes I specify on zapata.conf are inherited by the channel directive I specify after the attributes. meaning, the next attributes will be associated with the next channel directive I type. right? |
| 15:50.30 | [TK]D-Fender | wasabi: WRONG CHANNEL. |
| 15:50.39 | [TK]D-Fender | wasabi: GUI's are not supported here |
| 15:50.40 | wasabi | heheh |
| 15:50.53 | wasabi | Well, is this something switchvox does or something that phone does? |
| 15:51.06 | wasabi | I was thinking, since they're all just polycoms, maybe the dtmf stuff is just 'wrong' |
| 15:51.13 | EmleyMoor | HonestWorker: Yes, and all further ones if you don't type lines for those attributes again. |
| 15:51.21 | [TK]D-Fender | wasabi: Your phone could be wrong and not sending it in the first place, or it could be your switchvox configs |
| 15:51.26 | mikeshank | [TK]D-Fender: should be obvious but it's not |
| 15:51.45 | [TK]D-Fender | mikeshank: register => user:pass@host/extentodialoritsgonnahitS <- what part of this sample format doesn't jump out at you? |
| 15:51.52 | *** join/#asterisk padski (n=padski@93-152-9-9.vibecoms.managedbroadband.co.uk) |
| 15:52.10 | mikeshank | the /exten |
| 15:52.15 | HonestWorker | EmleyMoor, so, I could inherit those attributes and just change the callerid one for that individual channel, right? |
| 15:52.22 | [TK]D-Fender | mikeshank: seems to now... |
| 15:52.23 | EmleyMoor | Yes |
| 15:52.35 | mikeshank | yeah, but i dont know what to put there |
| 15:52.38 | wasabi | Curiously how is this implemented? |
| 15:52.54 | wasabi | rfcwhatever is some standard for how the digits are sent in the middle of the sip stream or do they go over some other connection? |
| 15:53.07 | padski | I'm trying to run asterisk multi-homed, but I'm having trouble getting packets routed out the right interfaces. |
| 15:53.13 | [TK]D-Fender | mikewhatever you put there is the EXTEN that BV is going to DIAL when they call you. thats what you MATCH for in your dialplan to be able to receive the call |
| 15:53.13 | mikeshank | i tried every conceivable value, i think |
| 15:53.22 | C4colo | mikeshank: when a call comes in on that trunk it will be sent to /whateverishereattheend |
| 15:53.50 | C4colo | so just put whatever you want on the end like /1234 will call 1234 |
| 15:53.57 | [TK]D-Fender | wasabi: its in the INVITE. Either your phone's local dialplan is wrong, or your switchvox configs. |
| 15:54.47 | wasabi | isn't the invite something at the beginning of the call? |
| 15:54.52 | C4colo | wasabi: you said while on a call... are you trying to transfer the active call to voicemail? |
| 15:54.58 | wasabi | or is that a sustained open socket? |
| 15:55.00 | [TK]D-Fender | wasabi: the invite IS THE CALL |
| 15:55.02 | wasabi | C4colo: yeah. |
| 15:55.04 | C4colo | you need to hit the transfer button and then *55 or whatever |
| 15:55.10 | wasabi | yeah, i actually just figured that out. :0 |
| 15:55.19 | wasabi | thanks. =) |
| 15:55.35 | C4colo | there are very few feature codes that can be triggered while on a call |
| 15:55.40 | mikeshank | so I put /my_number which would route to my incoming context |
| 15:55.46 | C4colo | yea |
| 15:55.57 | C4colo | if you just haev one DID comeing in that sip channel just put that number on there |
| 15:56.12 | [TK]D-Fender | mikeshank: Close. It ALREADY points toa context. you need to tell it what number to DIAL in there now. |
| 15:56.29 | C4colo | yea, that |
| 15:56.38 | C4colo | speaking of which, I need to sleep |
| 15:57.15 | mikeshank | tell it what number to dial where |
| 15:57.35 | mikeshank | sorry man, i know this is tough |
| 15:57.57 | C4colo | if the trunk has context=foo and the register string has /1234 then under the [foo] context you need to create extension 1234 |
| 15:58.16 | C4colo | the trunk is defined in sip.conf and the context and extension are in extensions.conf |
| 15:58.25 | [TK]D-Fender | mikeshank: http://pastebin.com/m559d9901 <- BV should be auth-ing to [sip.broadvoice.com] which you pointed to "context=incoming". |
| 15:58.43 | [TK]D-Fender | mikeshank: its going to dial the # you told them IN THAT BLOODY CONTEXT |
| 15:59.08 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:59.32 | C4colo | ok night all |
| 15:59.41 | C4colo | wake me when I'm not tired anymore |
| 16:00.42 | *** part/#asterisk kensukeido (i=c829e4f4@gateway/web/ajax/mibbit.com/x-789961301aa9d791) |
| 16:01.18 | wasabi | okay, how about this one: when I hit transfer, and hit blind, and type *5, it exits the transfer menu. |
| 16:01.22 | wasabi | Not letting me typ the next 5. |
| 16:01.33 | wasabi | if I do not hit blind, it looks like it goes... though it hickups slightly. |
| 16:02.04 | [TK]D-Fender | wasabi: Move along... you're in the wrong channel |
| 16:02.12 | wasabi | oh come on, i'm just asking about polycoms |
| 16:02.16 | *** join/#asterisk XnOSX (n=XnOSX@212.145.173.80) |
| 16:02.27 | wasabi | next time i'll forget to mention what pbx i'm using. :) |
| 16:02.28 | [TK]D-Fender | wasabi: Go read the admin guide and look up "digitmap" |
| 16:03.11 | jonsmith1982 | is there an example anywhere that explains how to configure gtalk for incomming/outgoing calls on asterisk? |
| 16:03.34 | XnOSX | hello friends; anybody know a good software for convert audio files with a good quality for asterisk? |
| 16:03.55 | jonsmith1982 | found a few but theyre not explaining things enough. |
| 16:03.55 | [TK]D-Fender | XnOSX: sox |
| 16:04.12 | XnOSX | [TK]D-Fender, ok! |
| 16:04.15 | XnOSX | thanx |
| 16:04.19 | [TK]D-Fender | jonsmith1982: gtalk.conf.sample |
| 16:04.56 | mikeshank | [TK]D-Fender: confused, because i do have a number in the incoming context |
| 16:05.14 | [TK]D-Fender | mikeshank: .. and you didn't tell BV to USE IT |
| 16:05.52 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:05.56 | *** join/#asterisk Levon1 (n=Levon@adsl-76-227-119-202.dsl.lsan03.sbcglobal.net) |
| 16:06.09 | mikeshank | and i do that by using that number as the exten in register |
| 16:06.12 | [TK]D-Fender | mikeshank: register => my_number@sip.broadvoice.com:my_pass:my_number@sip.broadvoice.com/PUTYOURFINGNUMBERHERE |
| 16:06.38 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 16:07.16 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:07.24 | mikeshank | [TK]D-Fender: right, like I've said I have done that and still not working |
| 16:07.42 | [TK]D-Fender | mikeshank: And like always you're NOT SHOWING US |
| 16:08.05 | mikeshank | there isnt anything to show it never gets to asterisk |
| 16:08.30 | jeev | hi fender |
| 16:08.50 | mikeshank | when i call says party not avail |
| 16:09.00 | [TK]D-Fender | mikeshank: Thats a BS BV message |
| 16:09.08 | mikeshank | right |
| 16:09.19 | [TK]D-Fender | mikeshank: Go do "iptables --list" at linux CLI and PB it |
| 16:09.58 | angryuser | try of the day TK Team King ? |
| 16:10.25 | [TK]D-Fender | angryuser: "no comment" |
| 16:10.48 | angryuser | is is easyer if you tell me |
| 16:11.16 | mikeshank | [TK]D-Fender: http://pastebin.com/m56316c58 |
| 16:13.15 | *** join/#asterisk harryjr (n=harry@67-207-147-205.slicehost.net) |
| 16:13.53 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:13.54 | harryjr | hello. if the caller hangs up DIALSTATUS will not be set. will it be stated anywhere else that the caller hung up? or shoould i just assume that is what happened if DIALSTATUS is emtpy? |
| 16:14.13 | [TK]D-Fender | mikeshank: restart *. Do "sip show peers", "sip show registry", include your sip.conf, and try a sample call. |
| 16:14.14 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 16:14.35 | ZenBSDi | Anyone got experience setting up asterisk with a T1 circuit? |
| 16:14.35 | *** join/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 16:15.10 | [TK]D-Fender | ZenBSDi: plenty of us. Better questions get better answers |
| 16:16.16 | mikeshank | i see im getting a sip_reg_timeout after adding the /exten |
| 16:16.49 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 16:17.00 | padski | Hi! I'm trying to run asterisk multi-homed, but I'm having trouble getting packets routed out the right interfaces. Seems like every app I test will set the source address on reply packets but my asterisk does not ? |
| 16:17.07 | [TK]D-Fender | mikeshank: FFS stop talking about it and PASTEBIN. |
| 16:17.26 | ZenBSDi | Ok, wildcard TE122, I havea T1 circuit. should that circuit be set for PRI or leave it as data? |
| 16:17.26 | mikeshank | [TK]D-Fender: haha, ok |
| 16:17.46 | [TK]D-Fender | ZenBSDi: PRI. |
| 16:18.02 | ZenBSDi | in asterisk, pri show spans should show me the channels? |
| 16:18.23 | [TK]D-Fender | ZenBSDi: In North America you should be ordering a T1 PRI, National2 signalling, ESF, B8ZS encoding & framing. |
| 16:18.36 | [TK]D-Fender | ZenBSDi: if you configured it right and it matches the telco, sure |
| 16:18.55 | ZenBSDi | This is why you're the best TK :) thanks a bunch! |
| 16:19.08 | [TK]D-Fender | padski: it can't trace back multi-homed, because its all UDP. STATELESS. |
| 16:19.25 | [TK]D-Fender | padski: Can't reall conn-track that. |
| 16:19.58 | [TK]D-Fender | padski: You can separate "internal" networks on one interface, but the rest mask out under another at most. |
| 16:21.58 | padski | [TK]D-Fender, ip_conntrack_sip would appear to address that, but in any case I can't see a way to get netfilter to mark the packets early enough to route them. what I'm looking for is a way to tell asterisk to remember the address that they are talking to a peer from (and use the address the peer registration came in to). |
| 16:22.54 | [TK]D-Fender | padski: * isn't great at this sort of thing... if you really need it, its going to be trouble at best. |
| 16:23.34 | padski | what is Defaddr ? |
| 16:24.36 | padski | [TK]D-Fender, even if it doesn't already do it I would have thought it would be relatively straight-forward to put in. |
| 16:25.29 | mikeshank | [TK]D-Fender: http://pastebin.com/m756e7339 |
| 16:25.35 | *** part/#asterisk harryjr (n=harry@67-207-147-205.slicehost.net) |
| 16:26.00 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 16:28.10 | [TK]D-Fender | mikeshank: I asked you for a pile of things. I'm not repeating myself anymore. Provide everything you're requested up front and inclusive. |
| 16:28.24 | *** join/#asterisk Rico29 (n=Rico@nor75-29-88-188-75-101.fbx.proxad.net) |
| 16:28.30 | [TK]D-Fender | padski: Try asking that in #asterisk-dev :) |
| 16:29.10 | [TK]D-Fender | padski: I mentioned a formatting change to handle multiple NAT's that I though was simple, and to be sure it SHOULD be alot simpler than your request, and it wasn't pretty :) |
| 16:30.26 | padski | [TK]D-Fender, heh. thanks, I'll try that :-) |
| 16:31.40 | [TK]D-Fender | arikel |
| 16:31.54 | [TK]D-Fender | oops, wrong chan :) |
| 16:41.26 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 16:44.40 | *** join/#asterisk ManxPower (n=manxpowe@147.sub-75-251-150.myvzw.com) |
| 16:44.44 | *** join/#asterisk cabbiepete (n=cabbiepe@83-244-133-169.cust-83.exponential-e.net) |
| 16:49.03 | mikeshank | [TK]D-Fender: here's everything except the sample call since I can't because of the reg_timeout i suppose http://pastebin.com/m36493b32 |
| 16:50.35 | [TK]D-Fender | mikeshank: your "register" has to be AFTER everything else in [general] |
| 16:51.00 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 16:51.46 | hi365 | is there anyway to redefine the key(s) sequence used by 'H' in dial? |
| 16:52.25 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 16:54.44 | hi365 | can you use d after answer as well (as during ringing)? |
| 16:54.56 | mikeshank | ok, made register last, did restart now, sip reload, still getting the timeout |
| 16:58.20 | [TK]D-Fender | hi365: All involve serious recoding. |
| 16:58.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:58.51 | [TK]D-Fender | mikeAnother comment we shouldn't have seen without comprehensive SIP DEBUG and your updated configs |
| 16:59.05 | hi365 | [TK]D-Fender: would the d option work? can you give me a hint of whats involved? |
| 16:59.19 | [TK]D-Fender | hi365: SOURCE REPROGRAMMING |
| 17:00.04 | ManxPower | [TK]D-Fender: maybe he's just not cut out to work in the tech industry? |
| 17:00.10 | hi365 | never mind :) how about the d option? does that intercept keys after the channel is answered? |
| 17:00.38 | ManxPower | hi365: do the docs say it does? |
| 17:00.52 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 17:00.56 | *** join/#asterisk InfiL00p (n=bowserj@S0106000c418bd9f7.vc.shawcable.net) |
| 17:01.08 | *** join/#asterisk Zhad (n=tom@cpc4-sout6-0-0-cust165.sotn.cable.ntl.com) |
| 17:01.22 | [TK]D-Fender | hi365: d - Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current context, or the context defined in the EXITCONTEXT variable, if it exists. <- RTFM |
| 17:01.38 | hi365 | thanks :) |
| 17:01.46 | [TK]D-Fender | hi365: "while waiting for a call to be answered" = NO. |
| 17:02.02 | ManxPower | hi365: But you SAW THAT ALREADY, right? |
| 17:02.14 | *** join/#asterisk dni (n=john@8.10.2.83) |
| 17:02.31 | hi365 | was reading the doc's on the voip-info wiki which arent always 100% accurate/up to date |
| 17:02.31 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 17:02.49 | ManxPower | hi365: why are you using the worst source on the planet for application docs. |
| 17:02.55 | Zhad | Can someone please check that I'm not being stupid here? |
| 17:02.59 | ManxPower | Use the built in asterisk docs. "core show application dia: |
| 17:03.04 | dni | hello room,. has anyone flashed cisco 7961 to use the sip firmware and have issue getting the phone registered ?? i recieve 401 unauthorized errors,.. but while tcpdump i notice the correct user and pass being passed in the config ... anyone have some feedback ?? |
| 17:03.04 | hi365 | cause google does! |
| 17:03.17 | ManxPower | Zhad: Since you did not start out with a tech question, I would say "tes" |
| 17:03.18 | Zhad | http://pastebin.com/m1c05ce22 suggests that the remote server is not sending a callerid (number or name) right? |
| 17:03.19 | ManxPower | yes |
| 17:04.20 | hi365 | so, no way to end a call other than '*' |
| 17:05.00 | ManxPower | Zhad: that is correct |
| 17:05.06 | Zhad | Thanks |
| 17:05.08 | ManxPower | hi365: correct |
| 17:05.15 | *** part/#asterisk Levon1 (n=Levon@adsl-76-227-119-202.dsl.lsan03.sbcglobal.net) |
| 17:05.17 | hi365 | oh well |
| 17:05.27 | [TK]D-Fender | hi365: "core show application dial" <- NO EXCUSES |
| 17:05.30 | Zhad | the support people at the company who is forwarding the calls thinks it's a problem my end and wants ssh access. |
| 17:09.05 | hi365 | thinks that the d option should be set-able in features.conf akin to TtW and w |
| 17:09.07 | ManxPower | Zhad: if you don't want to do that, sending them the iax2 debug might help also a tcpdump output |
| 17:09.25 | ManxPower | hi365: then go modify the source code, sign a disclaimer, and submit the change |
| 17:09.47 | hi365 | i dont knwo jack about c. can i submit a request to the bug tracker? |
| 17:10.05 | [TK]D-Fender | hi365: That isn't a "request forum" |
| 17:10.20 | hi365 | ok. so how do i go about requesting then? |
| 17:11.19 | [TK]D-Fender | hi365: Feel free to ask in #asterisk-dev , and as a bounty on the wiki |
| 17:11.19 | *** part/#asterisk padski (n=padski@93-152-9-9.vibecoms.managedbroadband.co.uk) |
| 17:11.19 | hi365 | thanks |
| 17:11.19 | twisted | omg... [TK]D-Fender: why not point him at features.conf? |
| 17:11.46 | twisted | in the feature map section you can set the hangup key |
| 17:11.48 | twisted | (or keys) |
| 17:12.02 | [TK]D-Fender | twisted: Careful of context and think about which questions I'm answering :) |
| 17:12.25 | twisted | i did |
| 17:12.49 | [TK]D-Fender | twisted: Read a little closer, you seem to have missed it. |
| 17:12.52 | twisted | <hi365> is there anyway to redefine the key(s) sequence used by 'H' in dial? |
| 17:12.58 | twisted | <hi365> is there anyway to redefine the key(s) sequence used by 'H' in dial? |
| 17:13.13 | hi365 | twisted: are you saying there is a was? |
| 17:13.17 | Zhad | ManxPower: I already dent them the debug, it's from that they decided it was my fault (even though it's only 2 of their servers that is doing it). |
| 17:13.37 | twisted | sorry, didn't mean to paste twice |
| 17:13.44 | [TK]D-Fender | twisted: and following that he asked : [12:54]<hi365>can you use d after answer as well (as during ringing)? |
| 17:14.06 | [TK]D-Fender | twisted: You going to tell me that features.conf is going to let him allow DTMF for any little thing during a call as an exit? |
| 17:14.11 | twisted | [TK]D-Fender: yes. |
| 17:14.30 | twisted | see applicationmap seciton. |
| 17:14.46 | ManxPower | I still don't think it will do what he wants |
| 17:15.08 | twisted | it will, if it's enabled. |
| 17:15.29 | [TK]D-Fender | twisted: fugly.... but maybe |
| 17:15.38 | twisted | how is that fugly? |
| 17:15.42 | hi365 | twisted: how would end the call without totaly hanging up the channel? |
| 17:15.43 | ManxPower | I wonder why he was unable to make it work. In any case, his question was answered. |
| 17:15.47 | twisted | that's the kind of thing it was designed for |
| 17:15.58 | ManxPower | he asked about d and H |
| 17:16.11 | twisted | true |
| 17:16.27 | twisted | for the 'd' option, no. that's the point where you look at the applicationmap section of features.conf |
| 17:17.01 | twisted | for the 'H' option, yes, look at the featuremap section of features.conf |
| 17:17.51 | twisted | sorry for trying to provide hi365 an alternative method other than posting a bounty for code change when it's not required. |
| 17:17.56 | twisted | i'll go back to idling now. |
| 17:18.25 | mikeshank | [TK]D-Fender: listnen, im tryin to follow you directions as close as possible. with sip debug running i get out put every second or so it make it hard to give you want you want |
| 17:19.00 | hi365 | twisted: are you refering to the "disconnect " option? |
| 17:19.56 | dni | anyone here use cisco ip phones 7961 with flashed sip firmware with aasterisk ??? if so anyone get authentication issues ?? |
| 17:21.30 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 17:21.54 | rwaite | is there a way to disable to "Old" voicemail folder? so that messages stay in the INBOX until you delete them? |
| 17:25.42 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 17:26.02 | MaliutaLap | dna: I have a 7941, same thing under the hood |
| 17:26.12 | MaliutaLap | dna: you have the sip firmware? |
| 17:26.25 | MaliutaLap | dni: sorry |
| 17:28.11 | *** part/#asterisk mintee (n=mintone@75.150.132.150) |
| 17:28.15 | mikeshank | [TK]D-Fender: here it is again with sip debug http://pastebin.com/m6524ec99 |
| 17:30.08 | [TK]D-Fender | mikeshank: .......... |
| 17:30.24 | [TK]D-Fender | rwaite: Not without recoding the app |
| 17:31.30 | [TK]D-Fender | twisted: Quick Q : H - Allow the calling party to hang up by hitting the '*' DTMF digit. <- doesn't say it uses features.conf and implies a fixed value. Is this inaccurate? |
| 17:31.51 | mikeshank | [TK]D-Fender: ? |
| 17:32.16 | [TK]D-Fender | mikeshank: Another comment we shouldn't have seen without comprehensive SIP DEBUG and your updated configs <--------- CONFIG |
| 17:33.58 | mikeshank | [TK]D-Fender: my sip conf http://pastebin.com/m2dd79895 |
| 17:34.14 | MaliutaLap | mikeshank: can you tell us what your sip.conf looks like? and what the dialplan looks like? |
| 17:35.38 | MaliutaLap | mikeshank: I'm also assuming you have looked at http://www.broadvoice.com/support_install_asterisk.html |
| 17:35.58 | [TK]D-Fender | mikeshank: [sip.broadvoice.com] should be "nat=no". |
| 17:39.44 | mikeshank | [TK]D-Fender: ok, ive fixed that, sip reload, still timing out |
| 17:40.05 | [TK]D-Fender | mikeshank: PASTEBIN. |
| 17:41.18 | [TK]D-Fender | mikeshank: Actually save it up for someone else to help out with. I can't waste any more time chasing around trying to help you with this. |
| 17:41.32 | mikeshank | MaliutaLap: yes, had it working but started with this nat bullshit and been going in circles |
| 17:41.59 | mikeshank | [TK]D-Fender: thanks anyway for your help |
| 17:43.26 | *** join/#asterisk Levon2 (n=Levon@adsl-76-227-119-202.dsl.lsan03.sbcglobal.net) |
| 17:46.37 | *** join/#asterisk Rico29 (n=Rico@nor75-29-88-188-75-101.fbx.proxad.net) |
| 17:46.49 | *** join/#asterisk [Akemi] (n=akemi@206-248-158-200.dsl.teksavvy.com) |
| 17:51.39 | rwaite | how hard do you suppose it would be to make an options nooldfolder for the voicemail system and get it included in the main code for *? |
| 17:52.57 | *** join/#asterisk unaffiliate (n=md5@unaffiliated/unafilliate) |
| 17:53.01 | *** join/#asterisk implicit (n=bayan@ip68-105-71-13.sd.sd.cox.net) |
| 17:53.10 | Qwell | rwaite: 1.6 has moveheard=no |
| 17:53.16 | [TK]D-Fender | rwaite: #asterisk-dev awaits you. |
| 17:53.24 | [TK]D-Fender | Qwell: FTW |
| 17:53.47 | Qwell | That'll be $19.99 - please paypal qwell@cayman-islands-not-a-tax-shelter.com |
| 17:56.00 | Katty | sends donation to Qwell so he can buy new shoes. |
| 17:56.10 | Qwell | what's wrong with my chucks? |
| 17:56.17 | Katty | they're not pretty. |
| 17:56.20 | Qwell | yes they are |
| 17:56.24 | Katty | post gifs. |
| 17:56.31 | Qwell | I even got complimented on them by the cleaning lady like 3 weeks ago |
| 17:56.40 | Katty | that was 3 weeks ago. |
| 17:56.44 | Qwell | this is true |
| 17:56.44 | Katty | those shoes are old and icky now. |
| 17:56.54 | Qwell | which, for chucks, is a good thing |
| 17:57.18 | Katty | Qwell: http://youtube.com/watch?v=wCF3ywukQYA |
| 17:57.36 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 17:58.22 | Katty | Qwell: go forward about 1 minute. |
| 17:58.31 | Qwell | youtube != work |
| 17:58.38 | MaliutaLap | Katty: if you're giving Qwell new shoes, can I have a new hip? |
| 17:59.36 | Katty | Qwell: look at it later then (= |
| 17:59.44 | *** join/#asterisk kensukeido (i=be0210a1@gateway/web/ajax/mibbit.com/x-38fdd34da695990a) |
| 18:06.12 | *** join/#asterisk hi365_m (n=hi365@213.151.63.7) |
| 18:07.27 | *** join/#asterisk implicit (n=bayan@ip68-105-71-13.sd.sd.cox.net) |
| 18:07.43 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:09.13 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:10.35 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:11.47 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 18:16.54 | Qwell | Katty: msg it to me, so I have the link at home |
| 18:18.15 | codestr0m | I'm getting one way audio only from one provider.. (I can't hear them) what should I look for in the sip debug or would tcpdump looking for the udp packets be more helpful? |
| 18:19.46 | *** join/#asterisk nny_2 (n=Scott_My@64.203.244.146) |
| 18:19.54 | [TK]D-Fender | codestr0m: read up : |
| 18:19.56 | [TK]D-Fender | ~sipnat |
| 18:19.57 | jbot | [~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 |
| 18:19.58 | [TK]D-Fender | ^^^^^^^^^ |
| 18:20.17 | rwaite | nat makes me want to drown babies |
| 18:20.32 | implicit | codestr0m: who's your provider |
| 18:20.35 | codestr0m | [TK]D-Fender: I'd think it was nat, but why only on this provider? |
| 18:20.35 | nny_2 | can i use a variable for the exten => value like exten => $NUMBER,1,Goto(Something)? |
| 18:20.48 | codestr0m | implicit: this is an ipkall number being forwarded.. should be dead simple |
| 18:21.01 | [TK]D-Fender | nny_2: variable no, constant yes. |
| 18:21.28 | codestr0m | I'm not in control of the router atm. I'm behind some stupid cisco 26xx I think |
| 18:21.34 | ManxPower | nny_2: NO! |
| 18:21.49 | nny_2 | THANKS! |
| 18:21.51 | nny_2 | :) |
| 18:21.52 | codestr0m | so can't do any port forwarding or the lazy setting up the phone as the dmz |
| 18:22.08 | rwaite | i made my * in the dmz, because dd-wrt sux. |
| 18:22.11 | [TK]D-Fender | codestr0m: Without forwarding you're FUBAR'd |
| 18:22.27 | [TK]D-Fender | hands codestr0m a tissue |
| 18:22.31 | ManxPower | codestr0m: If Asterisk is behind NAT then you MUST do port forwarding or it will not work. However, if you just have a phone behind NAT then you must NOT fort forward. |
| 18:22.35 | jeev | hi fender |
| 18:23.49 | implicit | and if both are behind NAT? :-D |
| 18:24.25 | codestr0m | [TK]D-Fender: no. asterisk is *not* behind nat.. ipkall is not behind nat and the client *is* behind nat |
| 18:24.28 | [TK]D-Fender | implicit: READ THE GUIDE |
| 18:24.39 | implicit | i'm just kidding man |
| 18:24.45 | implicit | chill the fuck out |
| 18:24.48 | codestr0m | with the other two providers.. I register and it all works fine.. with ipkall I don't. that's the only difference |
| 18:24.52 | [TK]D-Fender | codestr0m: Then prevent reinvites, enable qualify and you're good to go. |
| 18:25.21 | codestr0m | [TK]D-Fender: thanks I'll give it a shot.. I'm doing that on the other two, but didn't think it's needed with this |
| 18:25.34 | implicit | codestr0m, there is a rtp deadlock issue in asterisk too |
| 18:25.40 | implicit | i'm working w/ oej on it and it's confirmed |
| 18:26.00 | implicit | has to do with symmetric rtp |
| 18:26.06 | implicit | even when everyone is in front of a nat |
| 18:26.16 | implicit | try a Playback(silence/1|noanswer) |
| 18:26.22 | implicit | before doing anything else |
| 18:26.25 | implicit | just to make sure that's not it |
| 18:26.45 | *** join/#asterisk Levon1 (n=Levon@nat/yahoo/x-f71263ef494729e9) |
| 18:27.32 | Qapf | hey guys, the wiki page on voip info for faxing is a bit of a mess, i was wondering if someone could point me the most reliable way of setting up faxing involving my asterisk server. be it passthrough to a real fax machine or taking the faxes on the box itself. |
| 18:27.45 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:28.31 | codestr0m | [TK]D-Fender: fixed. thanks.. |
| 18:28.34 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 18:28.45 | *** join/#asterisk silvertip257 (n=chatzill@unaffiliated/silvertip257) |
| 18:28.46 | [TK]D-Fender | codestr0m: All good? |
| 18:28.50 | *** part/#asterisk Levon1 (n=Levon@nat/yahoo/x-f71263ef494729e9) |
| 18:28.51 | codestr0m | implicit: I wasn't hitting the rtp deadlock issue |
| 18:29.06 | implicit | ok |
| 18:29.13 | codestr0m | yeah.. getting audio both ways.. seemed so strange for me to have to set that though |
| 18:29.15 | silvertip257 | [TK]D-Fender: I got the Asterisk SIP working on my little blue box |
| 18:29.16 | [TK]D-Fender | codestr0m: Glad to hear. You're welcome. |
| 18:29.23 | [TK]D-Fender | codestr0m: Its completely normal. |
| 18:29.37 | [TK]D-Fender | silvertip257: taht the WRT install? |
| 18:29.42 | silvertip257 | yes sir |
| 18:29.43 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:30.06 | silvertip257 | [TK]D-Fender: but of course now I'm not able to get it to register (IAX) with a voip tele provider... |
| 18:30.32 | codestr0m | [TK]D-Fender: I'm already not reinviting on the uas-uac so I assumed the ipkall would just foward and not reinvite to the client directly.. |
| 18:31.00 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:31.14 | codestr0m | now my next little outstanding issue.. 1) setup gtalk 2) hack skype :P |
| 18:31.23 | silvertip257 | haha codestr0m |
| 18:31.25 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 18:31.54 | silvertip257 | ponders little & outstanding in one sentence ... ;-) |
| 18:33.21 | silvertip257 | [TK]D-Fender: if you have time, could I run my config past you? |
| 18:33.51 | [TK]D-Fender | silvertip257: Go compare it against your itsp's sample configs and check that its routing/firewall is not screwing you like before. |
| 18:34.38 | silvertip257 | [TK]D-Fender: firewall has been opened up on WAN port & I've been loolking at their example configs, but I don't know enough asterisk to fit that in with what I've got (sounds simple, but i'm complicated ;-) ) |
| 18:34.41 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 18:35.32 | [TK]D-Fender | silvertip257: get Googling and just try stuff for a while |
| 18:35.59 | *** join/#asterisk jpcansa (n=jpbenavi@190.10.19.145) |
| 18:36.55 | silvertip257 | [TK]D-Fender: I have been (and will continue to) |
| 18:36.55 | redax | if my voip provider sometimes (for international calls) don't send ringback tones... |
| 18:37.06 | redax | what should I do? use the 'r' Dial option? |
| 18:37.19 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:39.40 | redax | the main problem using 'r' Dial option is, if the called party is busy I get ringing even if I should hear the remote busy tone... |
| 18:40.31 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 18:40.36 | *** join/#asterisk PakiPenguin (n=junaid@linuxpakistan/admin/pakipenguin) |
| 18:40.38 | PakiPenguin | hi |
| 18:40.44 | ManxPower | redax: correct and that is why you should not use it. |
| 18:40.45 | Hertzy3 | ~centos52bug |
| 18:40.45 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 18:40.54 | ManxPower | complain to your provider |
| 18:40.59 | PakiPenguin | does asterisk support sip-t or sip-i? |
| 18:41.26 | redax | ManxPower: ok. but outbound call without ringback tone is annoying... just silence, and somebody picks up the phone ... :) |
| 18:42.22 | redax | ManxPower: I see... they told me, the uplink provider didn't give the ringback tone... |
| 18:42.51 | Qwell | tell them to yell at the uplink provider :p |
| 18:43.06 | Qwell | besides, if somebody is calling internationally, your ringing tone is going to sound very odd and unfamiliar to them |
| 18:43.21 | [TK]D-Fender | PakiPenguin: Which are what exactly? |
| 18:44.19 | redax | Qwell: ok. then what to do? :) give 'r' option for internation meanwhile no for national... |
| 18:44.31 | jpcansa | how can I disconnect a SIP registered exten from CLI ? |
| 18:44.35 | PakiPenguin | [TK]D-Fender: sip-t is for the telephony , sip for telephones |
| 18:44.37 | Qwell | that would "work", but like I said, the caller may not know what that "odd sound" is. |
| 18:45.13 | implicit | pakipenguin, there is no sip-t support in asterisk |
| 18:45.22 | PakiPenguin | implicit : no sip-i aswell? |
| 18:45.33 | implicit | i don't even think it works w/ multi-part body processing |
| 18:46.02 | implicit | sorry |
| 18:47.13 | PakiPenguin | hmm |
| 18:47.17 | implicit | even w/ the parsing there is no way right now for asterisk to decode the info and let you use it in dialplan |
| 18:47.23 | implicit | i do my sip-t stuff in openser |
| 18:47.37 | implicit | doing some custom additions and then textops to clean up the multipart body |
| 18:47.43 | HonestWorker | Ho ho ho...after several characters typed and lots of painfull copy and paste, I have change my dialplan to coordinate the CALLERID. |
| 18:47.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:48.10 | implicit | PakiPenguin: what are you trying to do? just get calls to work? or actually process the ISUP info |
| 18:48.25 | PakiPenguin | implicit , i just want calls to work |
| 18:48.37 | PakiPenguin | i see a patch for sip-t in bugs.digium |
| 18:48.46 | HonestWorker | EmleyMoor, hello. Are you still around? I've decided to branch through a GotoIf and then run Set(CALLERID(num)=number) before calling the Dial() application |
| 18:48.55 | implicit | ah if u just want calls to work i can show u how to do it in openser easily |
| 18:49.03 | implicit | that way you could even save the ISUP info |
| 18:49.13 | implicit | PakiPenguin: mesage me, i'd like to see the INVITes you get |
| 18:50.45 | nny_2 | well anyone got a nice sharp sword? |
| 18:51.47 | *** join/#asterisk emiller (n=ed@UNUSED-216-37-164-100.UNUSED.epix.net) |
| 18:52.08 | nny_2 | May commit seppuku today. First T1 system going live. I assume the minor differences between using FXO Cards and using a T1 have been thoroughly overlooked |
| 18:52.38 | nny_2 | have the dialing sequences offset, think i answered all the right questions in regards to PRI type etc |
| 18:53.06 | nny_2 | have the channels set up in my zaptel.conf |
| 18:53.20 | [TK]D-Fender | nny_2: I have several, and even 1 for sale :) |
| 18:53.31 | nny_2 | cards? |
| 18:53.42 | nny_2 | er wait lol |
| 18:53.42 | tzafrir_laptop | swords? |
| 18:53.46 | [TK]D-Fender | nny_2: No... swords :) |
| 18:54.05 | nny_2 | eh the card as a weapon would have been a lot more interesting for disembowelment |
| 18:55.05 | emiller | having problems getting my cdr to insert into mysql. trying to install asterisk-addons and i then get these errors: anyone make any sense? http://pastebin.ca/1081042 |
| 18:55.14 | nny_2 | besides, wouldn't i rather just rent a sword for seppuku? not like i'll use it again |
| 18:55.35 | emiller | since M1s3ry isn't allowed to help me :P |
| 18:55.38 | [TK]D-Fender | nny_2: Sorry, hard to sell a stained weapon like that :) |
| 18:55.46 | M1s3ry | :p |
| 18:56.17 | [TK]D-Fender | emiller: What ver of *, and Addons? |
| 18:56.25 | M1s3ry | I could, but it's something we don't particularly support and I have about this much ==> | | <== knowledge on it. :D |
| 18:57.01 | emiller | im using an svn ver 1.4-r131491 and addons 1.2 |
| 18:57.25 | emiller | haha M1s3ry, its cool. |
| 18:57.26 | [TK]D-Fender | emiller: You cannot mix versions like that |
| 18:57.33 | [TK]D-Fender | eillYou need 1.4 addons |
| 18:57.48 | emiller | ill try that. Thanks D-Fender. |
| 18:58.01 | M1s3ry | now i feel bad... |
| 18:58.32 | emiller | haha |
| 18:58.32 | M1s3ry | because I didn't even begin to ask that, I just went from "how can I help you" to"sorry we don't support that" |
| 18:58.42 | M1s3ry | I could've told you that... |
| 18:58.53 | M1s3ry | cries at his cubical |
| 18:59.12 | emiller | as soon as you heard open source, i heard u mutter 'cheap!' |
| 18:59.16 | emiller | :P |
| 18:59.49 | M1s3ry | blasphemy! |
| 19:01.22 | nny_2 | so what say the channel on qualify=yes vs. qualify=no for phones on the local network? |
| 19:01.58 | *** join/#asterisk Dovid (n=Dovid@tony09-121-90.inter.net.il) |
| 19:02.26 | ManxPower | nny_2: qualify has always caused me problems |
| 19:02.45 | ManxPower | if your local phones are so unstable that you need qualify then you have much bigger problems |
| 19:02.49 | *** join/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 19:03.03 | nny_1 | sorry if i missed a response, had to de VPN right quick |
| 19:04.04 | ManxPower | (2:02:26 PM) ManxPower: nny_2: qualify has always caused me problems |
| 19:04.04 | ManxPower | (2:02:45 PM) ManxPower: if your local phones are so unstable that you need qualify then you have much bigger problems |
| 19:07.06 | nny_1 | ManxPower: ok good response. Wasn't sure if qualify was a good thing (tm) or a fix for certain issues |
| 19:07.39 | ManxPower | nny_1: if your phone is on an unstable link it could solve more problems than it solves, but other than that.... |
| 19:07.48 | ManxPower | ..er..solve more problems than it creates |
| 19:09.16 | keith4 | i was always under the impression that qualify was used with NAT |
| 19:09.29 | [TK]D-Fender | keith4: it is |
| 19:10.10 | keith4 | and that it wasn't need without NAT |
| 19:10.18 | nny_1 | ManxPower: k |
| 19:10.22 | keith4 | er, needed |
| 19:10.53 | ManxPower | keith4: qualify basically does a SIP ping. A side effect of that is it can keep NAT translation port mapping active. There are better ways to accomplish that |
| 19:11.14 | nny_1 | I am getting DEBUG[20089]: res_musiconhold.c:577 monmp3thread: Read 600 bytes of audio while expecting 640. There is no apparent issue, but the debug message shows up a lot in my console. I am looking for the cli to disable it. Any advice? |
| 19:12.54 | ManxPower | nny_1: MP3 is not recommended |
| 19:13.07 | redax | is callprogress=yes and busycount=X are useless on PRI zap interfaces? |
| 19:13.23 | [TK]D-Fender | nny_2: Make sure they are non-VBR, with no ID3 tags |
| 19:13.31 | [TK]D-Fender | redax: Yes |
| 19:13.32 | nny_1 | ManxPower: yeah I know.. it's for an audio stream that these brokers listen to through the phone. I was not all about it myself. |
| 19:13.34 | ManxPower | redax: totally completely 100% useless on PRI |
| 19:13.46 | ManxPower | nny_1: It sucks to be you |
| 19:14.17 | nny_1 | ManxPower: nope. Although a way to disable the error repeating would be elegant enough |
| 19:14.41 | redax | that's bad. I have to listen to the busy tone for 40-50seconds until I get cause 17 |
| 19:14.54 | ManxPower | redax: no you don't. |
| 19:15.13 | ManxPower | maybe you did something stupid like set progrogressinband or whatever that option is called |
| 19:15.17 | redax | ok. I can hangup ;-) |
| 19:15.20 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 19:15.46 | ManxPower | redax: our hangups process in less than 1/4 of a second |
| 19:15.48 | codestr0m | do I need res_jabber for gtalk to work? |
| 19:15.56 | Qwell | yes |
| 19:16.06 | Qwell | codestr0m: ^ |
| 19:16.13 | codestr0m | Qwell: ok. thanks |
| 19:16.30 | redax | ManxPower: it's a PRI link to our mobile provider... |
| 19:16.32 | ManxPower | redax: you are not supposed to year a busy tone from a PRI |
| 19:16.36 | Cart_ | I have a problem that some module will cause asterisk not to start. asterisk doesn't tell which one. how can I find out it? |
| 19:17.02 | redax | am I missconfigured my zapata channels? |
| 19:17.05 | ManxPower | redax: we can't do much about sucky providers |
| 19:17.15 | ManxPower | Cart_: I can answer your question on #asterisk-cli |
| 19:17.32 | ManxPower | redax: only if you used the option I mentions |
| 19:17.51 | nny_1 | ManxPower: i have disabled the mp3 streaming, and told them to use the linein with their streamin client of choice |
| 19:18.07 | nny_1 | ManxPower: hopefully there is not a reason why that error would come from /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw |
| 19:18.35 | ManxPower | nny_1: Hint: Once I say "it sucks to be you" I'm not interested in helping you anymore. |
| 19:18.47 | redax | ManxPower: callingprogress you mean? |
| 19:19.05 | ManxPower | redax: no priindication |
| 19:19.18 | redax | ManxPower: would you be so kind and take a look at my zapata.conf : http://pastebin.ca/1081079 |
| 19:19.40 | ManxPower | redax: what country are you in? |
| 19:20.17 | redax | ManxPower: Europe/Hungary |
| 19:21.23 | ManxPower | redax: your provider sucks. I can't help with that. |
| 19:22.22 | redax | ;) thanks |
| 19:22.27 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:22.48 | redax | ManxPower: it's T-Mobile ;) |
| 19:25.59 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 19:29.16 | codestr0m | to do jabber test.. instead of calling the label/context gtalk_account I should call it asterisk? |
| 19:31.24 | *** join/#asterisk pjezek (n=pj@193.85.164.154) |
| 19:33.00 | tompaw | hello |
| 19:33.17 | pjezek | is legal to use Return('some value') in extensions.ael? I still got warnings, like: application call to Return affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! |
| 19:33.21 | tompaw | what's 0x80100 and how's it different from 0x100? |
| 19:33.41 | tompaw | I just noticed that DTMF works properly only with that 0x80100, but I don't know how to force it. |
| 19:34.01 | codefreeze-lap | pjezek: no return vals from macros yet. |
| 19:34.34 | codefreeze-lap | The Return() you are using is an app. "core show application return" for details |
| 19:35.30 | [TK]D-Fender | pjezek: AEL is not a separate dialplan language. It gets parsed back and can't do more than you could do yourself. |
| 19:37.18 | tompaw | Strange. According to google, 0x80100 uses g729 + video codec (?). |
| 19:37.50 | tompaw | What does video codec have to do with my GSM gateways and why DTMFs work only when it's present? |
| 19:37.53 | pjezek | yes, I know, I try to use some context as function block, that I call with Gosub for some processing and expecting some return value after return from this function, |
| 19:38.35 | pjezek | it working quite well, except this warning during ael compiling/reloading |
| 19:38.44 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 19:39.39 | codefreeze-lap | pjezek: It warns, but it won't refuse to let you use the control flow apps. Sometime, mayhaps, we can do funcs with retvals in AEL. |
| 19:40.56 | nny_1 | the telco is complaining the dchannel is in lockout on this T1, |
| 19:41.04 | nny_1 | i have dchannel = 24 in zaptel.conf is this needed? |
| 19:41.13 | nny_1 | er dchan=24 |
| 19:41.15 | nny_1 | sorry |
| 19:43.17 | pjezek | OK, I asked only for confirmation, if something wrong can occur, so far, it working for me:) |
| 19:45.26 | Peri | alright, little bit of fun here |
| 19:45.31 | codestr0m | for my gtalk. I think I have everything setup correctly, but when I call or answer a call.. I just hear myself.. I don't even think it's hitting the server like an echo test either |
| 19:45.43 | Peri | I'm trying to set up my asterisk server as pure voicemail for a 3rd party AS |
| 19:45.54 | Peri | it works in all respects as a voicemail server no problem |
| 19:46.00 | Peri | where i'm having issues is with the MWI |
| 19:47.18 | Peri | now, as the asterisk server takes no registrations itself and actually doesn't register with the AS, i've tried to shortcut and use host= <IP> in the users.conf for the individual user |
| 19:47.32 | Peri | the IP in that btw is the IP of the AS |
| 19:47.55 | Peri | since the end users are dynamic and the AS is supposed to be able to forward the notify packet |
| 19:48.48 | Peri | the problem i'm running into is that the notify packet is only putting the IP in the TO field |
| 19:49.20 | Peri | To: <sip:65.xx.xx.xx> instead of To: <sip:<phonenumber>@65.xx.xx.xx> |
| 19:49.39 | Peri | anyone have any ideas on how to get that phone number in there |
| 19:50.02 | Peri | the rest of the notify packet is perfect btw |
| 19:50.41 | Peri | i think i killed the room >.< |
| 19:50.48 | *** join/#asterisk el_4_jinete (n=el_4_jin@190.144.46.3) |
| 19:50.59 | el_4_jinete | Hi everyone! |
| 19:51.32 | *** join/#asterisk bsaxon (n=bsaxon@205.sub-75-251-33.myvzw.com) |
| 19:51.41 | tompaw | what options do I have for dtmf with g729? |
| 19:51.57 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 19:52.35 | el_4_jinete | Someone to can help me... |
| 19:52.57 | el_4_jinete | I've got this message when I reload asterisk from the Cli |
| 19:53.00 | el_4_jinete | <PROTECTED> |
| 19:53.14 | el_4_jinete | What's that |
| 19:57.23 | *** join/#asterisk yakimbo64 (n=yakimbo6@196.20.72.121) |
| 19:58.56 | silvertip257 | I have a functional asterisk PBX that allows internal sip calls. Now I'm trying to hook it up to a ITSP with IAX. It acts like it's calling in, but the * debug says "No one is available to answer at this time". I've been debugging IAX for a long while now, but nothing really stands out to me...the ports are forwarded to the PBX |
| 20:00.06 | [TK]D-Fender | silvertip257: WAHT says that? |
| 20:00.29 | silvertip257 | [TK]D-Fender: the end of the call when it's hung up in the * CLI debug |
| 20:00.48 | [TK]D-Fender | silvertip257: pastebin the complete attempt |
| 20:00.54 | silvertip257 | ok i will |
| 20:01.29 | Peri | [TK]D-Fender there's nothing like a touser similar to a fromuser is there? |
| 20:01.41 | HonestWorker | I have got to go |
| 20:01.44 | HonestWorker | bye bye all |
| 20:03.20 | [TK]D-Fender | Peri: thats the TO: from your dial. |
| 20:03.39 | Peri | i know :-/ |
| 20:03.50 | [TK]D-Fender | Peri: Dial(SIP/provider/thisistouser) |
| 20:03.58 | Peri | right |
| 20:04.06 | silvertip257 | [TK]D-Fender: http://pastebin.com/d1bfa13c0 |
| 20:05.43 | [TK]D-Fender | silvertip257: Oh yeah... ANCIENT version. I'm not sure about protocol compatibility going that far back. might be an issue |
| 20:05.44 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:06.01 | silvertip257 | [TK]D-Fender: yikes ... well then - maybe use SIP ? |
| 20:06.16 | [TK]D-Fender | silvertip257: definitely |
| 20:06.30 | silvertip257 | so that is really my only other option? |
| 20:06.41 | silvertip257 | or easiest?... |
| 20:07.54 | silvertip257 | [TK]D-Fender: ^ |
| 20:08.00 | [TK]D-Fender | silvertip257: Best choice. |
| 20:08.06 | silvertip257 | ok thanks [TK]D-Fender |
| 20:08.15 | [TK]D-Fender | silvertip257: My theory is never use IAX unless you have to. |
| 20:08.34 | codestr0m | [TK]D-Fender: would you be able to help with chan_gtalk.. my buddy gets the call, but when we answer. we hear nothing.. when I call I hear myself.. (which seems really strange), but with no delay so can't be like an echo request |
| 20:08.36 | silvertip257 | [TK]D-Fender: somewhere I read IAX is recommended between * systems ... |
| 20:08.41 | [TK]D-Fender | silvertip257: ITSP's have version issues, and there have been many quality issues. |
| 20:09.04 | [TK]D-Fender | silvertip257: </pimp> :) |
| 20:09.13 | silvertip257 | [TK]D-Fender: I'm going to fire up a sip config for that and try it out - thanks. I'll report success |
| 20:09.22 | silvertip257 | heh |
| 20:10.48 | codestr0m | I'm thinking something is wrong in my extensions.conf |
| 20:10.49 | codestr0m | [google-in] |
| 20:10.49 | codestr0m | exten => s,1,NoOp( Call from Gtalk ) |
| 20:10.56 | _Raptor_ | does any have a recommendation for an IAX client for win, linux and macos? |
| 20:11.01 | codestr0m | cause I'm not seeing the noop |
| 20:11.58 | [TK]D-Fender | codestr0m: PASTEBIN is your friend, and you'll want channel debug enabled for this |
| 20:16.02 | silvertip257 | [TK]D-Fender: here's the page for my (test) provider -http://www.callwithus.com/configuration - I'm wondering about the register stuff for SIP - which is the actual username [username:password@username] and which is the entry for my info below? Thanks. |
| 20:16.16 | silvertip257 | half-way down on the page is the Asterisk stuff ^ |
| 20:17.13 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 20:18.00 | [TK]D-Fender | silvertip257: REGISTER => user:pass@sip.callwithus.com/yourdidnumber |
| 20:18.21 | [TK]D-Fender | silvertip257: yeah they goofed... th @host should match the one you see listed up top |
| 20:18.27 | Peri | so i guess my question is, is there any way to have the MWI NOTIFY packet properly fill in the TO field? |
| 20:18.44 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 20:19.18 | [TK]D-Fender | _Raptor_: ... |
| 20:19.20 | [TK]D-Fender | ~zoiper |
| 20:19.20 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 20:19.22 | [TK]D-Fender | ^^^ |
| 20:19.51 | _Raptor_ | [TK]D-Fender: thanks |
| 20:20.44 | silvertip257 | [TK]D-Fender: you're right - I should have compared it to the iax one ... which is why I chose the iax ;-) |
| 20:20.45 | codestr0m | [TK]D-Fender: http://codepad.org/xAei8Gbc |
| 20:20.59 | codestr0m | that's me attempting 1 outbound call |
| 20:21.29 | tompaw | is there a way to DEBUG/SNIFF dtmf sent during call? |
| 20:22.22 | *** part/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 20:22.25 | [TK]D-Fender | tompaw: "set debug 10 |
| 20:23.09 | [TK]D-Fender | codestr0m: "set verbose 10" and retry. If you get no more then I can't help you further. |
| 20:23.14 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:23.42 | silvertip257 | Would it be a good idea to try and disable IAX if I'm only using SIP? Is that entirely possible? Thanks. |
| 20:23.43 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 20:24.03 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:24.18 | tompaw | [TK]D-Fender: thx, is that supposed to be showing those dtmfs on CLI? cause it doesn't. |
| 20:24.27 | Peri | silvertip257: yes |
| 20:24.51 | silvertip257 | Peri: I'm trying to find how with Google.. |
| 20:25.16 | Peri | silvertip257: it's not really necessary to disable it, just ensure you have nothing configured to use it |
| 20:25.27 | [TK]D-Fender | silvertip257: entirely unnecessary |
| 20:25.32 | Peri | silvertip257: it will make debugging a lot easier |
| 20:25.36 | silvertip257 | I'd rather disable it if it won't hurt -- so dont disable |
| 20:26.03 | *** join/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 20:27.43 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 20:28.22 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 20:28.29 | shmaltz | I have a provider that has more than one IP address for failover, how do I configure that in asterisk? it's for the same call, meaning if one IP fails the other will take over. |
| 20:28.49 | [TK]D-Fender | shmaltz: Setup separate peers and dial them back to back |
| 20:29.25 | shmaltz | [TK]D-Fender, that much I know :P, how can I give the same peer 2 IP address? |
| 20:29.25 | scooby2 | Anything better (commercial or free) for call center monitoring than QueueMetrics? |
| 20:29.25 | tompaw | is there a way to debug dtmf sent during call / display them in CLI? |
| 20:29.25 | ManxPower | shmaltz: you use two differnet peers |
| 20:29.39 | ManxPower | tompaw: yes |
| 20:29.40 | tompaw | I don't even know which side of my call is causing the problems. |
| 20:29.48 | tompaw | ManxPower: how do I do that? |
| 20:29.52 | [TK]D-Fender | shmaltz: you CAN'T |
| 20:29.55 | ManxPower | Which is why [TK]D-Fender answered your question |
| 20:30.28 | [TK]D-Fender | shmaltz: when we tell you one way, figure that to other isn't possible |
| 20:30.40 | tompaw | and then was busy with somebody else. debug 10 doesn't show anything here. |
| 20:30.54 | [TK]D-Fender | shmaltz: You're welcome to rewrite chan_sip if you are displeased with it ;) |
| 20:30.56 | ManxPower | tompaw: the look at /etc/asterisk/logger.conf |
| 20:31.01 | shmaltz | Thanks [TK]D-Fender, ManxPower |
| 20:31.18 | ManxPower | shmaltz: Fortunatly the solution to calling two different servers was solved years ago. |
| 20:31.30 | shmaltz | [TK]D-Fender, I am ok with it, the provider gave me 4 IP address and I was just trying to configure tham |
| 20:31.34 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 20:31.34 | shmaltz | them **** |
| 20:31.42 | [TK]D-Fender | ok, cheakout time. Heading home. |
| 20:31.50 | ManxPower | shmaltz: you'd think the provider would use SRV or round robbin DNS for those sort of thing |
| 20:32.00 | shmaltz | [TK]D-Fender, it's not yet 5PM how dear you? |
| 20:32.01 | tompaw | ManxPower: that explains a lot and completes his answer. thanks. |
| 20:32.11 | drfreeze | Can someone clarify how I need to set the callgroup and the pickupgroup |
| 20:32.19 | shmaltz | ManxPower, no this is meant to actualy take over during a call |
| 20:32.23 | ManxPower | drfreeze: you can't use those options for |
| 20:32.25 | ManxPower | IAX2 |
| 20:32.36 | drfreeze | I have two offices and want *8 to work in each office, but don't want one office picking up the calls for another office |
| 20:32.59 | ManxPower | drfreeze: Are you using IAX2? |
| 20:33.07 | ManxPower | for the calls you want to pick up? |
| 20:33.09 | drfreeze | ManxPower: nop - all sip |
| 20:33.32 | drfreeze | or incoming zap calls |
| 20:33.34 | ManxPower | shmaltz: then perhaps they have only ONE SIP server and the rest are RTP. For moving RTP that's called reinvites and does not work with NAT |
| 20:33.48 | *** join/#asterisk anthm (n=anthm@m990736d0.tmodns.net) |
| 20:34.09 | ManxPower | drfreeze: Well don't put company's callgroup the same as the other company's pickupgroup |
| 20:34.37 | shmaltz | ManxPower, no sip server to register, just gateway switches signalling and rtp |
| 20:34.44 | ManxPower | shmaltz: but if they are moving the audio around, then everything I told you and everything anyone else told you on this channel is either wrong or does not apply to your situation |
| 20:34.47 | tompaw | hm... it reports the DTMF activity on one end of the call, doesn't say anything about forwarding/transcoding it though. |
| 20:34.48 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 20:35.20 | shmaltz | ManxPower, to be honest I dont know, I'm assuming it's for both |
| 20:35.29 | shmaltz | signal and rtp, but not register |
| 20:36.07 | ManxPower | drfreeze: Company A's SIP devices pickupgroup=1 callgroup=1 for Company B's SIP devices pickupgroup=2 and callgroup=2 |
| 20:36.23 | ManxPower | shmaltz: until you actually know something it's pretty pointless to ask questions here. |
| 20:36.38 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 20:36.54 | AlexTO | Hi There is someone can explain to me how send TXT MSG to a cells using Asterisk, what do i need? |
| 20:37.12 | ManxPower | I will answer further questions on #asterisk-cli |
| 20:37.27 | ManxPower | AlexTO: you can only do that with european carriers, not usa/canada carriers |
| 20:37.47 | shmaltz | A1exTO, a cell phone and some robotic interface to a robotic arm that communicates with asterisk |
| 20:38.21 | ManxPower | You can, of course, use standard carrier e-mail or web interface to send messages, but that has NOTHING whatsoever to do with Asterisk |
| 20:38.28 | AlexTO | ManxPower, Can i send TXT MSG to US using european carriers? |
| 20:38.46 | ManxPower | AlexTO: you would have to ask your european carrier? |
| 20:38.59 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 20:39.03 | teknoprep | hey all |
| 20:39.15 | teknoprep | when i press 7 to delete a msg... it deletes it.. but doesn't move onto the next message |
| 20:39.20 | teknoprep | i don't know why |
| 20:39.26 | ManxPower | But the Asterisk app_SMS is SPECIFICALLY for sending FSK encoded SMS mesages to carriers using LAND LINES |
| 20:39.43 | ManxPower | any other form of SMS is not covered in app_SMS |
| 20:39.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:40.04 | ManxPower | AlexTO: why don't you just use the standard carrier e-mail interface? |
| 20:40.44 | ManxPower | out of smokes. must leave |
| 20:41.05 | AlexTO | i'll do that, but i'm not familiar with that, can you give me some providers information? |
| 20:41.28 | teknoprep | ???? |
| 20:42.08 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 20:42.13 | AlexTO | ManxPower, if i have a PRI connected could i send SMS, i what you said? |
| 20:42.32 | AlexTO | is it what you said? |
| 20:43.38 | *** join/#asterisk masus (n=Ethem@85.106.162.213) |
| 20:44.12 | shmaltz | A1exTO, no you can't send SMS with PRI in the USA/Canada |
| 20:44.20 | drfreeze | ManxPower: thx. will try that tonight |
| 20:47.02 | AlexTO | i got it.. Thanks, Any carrier email interfase that you can suggest to me? |
| 20:47.04 | teknoprep | so doesanyone know why when i press 7 to delete a msg... it deletes it.. but doesn't move onto the next message... i have to press 6 to move on to the next message after deleting it.. i never had to do this before |
| 20:47.37 | keith4 | teknoprep: did you recently upgrade? |
| 20:47.43 | shmaltz | teknoprep, by never you mean in asterisk? |
| 20:47.53 | teknoprep | yes in asterisk i never had to |
| 20:48.03 | keith4 | isn't there an option in voicemail.conf that controls that behavior? |
| 20:48.03 | teknoprep | no still on 1.4.18-3 |
| 20:48.08 | shmaltz | teknoprep, you sure you had more than one message before? |
| 20:48.13 | keith4 | heh |
| 20:48.16 | teknoprep | yes |
| 20:48.27 | teknoprep | i'll add more now |
| 20:48.34 | teknoprep | i just left myself 4 to do some tests |
| 20:48.51 | teknoprep | i never had this happen where i press 7... and it just didn't move onto the next email after deleting it |
| 20:48.56 | teknoprep | does anyone know what setting this would be ? |
| 20:49.25 | keith4 | nextaftercmd |
| 20:49.33 | teknoprep | ? |
| 20:49.45 | keith4 | the setting |
| 20:49.50 | keith4 | is "nextaftercmd" |
| 20:50.01 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 20:50.37 | teknoprep | i have nextaftercmd=yes |
| 20:50.44 | teknoprep | in the vm_general.inc |
| 20:51.12 | keith4 | shrugs |
| 20:51.37 | silvertip257 | I'm still not able to dial an outside number and SIP that to my ITSP... here's some of my config > http://pastebin.com/d47683dd9 |
| 20:51.49 | *** join/#asterisk BlargMaN_ (n=blarg@63.97.88.54) |
| 20:52.47 | teknoprep | so thats it ? |
| 20:53.30 | errr | yep thats it |
| 20:53.34 | teknoprep | heh |
| 20:53.46 | teknoprep | so even tho i have it setup right |
| 20:53.50 | teknoprep | and its not working |
| 20:54.03 | BlargMaN_ | hello everyone.... I'm an * noob, and I was wondering if anyone could point me in the direction of a doc or a web site that list all the dependencies for the different portions of the * system |
| 20:54.20 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 20:55.06 | shmaltz | BlargMaN_, just do make when you hit an error install that dep |
| 20:55.32 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:55.56 | BlargMaN_ | i tried that, but some of the issues I'm having is like getting the imap VM to let me select it, and also the CDR through mySQL to let me select that too... |
| 20:56.14 | BlargMaN_ | if i can't select it, make can't try to build it |
| 20:56.18 | keith4 | this is weird. my outgoing CID isn't working anymore |
| 20:56.23 | teknoprep | my include for vm_general.inc was not inside the voicemail.conf |
| 20:56.50 | teknoprep | i don't know how the f it went away |
| 20:57.43 | Qwell | #include + voicemail.conf == bad |
| 20:57.50 | BlargMaN_ | i tried google'ing it, but i can't find a straight answer... |
| 20:58.04 | keith4 | isn't include + voicemail.conf explicitly listed as a no-no? |
| 20:58.20 | Qwell | teknoprep: I bellieve your #include will be removed any time Asterisk changes the file (read: when a user changes their password) |
| 20:58.23 | Qwell | keith4: yep |
| 20:58.55 | teknoprep | Qwell, wow that sucks |
| 20:59.18 | keith4 | doesn't it say not to do that, in the sample voicemail.conf? |
| 20:59.24 | *** join/#asterisk bsaxon (n=bsaxon@120.sub-75-202-123.myvzw.com) |
| 20:59.35 | keith4 | ; NOTE: Asterisk has to edit this file to change a user's password. This does |
| 20:59.35 | keith4 | ; not currently work with the "#include <file>" directive for Asterisk |
| 20:59.35 | keith4 | ; configuration files, nor when using realtime static configuration. |
| 20:59.35 | keith4 | ; Do not use them with this configuration file. |
| 20:59.35 | *** part/#asterisk scooby2 (n=scooby2@unaffiliated/scooby2) |
| 21:02.10 | *** join/#asterisk ibnolqaiyem (n=ibnolqai@41.196.251.156) |
| 21:03.38 | ibnolqaiyem | are these all thing i need to make voip connection btw two telephone,(asterisk server, digium card,analog phone or softphone)? |
| 21:04.02 | [TK]D-Fender | Qwell: nobody ever reads the BIG print |
| 21:04.25 | [TK]D-Fender | ibnolqaiyem: No need for a card, and for PHONES, I wouldn't recommend it anyways |
| 21:04.51 | [TK]D-Fender | ibnolqaiyem: For Analog phones, get ATA's instead |
| 21:05.25 | ibnolqaiyem | [TK]D-Fender, is diguim card for ip phone only? |
| 21:05.36 | keith4 | uhhhh |
| 21:06.00 | keith4 | digium makes analog and digital phone cards, neither of which you need for IP phones |
| 21:06.01 | ibnolqaiyem | [TK]D-Fender, is digium card for ip phone only? |
| 21:06.34 | [TK]D-Fender | ibnolqaiyem: No, I suggest them only to use physical LINES of some kind |
| 21:06.55 | [TK]D-Fender | ibnolqaiyem: to plug a phone in I suggest ATA's Far cheaper and more flexible |
| 21:07.24 | [TK]D-Fender | ibnolqaiyem: Where are you located? |
| 21:07.39 | ibnolqaiyem | [TK]D-Fender, egypt |
| 21:08.12 | [TK]D-Fender | ibnolqaiyem: Ok, I'm not sure who the best choice of retailer would be for you but look up the Linksys PAP2T-NA and SPA-2102 |
| 21:09.30 | ibnolqaiyem | keith4, what you mean about you say? |
| 21:09.54 | [TK]D-Fender | ibnolqaiyem: IP phones don't NEED cards. |
| 21:10.14 | ibnolqaiyem | [TK]D-Fender, what they need? |
| 21:10.28 | Qwell | IP |
| 21:11.03 | Dovid | what is being used in place of n+101 |
| 21:11.04 | Dovid | ? |
| 21:11.23 | [TK]D-Fender | ibnolqaiyem: they are network devices. Go read the book and get an understanding of the technologies * can work with. |
| 21:11.25 | [TK]D-Fender | ~book |
| 21:11.26 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:11.39 | [TK]D-Fender | Dovid: Dear God, catch up to 4 YEARS AGO |
| 21:12.05 | [TK]D-Fender | Dovid: read your applications instructions as to values it returns. |
| 21:12.18 | *** part/#asterisk kensukeido (i=be0210a1@gateway/web/ajax/mibbit.com/x-38fdd34da695990a) |
| 21:12.20 | Juggie | [TK]D-Fender, his question is really the wiki's fault |
| 21:12.22 | [TK]D-Fender | Dovid: You are expect to test these values and do whatever you want based on them |
| 21:12.24 | Juggie | its incredibally outo f date |
| 21:12.28 | Juggie | and i bet he read that there |
| 21:12.38 | [TK]D-Fender | Juggie: This has been documented through 2 books, and in the apps themselves. |
| 21:12.45 | [TK]D-Fender | Juggie: Sorry, no dice |
| 21:12.57 | Juggie | its not an excuse but i bet thats where he saw it |
| 21:13.06 | [TK]D-Fender | Juggie: And he's been coming here for years and not showing any more of a clue. |
| 21:13.23 | Juggie | i dont have a personal !seen database, i guess you do :) |
| 21:14.00 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 21:14.04 | [TK]D-Fender | Juggie: I know who I've been helping for a long time :) |
| 21:14.39 | *** join/#asterisk remedex (n=none@outside.guavanyc.com) |
| 21:16.50 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 21:18.08 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 21:19.22 | silvertip257 | [TK]D-Fender: I've set my externip, localip. It registers fine - the provider's system says I'm registered fine. I can't make any outbound calls though. Ports are open 5060&RTP ones - SIP now. |
| 21:20.12 | silvertip257 | here's snippets of my config and try > http://pastebin.com/d47683dd9 (Thanks!) |
| 21:20.36 | *** join/#asterisk jonsmith1982 (n=jon@82-47-175-34.cable.ubr01.donc.blueyonder.co.uk) |
| 21:21.12 | [TK]D-Fender | silvertip257: try again after doing "sip debug" in CLI |
| 21:21.19 | [TK]D-Fender | silvertip257: We need to see the packets. |
| 21:21.27 | silvertip257 | [TK]D-Fender: ok |
| 21:24.21 | nny_1 | this out to be interesting |
| 21:24.35 | nny_1 | i was using exten => _84398750xx,1,Goto(transfer,${EXTEN:8:2},1) as my DID routing |
| 21:24.44 | nny_1 | where XX = the extension number |
| 21:24.48 | nny_1 | DIDs changed |
| 21:24.54 | nny_1 | so now its XX + 50 |
| 21:25.29 | nny_1 | wondering if i can mangle the ${EXTEN:8:2} to add 50 the outcome of that variable |
| 21:25.33 | Dovid | TK: I still use it. i know its going out and time to learn new things |
| 21:25.49 | Dovid | i need to break em old habbits |
| 21:27.06 | [TK]D-Fender | Dovid: It was going out 4 years ago when 1.2 was introduced. |
| 21:27.13 | [TK]D-Fender | Dovid: Go read each apps INSTRUCTIONS. |
| 21:28.31 | [TK]D-Fender | nny_1: Can you provide a sample to make sure I'm on the same page as you? |
| 21:29.36 | *** join/#asterisk Katty (n=angela@adsl-209-30-144-78.dsl.stlsmo.swbell.net) |
| 21:30.26 | nny_1 | [TK]D-Fender: sure. DID 843-987-7261 goes to extension 11 (50+11), DID 843-987-7267 goes to 17 (50+17) etc etc |
| 21:30.39 | mags2 | just subtract 50 |
| 21:31.16 | nny_1 | [TK]D-Fender: originally they were supposed to get 5800-5849 but the provider decided it was not available after setting it up :\ |
| 21:31.19 | [TK]D-Fender | nny_1: exten => _84398750xx,1,Goto(transfer,$[${EXTEN:8:2} - 50],1) |
| 21:31.39 | nny_1 | cool I was gonna read up more on variable math.. i am weak in that area anyways |
| 21:31.42 | nny_1 | [TK]D-Fender: thanks |
| 21:31.51 | silvertip257 | [TK]D-Fender: old stuff plus some more http://pastebin.com/d5e62cede |
| 21:32.01 | [TK]D-Fender | nny_1: Not to fear... * is weak in it too :p |
| 21:32.33 | Dovid | TK: didnt know it went out in 1.2.X |
| 21:32.42 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 21:33.26 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 21:34.22 | *** join/#asterisk angryuser (n=sldf@78.115.236.230) |
| 21:35.24 | *** join/#asterisk whymarkwh (n=dsfsdfsd@196.211.34.2) |
| 21:35.34 | whymarkwh | hi there |
| 21:35.42 | whymarkwh | anyone done the dcap? |
| 21:35.48 | [TK]D-Fender | silvertip257: I don't see the outbound dial attempt generating any SIP details |
| 21:36.06 | MaliutaLap | dcap's whymarkwh |
| 21:36.27 | whymarkwh | yes for asterisk |
| 21:36.51 | Katty | [TK]D-Fender: ohai |
| 21:36.56 | whymarkwh | where can one find some sample test to better prepare for the cetification? |
| 21:37.19 | silvertip257 | [TK]D-Fender: ok (sorry * is foreign to me) ... what could be causing it? |
| 21:37.29 | [TK]D-Fender | Katty: Mew. I'm off depopulating the planet again :) |
| 21:37.44 | Katty | [TK]D-Fender: oh that sounds fun. how many to go? |
| 21:37.49 | [TK]D-Fender | silvertip257: Not sure. Stop masking stuff except passwords and try again |
| 21:38.03 | [TK]D-Fender | Katty: Only just begun... haven't left the first country yet :) |
| 21:38.48 | Katty | oh boy. tons of fun still to go! |
| 21:39.19 | [TK]D-Fender | continues on his path to genocide |
| 21:39.35 | MaliutaLap | [TK]D-Fender: well I started in .au ... you go east, I'll go west and I'll meet you at the Danube |
| 21:40.47 | MaliutaLap | yes; it was chlorine I put in the water supply. Chlorine, strichnine, whats the difference? ;) |
| 21:42.56 | [TK]D-Fender | MaliutaLap: C21H22N2O2 -vs- Cl ;) |
| 21:43.36 | MaliutaLap | [TK]D-Fender: but they both end with an "een" sound ;) |
| 21:44.02 | [TK]D-Fender | MaliutaLap: Only Madagascar & north Africa to go |
| 21:45.55 | [TK]D-Fender | dammit, Madagascar close their airport |
| 21:46.10 | tzanger | haha you're playing that again |
| 21:46.19 | *** part/#asterisk nny_1 (n=Scott_My@64.203.244.146) |
| 21:46.59 | tzanger | [TK]D-Fender: I grabbed a blues guitar book today |
| 21:47.24 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 21:47.54 | [TK]D-Fender | tzanger: Cool. |
| 21:48.13 | GhOnDiE | you playing that game with the virus thing? |
| 21:48.17 | GhOnDiE | cant remember the name |
| 21:48.20 | tzanger | pandemic 2 |
| 21:48.25 | GhOnDiE | thats the one |
| 21:48.26 | GhOnDiE | yeah |
| 21:48.40 | GhOnDiE | i was playing it for a good couple of hours last night |
| 21:52.51 | silvertip257 | [TK]D-Fender: here's the config and some debug > http://pastebin.com/d23c7cbd7 |
| 21:53.42 | [TK]D-Fender | silvertip257: CANCEL sip:97175827740@192.168.2.1 SIP/2.0 |
| 21:53.51 | [TK]D-Fender | silvertip257: Looks like you're cancelling as soon as you start |
| 21:53.59 | [TK]D-Fender | silvertip257: Watch what you're doing |
| 21:54.06 | silvertip257 | [TK]D-Fender: it just sits there and doesn't ring |
| 21:54.15 | silvertip257 | i'll continue doing more |
| 21:54.20 | [TK]D-Fender | silvertip257: Ekiga seems to be cancelling |
| 21:54.24 | silvertip257 | oh |
| 21:54.36 | silvertip257 | hmm |
| 21:54.42 | *** join/#asterisk mmarcost (n=mmarcos@151.193.220.28) |
| 21:55.11 | mmarcost | what's the most stable solution for h323? h323, oh323, ooh323? |
| 21:59.45 | *** join/#asterisk quaqo (n=quaqo@62.13.173.89) |
| 22:02.37 | silvertip257 | [TK]D-Fender: something has to be wrong - I left ekiga go until it hangs up and I dont see any cancels... |
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| 22:03.54 | *** join/#asterisk Xaviertoor (n=Xavierto@200-146-243-009.xf-static.ctbcnetsuper.com.br) |
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| 22:06.19 | *** join/#asterisk SplasPood (i=jwb@69.9.183.119) [NETSPLIT VICTIM] |
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| 22:06.19 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) [NETSPLIT VICTIM] |
| 22:06.19 | *** join/#asterisk grantm (n=grant@68.142.138.4) [NETSPLIT VICTIM] |
| 22:06.19 | *** join/#asterisk ibnolqaiyem (n=ibnolqai@41.196.251.156) [NETSPLIT VICTIM] |
| 22:06.19 | *** join/#asterisk Corydon76-dig (i=silver@pdpc/supporter/bronze/Corydon76-home) |
| 22:06.19 | *** mode/#asterisk [+oo file Corydon76-dig] by irc.freenode.net |
| 22:07.49 | [TK]D-Fender | *b00m* |
| 22:08.01 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:10.42 | MaliutaLap | *t15H* |
| 22:11.05 | silvertip257 | [TK]D-Fender: Ekiga says "Security check failed" - but what does it know ;-) ... |
| 22:12.22 | silvertip257 | I'll be back later [TK]D-Fender |
| 22:25.16 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 22:25.34 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:27.47 | drfreeze | Hello |
| 22:28.11 | drfreeze | Anyone up for a little brain storming? |
| 22:28.40 | drfreeze | I am setting up polycom 550 phone in 2 offices connected by a T1 - on 122.x and 121.x networks |
| 22:28.48 | *** join/#asterisk remibemol (n=remibemo@81-66-205-202.rev.numericable.fr) |
| 22:29.02 | drfreeze | the * server is at 121.50 |
| 22:29.24 | drfreeze | the phone at the 121.x office register just fine |
| 22:29.35 | drfreeze | but the phones at the 122.x office don't |
| 22:30.38 | drfreeze | The <mac>-app.log files on a 122.x phone says: 0101000021|cfg |3|00|Phone successfully provisioned |
| 22:33.05 | [TK]D-Fender | drfreeze: SIP DEBUG <- |
| 22:33.14 | drfreeze | looking for ideas on where to check to see why phones are not registering |
| 22:33.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:35.15 | tompaw | hello again. |
| 22:35.28 | tompaw | that's one crappy weather out there, eh? |
| 22:36.50 | drfreeze | tompaw: you in austin? |
| 22:37.03 | drfreeze | [TK]D-Fender: well, the phones aren't even registering |
| 22:37.18 | tompaw | drfreeze: nope, in Krakow. |
| 22:37.21 | [TK]D-Fender | drfreeze: if you no packets on boot then your networking or configs are bad |
| 22:37.26 | drfreeze | they have a IP address |
| 22:37.35 | [TK]D-Fender | ooooooohhhh |
| 22:37.48 | [TK]D-Fender | So does pretty much every TCP/IP device. |
| 22:38.04 | [TK]D-Fender | drfreeze: Now to to look or show something MEANINGFUL |
| 22:38.35 | drfreeze | [TK]D-Fender: what good is sip debug if the phones don't register? |
| 22:38.49 | tompaw | what would you guys recommend for load-balancing (the same destination served through different trunks): playing with dialplan or using some external app like SER? |
| 22:39.13 | tompaw | the simplest example: 2 GSM gateways serving the same network, 10 channels each |
| 22:39.28 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 22:39.38 | [TK]D-Fender | drfreeze: you see NO packets reaching *? |
| 22:41.06 | *** join/#asterisk bbryant (n=brett@216.207.245.1) |
| 22:45.01 | drfreeze | [TK]D-Fender: packets are reaching * |
| 22:45.51 | [TK]D-Fender | drfreeze: Then you'd better be pastebin-ing them like I asked |
| 22:45.59 | MaliutaLap | then a sip debug should debug the registration process |
| 22:48.05 | *** join/#asterisk InfiL00p (n=bowserj@S0106000c418bd9f7.vc.shawcable.net) |
| 22:49.59 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 22:52.17 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 22:52.58 | *** join/#asterisk XVampireX (i=HydraIRC@unaffiliated/xvampirex) |
| 22:53.01 | XVampireX | Hi |
| 22:53.21 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:54.14 | XVampireX | I'm looking for serious people who are interested in working on a very serious project which already has the fundamentals. I need a person who is fluid with Asterisk and C++ if there's such a person please contact me. |
| 22:55.29 | XVampireX | Although Israel should be the preferred location of those people |
| 22:56.44 | *** join/#asterisk macros73 (n=cs@c-67-186-22-161.hsd1.pa.comcast.net) |
| 22:57.08 | drfreeze | ok guys, got it working |
| 22:57.10 | tompaw | XVampireX: what's the project about? |
| 22:58.11 | XVampireX | Internet Radio |
| 22:58.28 | jaytee | yawn |
| 22:59.10 | XVampireX | It's local but it should be better than other stations (Not just music wise) |
| 23:01.06 | tompaw | cool. |
| 23:01.25 | XVampireX | Can't tell you more than that in public. |
| 23:01.33 | XVampireX | But I'm working with someone from the muni here... |
| 23:01.58 | tompaw | Good for you! (I have no idea what muni is ;-) |
| 23:02.11 | XVampireX | municipality |
| 23:05.01 | *** join/#asterisk jplank (n=GBove@ool-18bb0206.dyn.optonline.net) |
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| 23:13.10 | *** join/#asterisk kusznir_ (n=kusznir@75-93-169-215.lew.clearwire-dns.net) |
| 23:13.43 | kusznir_ | Hi all: I'm trying to get asterisk running on an old box (that I can't upgrade right now). I've got it compiled, but when I try and run it, I get, "Illegal Instruction". |
| 23:14.18 | kusznir_ | When I run it in strace, all that it really says is "SIGILL (Illegal instruction) @ 0 (0) --- |
| 23:14.28 | kusznir_ | Line above that is: geteuid32() = 0 |
| 23:14.58 | kusznir_ | Any idea what would cause illegal instruction (or, I guess, what comprises an illegal instruction, and potential causes in asterisk)? |
| 23:15.44 | kusznir_ | btw, this is asterisk 1.6.0-beta2, compiled on this machine (fedora core 6 -- yea, old..can't fix it right now) |
| 23:17.43 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 23:21.40 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 23:23.40 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 23:27.59 | silvertip257 | [TK]D-Fender: What might I check/do since outbound calls do not connect? |
| 23:28.04 | *** join/#asterisk devhen|Work (n=devhen@216.194.118.110) |
| 23:34.04 | drfreeze | Oh, almost done |
| 23:34.17 | drfreeze | Just need to get this call pickup working |
| 23:34.29 | [TK]D-Fender | silvertip257: Last we saw, Ekiga was issuing a cancel |
| 23:34.37 | drfreeze | I have SIP phones in callgroup1 and pickupgroup 1 |
| 23:34.56 | silvertip257 | [TK]D-Fender: i let it run - i will create another pastebin for review |
| 23:35.25 | drfreeze | Anyone know the fine details of what to check for to get call pickup working? |
| 23:39.00 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 23:39.00 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 23:41.13 | drfreeze | Sip debug is telling me the followng: |
| 23:41.14 | drfreeze | Looking for * in ard (domain 192.168.121.50) |
| 23:41.38 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 23:41.38 | *** mode/#asterisk [+o file] by ChanServ |
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| 23:53.27 | silvertip257 | [TK]D-Fender: here's the newest debug - thanks >> http://pastebin.com/d45fa9ff8 |
| 23:53.53 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 23:55.25 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 23:55.29 | teknoprep | how are aastra phone ? |
| 23:55.36 | teknoprep | are they kick ass ? or do they have problems ? |
| 23:55.41 | teknoprep | like voice quality in particular |
| 23:58.03 | tompaw | thinks that * + openser is exactly what he needs. |