| 00:00.45 | mvanbaak | and you get the same support from junghanns.net, _IF_ you can get in touch with them |
| 00:00.51 | mackes | For $2000 a pci card, I guess its the least they could do. |
| 00:00.59 | mvanbaak | the last part is actually the main issue with junghanns |
| 00:01.12 | mackes | So, Sangoma gear is just as good? |
| 00:01.18 | mackes | Same Zap setup? |
| 00:01.21 | mvanbaak | I love sangoma gear |
| 00:01.26 | mackes | No differences? |
| 00:01.42 | mvanbaak | well, they use wanpipe |
| 00:01.47 | mvanbaak | so you have to install that |
| 00:01.59 | mvanbaak | but the setup is good, and it will patch zaptel for you |
| 00:02.07 | mvanbaak | I never had an issue with it |
| 00:02.17 | mackes | Hmmmm |
| 00:02.18 | mvanbaak | just run: ./Setup.sh install |
| 00:02.19 | mackes | Neat |
| 00:02.22 | mvanbaak | and there you go |
| 00:02.27 | Qwell | extra layer to break, not open source |
| 00:02.28 | Qwell | meh |
| 00:02.33 | mvanbaak | choose for 'TDM Only (asterisk)' |
| 00:02.35 | mackes | I run Poundkey (AsteriskNow) without the GUI |
| 00:02.39 | mvanbaak | lol Qwell, you're biased |
| 00:02.44 | [TK]D-Fender | mvanbaak, just did that this afternoon... for my S518 of course ;) |
| 00:02.45 | Qwell | am I wrong? |
| 00:02.53 | mackes | I love it, because it is super lean |
| 00:02.58 | mackes | and Very stable |
| 00:03.07 | mvanbaak | Qwell: wanpipe never breaks |
| 00:03.07 | mackes | And Digium supports it |
| 00:03.14 | mvanbaak | that is of course my opinion |
| 00:03.30 | mvanbaak | so nowhere near the truth |
| 00:04.03 | mvanbaak | it's the same as me telling chan_skinny is ready for production use if you can live with limited device support |
| 00:04.04 | [TK]D-Fender | mvanbaak, I'll through in my support to "so rare, I rarely care" |
| 00:04.08 | mackes | Wanpipe might be great. I havent played with it |
| 00:04.51 | mvanbaak | [TK]D-Fender: I never had a single issue with wanpipe |
| 00:05.09 | mvanbaak | It Just Works (tm) |
| 00:05.11 | [TK]D-Fender | mvanbaak, I had a minor one at the start about 2 years ago... long forgotten now ;) |
| 00:05.58 | mvanbaak | once we had a card that wouldn't put the line up. we called sangoma, they logged in 2 minutes later and the next day we had a new card |
| 00:06.05 | mackes | Does WanPipe use the same Zap config files- same setup? |
| 00:06.45 | mvanbaak | mackes: as soon as you installed wanpipe and zaptel it's just a PRI zaptel card like any other |
| 00:06.57 | mackes | That is cool |
| 00:07.17 | mackes | I know they are less expensive, is the hardware as good? |
| 00:07.21 | mvanbaak | mackes: but if you are going to order a new card now, get a digium card |
| 00:07.33 | mackes | why is that? |
| 00:07.33 | mvanbaak | no, they are the same price as digium cards |
| 00:07.57 | mvanbaak | mackes: because with buying digium cards you support the asterisk development |
| 00:08.10 | mvanbaak | and we all need money to go into that direction dont we ? |
| 00:08.21 | mvanbaak | <--- in no way affiliated or employed by digium |
| 00:08.32 | mackes | Ahhh. That is true, and I do want to support Digium |
| 00:08.47 | mackes | They are making a large impact on my career. |
| 00:09.07 | mvanbaak | mackes: the older digium cards have issues. but I hear only possitive stuff about the latest cards they make |
| 00:09.19 | mackes | I recieved a 19% jump in salary, 2 years in a row because of Asterisk |
| 00:09.28 | mvanbaak | gheh |
| 00:09.40 | mvanbaak | last year I made a 49% jump in salary |
| 00:09.47 | mvanbaak | up that is |
| 00:09.51 | mackes | There you go. |
| 00:09.57 | mackes | We all owe them |
| 00:10.06 | mvanbaak | yup, we do |
| 00:10.12 | mackes | So, I would pay alittle more for their cards. |
| 00:10.17 | mvanbaak | that's why I try to create patches for them |
| 00:10.23 | mackes | Its my companies money anyways |
| 00:10.26 | [TK]D-Fender | only got this lousy t-shirt.... |
| 00:10.30 | mackes | heheh |
| 00:10.32 | mvanbaak | because 99% or our asterisk stuff is pure voip |
| 00:10.53 | mvanbaak | actually, we hate pri and bri and analog |
| 00:10.56 | mackes | We are planing to bring them in for a onsite bootcamp to train our IT department |
| 00:11.06 | mackes | 12,000 for 6 people |
| 00:11.57 | mvanbaak | reason nr uno: it's not supported on anything besides linux |
| 00:13.02 | mvanbaak | and that's why I still buy stuff from sangoma. |
| 00:13.06 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-29-244.lns10.syd7.internode.on.net) |
| 00:13.09 | mvanbaak | they support OpenBSD |
| 00:13.23 | [TK]D-Fender | mvanbaak, You could always go to gateways... |
| 00:13.27 | mackes | Have any of you gone to VON or Astericon, or Bootcamp? |
| 00:13.34 | mvanbaak | [TK]D-Fender: that's what we do now |
| 00:13.47 | mackes | What is the most worthwhile trip? |
| 00:14.03 | mvanbaak | no idea |
| 00:14.13 | mvanbaak | I wanted to go to astricon, but cant afford it |
| 00:15.56 | dlynes | mackes: so cheap? |
| 00:16.18 | mackes | I'm Sorry? |
| 00:16.27 | dlynes | mackes: it's only $12000 for 6 people? |
| 00:16.47 | mackes | Yep. That was the Quote |
| 00:17.01 | mackes | Do you work for Digium Training? |
| 00:17.15 | dlynes | mackes: no...I was being facetious :) |
| 00:17.15 | mackes | Maybe I should be talking about numbers online |
| 00:17.19 | mackes | Ohhh |
| 00:17.27 | dlynes | mackes: $12000 I thought was pretty expensive :) |
| 00:17.30 | mackes | Yeah $2000 a person |
| 00:17.54 | mackes | The normal price is $3000 a person- Plus flights, Plus Room/Food |
| 00:17.58 | dlynes | mackes: but then again...I just blew $30K on my real estate education, too |
| 00:18.27 | dlynes | so that's a lot cheaper |
| 00:19.14 | mackes | Yep. And I hope that if I have a digium trainer in my hometown just for us for a full 5 days, I can pick his brain after hours |
| 00:19.54 | mackes | I would like our company to become a full Asterisk/ Digium poster child. |
| 00:20.01 | mvanbaak | I should go to Amsterdam tomorrow and bug kpfleming |
| 00:20.02 | mvanbaak | ;) |
| 00:20.32 | mackes | I don't think I would accomplish anything in Amsterdam |
| 00:20.42 | mvanbaak | lol |
| 00:20.49 | mvanbaak | I live in .nl all my life |
| 00:21.02 | mvanbaak | amsterdam is just another village in .nl |
| 00:21.05 | mackes | Really? |
| 00:21.19 | mackes | Do you smoke pot? |
| 00:21.42 | mvanbaak | only when I feel like it |
| 00:21.49 | mvanbaak | ;) |
| 00:21.56 | mackes | Its no "legal" in the US, so we cannot smoke it in public |
| 00:22.00 | Qwell | how often do you not feel about it? :p |
| 00:22.05 | Qwell | s/about/like/ |
| 00:22.09 | mvanbaak | Qwell: once a year |
| 00:22.13 | mackes | ahha |
| 00:22.19 | Qwell | once a year you don't feel like it? :p |
| 00:22.30 | mvanbaak | erm, yeah, something like that |
| 00:22.34 | Qwell | heh |
| 00:22.42 | mvanbaak | I can count the 'no thanks' on one hand |
| 00:22.45 | mvanbaak | and I'm 29 |
| 00:23.14 | mackes | I dont think I can "justsay no" if it were ready and waiting |
| 00:23.18 | mvanbaak | around the age of 12 you get in contact with it |
| 00:23.31 | dlynes | mvanbaak: pretty much the same here |
| 00:23.40 | dlynes | mvanbaak: Canada's pretty open when it comes to pot, too |
| 00:23.40 | mvanbaak | so that's like 4 to 5 times 'no thanks' in 17 years |
| 00:24.06 | mvanbaak | pot is one thing |
| 00:24.11 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-191-90.telkomadsl.co.za) |
| 00:24.15 | mvanbaak | a lot of stuff is available here |
| 00:24.25 | dlynes | mvanbaak: yeah...hashish is pretty open here, too |
| 00:24.32 | dlynes | mvanbaak: but you can get all kinds of stuff |
| 00:24.36 | xacatecas | V-chris, just so you know - i found the code in asterisk causing that ... issue. it's intentional. |
| 00:24.41 | mvanbaak | mushrooms, speed, benzedrine, amphetamine, coke, heroine, lsd etc etc |
| 00:24.53 | mackes | Wooooah |
| 00:25.04 | dlynes | mvanbaak: so you mean .nl is driving the world market in drugs? |
| 00:25.12 | mackes | That is a party of a sentence |
| 00:25.25 | mvanbaak | dlynes: it's actually 30% of our countries moneyflow |
| 00:25.42 | dlynes | mvanbaak: heh...my province's #1 export is marijuana :) |
| 00:25.43 | mvanbaak | and that's the part they know for sure |
| 00:26.00 | mvanbaak | dlynes: our countries nr1 export is xtacy |
| 00:26.00 | dlynes | mvanbaak: BC pot is pretty much internationally known |
| 00:26.02 | xacatecas | if you're interested, look at lines 5113 -> 5125 of channels/chan_iax2.c :$ |
| 00:26.21 | dlynes | mvanbaak: yeah...a lot of that crap is imported to Canada |
| 00:26.38 | mvanbaak | dlynes: we ship it, because it's crap |
| 00:26.43 | mvanbaak | we dont use it in .nl |
| 00:26.51 | mvanbaak | it's not used a lot here |
| 00:27.00 | mvanbaak | we prefer speed and cocaine |
| 00:27.03 | xacatecas | ok, is it possible to set the calling_pres value from asterisk? |
| 00:27.08 | xacatecas | well, from the dialplan? |
| 00:27.11 | [TK]D-Fender | dlynes, I would have thought cheap movies for the US market :) |
| 00:27.15 | dlynes | mvanbaak: yeah...and there's so many different things added to it that shouldn't even be in it, that peeps die like flies from it |
| 00:27.31 | mvanbaak | dlynes: indeed |
| 00:27.32 | dlynes | [TK]D-Fender: no idea...I mostly watch Hong Kongese movies |
| 00:28.02 | dlynes | [TK]D-Fender: Steven Chow and Bo Bo Chan and Jackie Chan....they're all pretty good |
| 00:28.02 | mvanbaak | but canabis and friends are 100% legal here in .nl |
| 00:28.11 | mvanbaak | they call it 'soft-drugs' |
| 00:28.26 | dlynes | mvanbaak: they're pretty much legal in BC....just the government hasn't publicly come out and said it is :) |
| 00:28.35 | mvanbaak | the heavy stuff is in a grey situation |
| 00:28.42 | dlynes | mvanbaak: We've got the Amsterdam Cafe and friends |
| 00:28.47 | dlynes | mvanbaak: Pot smoking cafe's |
| 00:29.13 | mvanbaak | but even here in the woods I know over 5 ppl who sell cocaine and stuff |
| 00:29.28 | mvanbaak | and not even that expensive |
| 00:29.57 | mvanbaak | if you want to go totally wasted for a weekend (24/7) it will cost you like 200 us dollar (120 euro) |
| 00:30.09 | mvanbaak | that means 4 gram of cocaine |
| 00:30.43 | mvanbaak | that's the far end |
| 00:31.10 | mvanbaak | in between you have GHB, speed, xtacy, benzedrine, amphetamine, mushrooms etc |
| 00:31.33 | mvanbaak | hell, you can even buy toxic frogs in the smartshop |
| 00:31.47 | mvanbaak | you lick them to get high on their poison |
| 00:31.54 | mvanbaak | that I never tried |
| 00:32.25 | mvanbaak | I dont like frogs, and the idea to scrape the stuff of them with your tongue is too much for me |
| 00:33.39 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 00:38.26 | *** join/#asterisk dlynes (n=chatzill@d206-116-189-12.bchsia.telus.net) |
| 00:58.36 | jameswf-home | heh http://video.stumbleupon.com/#p=6uo8sdq7w3 |
| 01:00.26 | *** join/#asterisk dlynes (n=chatzill@d206-116-189-12.bchsia.telus.net) |
| 01:06.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:09.16 | xacatecas | hi, would the patch at http://www.kroon.co.za/downloads/cid-chan_iax2.patch make sense? The idea is to always set the cid information if it's available from the user setup, but to still leave presentation alone. However, the behaviour is now different if at some point setcallerpres(allowed_passed_screen) from the dialplan. |
| 01:10.02 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 01:13.09 | xacatecas | without the patch above asterisk ignores the callerid= info in a user's setup (for iax at least) unless the caller provides at least some form of CID. |
| 01:14.10 | xacatecas | ok, well, need sleep, bye |
| 01:19.56 | *** join/#asterisk km2 (n=x@adsl-76-252-245-25.dsl.pltn13.sbcglobal.net) |
| 01:23.00 | Qwell | ~dos2unix |
| 01:23.00 | jbot | dos2unix is, like, in the sysutils package, or at http://www.bsd.org/dos2unix.html, or perl -pi -e 's/\r\n/\n/g' <filename>, or recode, or fromdos, or tr -d \\\r, or on some non-gnu systems would also (unexpectedly) convert the charset |
| 01:23.10 | *** join/#asterisk isamar (i=1000@201.12.126.242) |
| 01:23.17 | isamar | hi folks |
| 01:23.25 | isamar | anybody playing with mfcr2? |
| 01:26.33 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 01:27.55 | coppice | we don't play with MFC/R2. we take it seriously :-) |
| 01:30.00 | [TK]D-Fender | coppice, Have you worked with hylafax & IAXmodem before? |
| 01:30.15 | coppice | nope. never |
| 01:31.51 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 01:32.18 | [TK]D-Fender | hrm. was looking for input from anyone who might have run those 2 on 1 server but registers to another on local LAN (not localhost). They say it should still be almost as reliable as local if controlled. |
| 01:32.57 | rob0 | I did some hylafax work, but on a telco DID |
| 01:33.19 | dlynes | [TK]D-Fender: I've found iaxmodem is not terribly reliable, in general, even on the same machine |
| 01:33.31 | dlynes | [TK]D-Fender: I've used it on two different machines as well, on the same LAN |
| 01:33.33 | rob0 | and several years back, don't remember much |
| 01:33.47 | [TK]D-Fender | rob0, I have 2 boxes, 1 with the PRI on it, and the OTHER is the one I want to put iaxmodem & hylafax on (* box is a little sensitive at this point) |
| 01:34.50 | [TK]D-Fender | my hylafax box won't have * on it at all. |
| 01:35.27 | [TK]D-Fender | Gbit connection between the two. |
| 01:35.36 | coppice | [TK]D-Fender: iaxmodem tends to work best on the sam machine as the asterisk providing the IAX feed. Even on a LAN packet loss tends to be an issue between machines. If you get more than 1 or 2% call failures from valid calls (i.e. ones which have the potential to success) something is wrong |
| 01:36.29 | [TK]D-Fender | coppice, guess I'm going to find out one way or another. |
| 01:39.41 | coppice | people make assumptions about packet loss on LANs without trying it. you can find serious problems in a lot of installations, even when the LAN is lightly loaded |
| 01:44.18 | [TK]D-Fender | coppice, yeah I realize this might not work out and have a few contingency plans. Just looking for some input on plan-A |
| 01:46.05 | coppice | when iaxmodem and hylafax are set up right, the failure rates are very low. low enough that we sometimes go through them by hand on high volume installations to find the cause |
| 01:48.58 | [TK]D-Fender | I would prefer to do this right on my * box, but its running FC3 (ancient) and I'm waiting to port it to new hardware and an OS refresh. Right now I'm jsut a littel scraed to actually touch it. |
| 01:49.06 | [TK]D-Fender | little scared* |
| 01:50.45 | [TK]D-Fender | And for the purpose I have in mind "infrequent mass-fax", I can afford 1-2% failures as long as they retransmit ok. |
| 01:51.08 | [TK]D-Fender | to the tune of max 1500 faxes per batch once a month |
| 01:51.33 | [TK]D-Fender | even that number is high. |
| 01:54.18 | overra | ~book |
| 01:54.19 | 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 |
| 01:54.22 | overra | thanks |
| 01:54.22 | jbot | overra: de nada |
| 01:54.27 | coppice | there are iaxmodem setups running multiple E1s all day |
| 01:56.24 | coppice | apps inside * seem to have timing issues, so iaxmodem seems to give more predictable results than app_rxfax and app_txfax - especially app_txfax which works great for some installations, and very poorly for others. |
| 01:58.42 | *** join/#asterisk isamar (i=1000@201.12.126.242) |
| 01:58.55 | [TK]D-Fender | coppice, as an * app goes, rxfax is just fine for smaller installations because you already can separate based on how it came in and thats jsut minor scripting. Dealing with sending faxes with txfax is a whole other game. I much prefer the though of something like hylafax to manage that part. |
| 01:59.20 | [TK]D-Fender | rxfax already works fairly decently on that box. |
| 02:00.18 | coppice | yeah, but some people get great results with rxfax, while txfax on the same box fails most of the time |
| 02:00.38 | [TK]D-Fender | coppice, interesting. |
| 02:00.54 | [TK]D-Fender | coppice, then again, thats exactly what I don't want to do anyways :) |
| 02:01.30 | coppice | the T.30 engine in hylafax used to be more battle hardened than the one in spandsp, but they shouldn't be much different these days |
| 02:01.57 | isamar | coppice... just need a quick tip to go home " |
| 02:02.03 | isamar | :-) |
| 02:02.35 | isamar | in mfcr2.. protocol error, cause 32722... in BR... |
| 02:02.40 | isamar | how can I workaround that? |
| 02:03.39 | *** join/#asterisk uluatu (n=deg@189.58.6.78.adsl.gvt.net.br) |
| 02:03.57 | [TK]D-Fender | isamar, stop using MFCR2 ;) |
| 02:05.02 | isamar | [TK]D-Fender only if I leave Brazil or we explode Telmex :-) |
| 02:08.25 | coppice | MFC/R2 is the future |
| 02:09.30 | *** join/#asterisk implicit (n=implicit@ip68-105-68-39.sd.sd.cox.net) |
| 02:13.43 | *** join/#asterisk jumpie (n=jumpie@pool-96-231-155-171.washdc.fios.verizon.net) |
| 02:13.45 | jumpie | sup guys |
| 02:13.46 | [TK]D-Fender | coppice, actually... he has it now, so that guarantees that at a bare minimum, its the "present" ;) I might even venture a guess to say "past" |
| 02:13.50 | jumpie | hey hey fender |
| 02:14.35 | TrentCreek | yeah |
| 02:14.39 | coppice | [TK]D-Fender: but its not working. when its debugged it will be the future |
| 02:14.51 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 02:15.13 | drmessano | Until the future gets here |
| 02:15.17 | drmessano | Then it will be something else |
| 02:15.46 | drmessano | So long as it's not based on the OpenSSL library in Debian |
| 02:16.40 | coppice | there wil still be bugs, with or without that one |
| 02:16.55 | drmessano | But |
| 02:17.15 | drmessano | My cat just generated 100 keys and took down a nasa.gov website.. |
| 02:17.34 | coppice | and there wil be lemon soaked paper towels |
| 02:18.03 | [TK]D-Fender | "When will then be now?" "Soon!" |
| 02:18.10 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 02:18.23 | drmessano | WD Doc |
| 02:19.19 | drmessano | I just love how the most complicated apps and libraries fall to the dumbest of bugs |
| 02:19.54 | drmessano | Like, forgetting to uncomment # makeapp_fast(); |
| 02:20.45 | drmessano | 125% performance increase when we removed loop(); from non-looped subroutines! |
| 02:20.51 | coppice | "Your code does something broken" |
| 02:20.53 | coppice | "Oh yeah, that's code I specifically put in there during debug to test the error recovery on another machine" |
| 02:20.55 | coppice | ....been there. done that. |
| 02:21.21 | drmessano | I do have full confidence the asterisk team |
| 02:21.51 | drmessano | In asterisk 2.0, i'll be able to run 100 concurrent transcoded IAX3 calls on a 486 |
| 02:22.25 | drmessano | Except the source tarball will be 3GB in size |
| 02:24.18 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:26.44 | coppice | the live CD.... that is endlessly added to as you are downloading it |
| 02:27.38 | drmessano | lol |
| 02:28.01 | [TK]D-Fender | coppice, Nope... Live BD <- |
| 02:29.23 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:32.18 | coppice | I saw some HD-DVD players in a shop on Friday, still with a high price tag on them. weird :-\ |
| 02:32.40 | isamar | will build a statue to coppice.. |
| 02:35.58 | Nugget | has an hd-dvd player. |
| 02:36.23 | *** join/#asterisk secgod (n=secg0d@c-67-184-227-156.hsd1.il.comcast.net) |
| 02:37.27 | coppice | I've never seen an HD-DVD disc in a video shop |
| 02:37.59 | Nugget | I was really hoping hd-dvd would "win" over bluray, so I figured it was the least I could do. I bought a player and a half dozen discs or so. |
| 02:38.11 | Nugget | it got a lot of use via netflix (still does, actually) |
| 02:38.38 | Nugget | when someone comes out with a bluray player that doesn't suck I'll transition, though. it's over for hd-dvd |
| 02:39.01 | *** join/#asterisk Insectoid (n=noone@adsl-072-148-075-054.sip.gnv.bellsouth.net) |
| 02:39.18 | jaytee | is it possible to dial a number over a PRI channel and once it answers use SendDTMF to send it digits? |
| 02:41.02 | overra | can anyone suggest any DID trunk providers? |
| 02:42.11 | Insectoid | I'm looking to integrate the IBM tts as a synth on my new asterisk setup. I have a program I can call to generate a wave file containing spoken text with /usr/bin/ttsynth_say -8/-16/-22 -fwavefile.wav text. According to what I've read I want to make a macro? that I can pass text and it will generate and play the wave file. Can someone explain how I might do this? |
| 02:43.10 | Insectoid | overra: I use voip.ms -- they're relatively inexpensive. |
| 02:43.56 | [TK]D-Fender | jaytee, "D()" in Dial. |
| 02:44.36 | jaytee | [TK]D-Fender, thanks. can I use w as a wait in the Dial? |
| 02:46.58 | *** join/#asterisk AJayMN (n=me@99-203-204-164.area2.spcsdns.net) |
| 02:47.24 | AJayMN | How can i get asterisk to add a 1+ to the callerid of uncoming calls? (before asterisk does anything else with that caller) |
| 02:47.55 | *** part/#asterisk danp (i=danp@elmer.glueless.net) |
| 02:48.27 | rob0 | core show function CALLERID , core show application set |
| 02:50.22 | AJayMN | thanks rob0 |
| 02:50.22 | jbot | no worries, AJayMN |
| 02:50.41 | drmessano | lol |
| 02:50.47 | drmessano | thanks somename |
| 02:50.47 | jbot | gern geschehen, drmessano |
| 02:50.55 | drmessano | Bot is broken |
| 02:51.01 | drmessano | thanks jbot |
| 02:51.01 | jbot | pas de quoi, drmessano |
| 02:51.01 | AJayMN | what the ... |
| 02:51.10 | drmessano | thanks [TK]D-Fender |
| 02:51.10 | jbot | bitte, drmessano |
| 02:51.19 | drmessano | Credit taking SOB |
| 02:51.22 | drmessano | BAD KARMA |
| 02:51.28 | AJayMN | lol |
| 02:52.44 | drmessano | Hi jbot |
| 02:52.58 | drmessano | jbot, punch rob0 |
| 02:52.59 | jbot | ACTION lets fly with a wild haymaker which catches rob0 right on the nose |
| 02:53.10 | drmessano | jbot, clownkill rob0 |
| 02:53.14 | drmessano | Bah |
| 02:54.19 | drmessano | clownkill is where a clown pulls you inside a stall in a public restroom and chops you up with a machete.. Guess jbot doesn't know that one |
| 02:55.36 | AJayMN | wonders why drmessano knows this... was he a clown in the past years/ |
| 02:55.38 | AJayMN | ? |
| 02:55.47 | drmessano | Want some candy? |
| 02:56.32 | drmessano | >:o) |
| 02:56.38 | AJayMN | thats what i said to a kid at the mall today ;) lol |
| 02:59.54 | jaytee | [TK]D-Fender, dude that did the trick. I've now got * calls routing to Nortel Callpilot if there's no answer after a set timeout. |
| 03:00.13 | [TK]D-Fender | EW |
| 03:00.19 | [TK]D-Fender | Norhell |
| 03:00.31 | [TK]D-Fender | Call-Pilot = bleh |
| 03:00.49 | jaytee | I know, it's ugly but within a year to a year and a half it will be gone and all that will remain is Asterisk. |
| 03:01.04 | [TK]D-Fender | jaytee, Why so long? |
| 03:01.20 | jaytee | [TK]D-Fender, cuz we is po! |
| 03:01.34 | [TK]D-Fender | Ok, out for a while, back later. |
| 03:01.48 | jaytee | [TK]D-Fender, we are a non-profit with a CFO that squeezes the dollar till the eagle shits. |
| 03:02.19 | jaytee | [TK]D-Fender, later and thanks again |
| 03:03.05 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:03.09 | isamar | coppice.. still there? |
| 03:03.36 | drmessano | jaytee: Cant sell him on the virtues of trixbox? |
| 03:03.39 | jameswf-home | oh the excitement 80 RPMS |
| 03:05.00 | *** join/#asterisk gardo (n=gardo@121.97.178.31) |
| 03:07.24 | *** join/#asterisk Tinman_gz (n=noone@66.193.134.74) |
| 03:07.47 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:08.08 | jameswf-home | the first rule of trixbox is.... there is no trixbox |
| 03:08.44 | drmessano | Correction, there is no FreePBX |
| 03:08.50 | drmessano | Ignore the man behind the curtain |
| 03:08.53 | coppice | are they like 78RPMs speeded up slightly? |
| 03:08.58 | Tinman_gz | help... i am having CURL get the perameters of Dial when curls gets anything with a "|" in it, it only gets the characters up to that point... i have tried \| but still no luck... i can put in other chrs like : or something but i dont see in asterisk a way to do character substitution ne ideas? |
| 03:09.22 | jameswf-home | tried "" |
| 03:09.43 | Tinman_gz | "" where? |
| 03:10.31 | Tinman_gz | "s are literal in asterisk |
| 03:10.39 | Tinman_gz | (arent they?) |
| 03:11.45 | Tinman_gz | try it this way SET(blah=SIP/whatever|30) then Dial(${blah}) i wont get the timeout |
| 03:11.55 | Tinman_gz | s/i/it |
| 03:12.37 | Tinman_gz | of course "show application awk" doesnt work P-) |
| 03:15.27 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 03:16.13 | Tinman_gz | kinds quiet in here |
| 03:16.23 | Tinman_gz | kinda even... |
| 03:16.42 | Insectoid | Tinman_gz |
| 03:16.46 | Insectoid | Try my question |
| 03:16.52 | Tinman_gz | i just joined |
| 03:17.00 | Insectoid | I'm looking to integrate the IBM tts as a synth on my new asterisk setup. I have a program I can call to generate a wave file containing spoken text with /usr/bin/ttsynth_say -8/-16/-22 -fwavefile.wav text. According to what I've read I want to make a macro? that I can pass text and it will generate and play the wave file. Can someone explain how I might do this? |
| 03:18.04 | Tinman_gz | well i suppose you could like system the text to ttsynth_say ... then wait(1) or something then Playback that file |
| 03:18.15 | Tinman_gz | i dont see the point in a macro... |
| 03:18.33 | Tinman_gz | is ttsynth_say fast? |
| 03:18.35 | Insectoid | Let's pretend I'm a total asterisk clueless schmuck who installed it from portage. |
| 03:18.42 | Tinman_gz | lol |
| 03:18.42 | Insectoid | Pretty fast yeah |
| 03:18.48 | Tinman_gz | ok type asterisk -r |
| 03:18.57 | Insectoid | I'm using freepbx on top of it. |
| 03:19.01 | Insectoid | Don't shoot me |
| 03:19.07 | Tinman_gz | show application System |
| 03:19.10 | Insectoid | stand by |
| 03:19.31 | Tinman_gz | then |
| 03:19.33 | Tinman_gz | show application Playback |
| 03:19.57 | jaytee | Insectoid, you're using freepbx on top of Asterisk? |
| 03:20.08 | Tinman_gz | is tts tons better than festival? |
| 03:20.22 | Insectoid | Oodles |
| 03:20.26 | Insectoid | It's IBM TTS |
| 03:20.37 | Tinman_gz | i suppose its not opensource then... |
| 03:20.40 | Insectoid | No but |
| 03:20.46 | Insectoid | I can walk you through installation if you like. |
| 03:20.53 | coppice | is that the IBM TTS based on the old Eloquence code? |
| 03:20.55 | Insectoid | It is |
| 03:21.02 | Tinman_gz | emerge tts? nto that easy i bet |
| 03:21.06 | Insectoid | Bit difficult to get it installed and you need some emulation libraries but it works. |
| 03:21.15 | Insectoid | Oh no way near that easy :P |
| 03:21.28 | coppice | I thought that was abandoned long ago |
| 03:21.52 | Insectoid | see ttsynth.com |
| 03:21.58 | Insectoid | Although that's not the one I'm using |
| 03:22.07 | jaytee | Insectoid, did you install freepbx on Asterisk 1.4? |
| 03:22.07 | Insectoid | Trying to remember which one I'm using. |
| 03:22.30 | Insectoid | jaytee: I emerged asterisk from portage and got latest freepbx from svn and installed. |
| 03:22.41 | Insectoid | Asterisk 1.2.27 |
| 03:23.13 | drmessano | I run 1.4 SVN with that-app-we-cant-talk-about-in-here |
| 03:23.25 | drmessano | That reminds me, I need to update |
| 03:23.33 | jaytee | Insectoid, thanks. I'm wanting to add a gui to Asterisk for some functions but don't want to hose my existing dialplan. |
| 03:23.50 | drmessano | jaytee: Not gonna happen with FreePBX |
| 03:24.08 | Insectoid | For anyone who's interested in cheaper ttsynth: http://voxin.oralux.net/ |
| 03:24.12 | jaytee | drmessano, any suggestions? |
| 03:24.29 | drmessano | jaytee: Either you want to commit yourself to a GUI, or you dont |
| 03:24.40 | drmessano | jaytee: Theres no going back, in other words |
| 03:25.19 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 03:25.19 | Insectoid | jaytee: Back up your dialplans and just reenter them once freepbx is installed. |
| 03:25.33 | drmessano | It doesn't work that way |
| 03:25.38 | Insectoid | Of course not |
| 03:25.59 | drmessano | So why did you suggest it, if you know it doesn't work? |
| 03:26.09 | mackes | What App is that? |
| 03:26.10 | Insectoid | I didn't know it didn't work. It was an assumption that it work. |
| 03:26.19 | jaytee | I'm thinking I'd have to install freepbx and then recreate the dialplan using the GUI. |
| 03:26.24 | Insectoid | That's what I suggested |
| 03:26.26 | drmessano | jaytee: yes |
| 03:26.31 | Insectoid | But apparently that *doesn't* work |
| 03:26.38 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 03:27.08 | drmessano | You can *RECREATE* your box with the GUI but you *CAN NOT* reuse your existing dialplan |
| 03:27.14 | drmessano | Christ |
| 03:27.33 | Insectoid | Why I said reenter -- but irrelevant now. |
| 03:27.38 | coppice | I always liked Eloquence. Its not very natural, but beats anything else for clarity. The Chinese one sucks though. I worked with them to try to improve it, but they seemed to have other things on their mind at the time.... like being taken over |
| 03:28.15 | Insectoid | coppice: I'm actually a blind computer who uses Eloquence on a daily basis (I'm using it now to read this) and I thought it'd be brilliant if I could hook it to Asterisk. |
| 03:28.21 | Insectoid | s/computer/computer user |
| 03:28.29 | jaytee | I spent most of my day getting cdr working with mysql |
| 03:29.20 | coppice | Insectoid: well, if you were chinese, you probably wouldn't use their Mandarin version :-) |
| 03:29.50 | Insectoid | coppice: Probably not. I must say the American English is quite good and understandable up to insane speeds though |
| 03:29.52 | Tinman_gz | oh its only $40 not that bad... |
| 03:30.01 | Insectoid | Tinman_gz: $5 for the other one. |
| 03:30.52 | Tinman_gz | not to be a butthead but any idea on my chr substitution deal? if i could just echo ${blah} | awk 's/:/\|/g' i would be in business |
| 03:31.36 | Tinman_gz | it needs to fit into Exec(${appname}(${appperamsw})) |
| 03:33.37 | Insectoid | Should I just code up a quick daemon to fake festival and pass it on to ttsynth_say? |
| 03:33.40 | *** join/#asterisk javb (n=javb@190.94.2.77) |
| 03:34.05 | Tinman_gz | could probly do that in bash... |
| 03:34.28 | Insectoid | I was going to use Python |
| 03:34.47 | Tinman_gz | might be worth a try... i try to keep my dialplans very simple... |
| 03:35.04 | Tinman_gz | it seems asterisk cant even do chr substitution in a variable !! |
| 03:36.35 | Tinman_gz | i would rather not CUT/CUT/CUT then ${blah}|${bleech}|${floom} but ... |
| 03:38.59 | Nasra | hello everyone after reading alot about the future of telephony by O'reilly I think gonna need some help in walk me to instal Asterisk on my box ....CentOs 5.1 |
| 03:40.00 | Tinman_gz | if you have gentoo type "emerge asterisk" |
| 03:40.35 | overra | Nasra; me too :P |
| 03:40.47 | Nasra | so we are same boat |
| 03:41.05 | Nasra | Tinman_gz: are you talking to me? |
| 03:41.11 | overra | well, i'm on debian and installation is pretty easy |
| 03:41.17 | overra | just need to know what to configure |
| 03:41.36 | Nasra | what did you do from scratch? |
| 03:42.22 | overra | in debian it was easy as apt-get install asterisk |
| 03:42.48 | Nasra | where did you put it ? |
| 03:44.48 | overra | i dont know how to install on centos |
| 03:46.00 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 03:46.13 | Tinman_gz | Nasra yea i was... |
| 03:46.44 | jaytee | Nasra, here's a good how-to for Asterisk on CentOS 5.1, I've used it for 3 servers without problems http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 03:47.07 | Tinman_gz | Nasra when it installs (from apt-get or portage) just goto /etc/asterisk and take a look at extensions.conf (all all the other stuff in there) |
| 03:47.12 | Nasra | jaytee: just follow intructions....right/ |
| 03:47.30 | Tinman_gz | most of the stock config files are documented inline |
| 03:47.36 | Nasra | ok |
| 03:48.35 | Nasra | Tinman_gz: I am just ready to start this thing alltogether.... |
| 03:48.49 | Tinman_gz | Nasra when the service is running type "asterisk -r" at the command line inside there its all tab completion lots of help "show /t/t" "show application /t/t" etc... |
| 03:48.50 | overra | same here |
| 03:49.07 | Nasra | ok |
| 03:49.22 | Nasra | I like you Tinman_gz |
| 03:50.00 | Nasra | Tinman_gz: I also have SPA3102.... |
| 03:50.16 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 03:50.26 | Nasra | I am in the process of getting a polycom ip phone |
| 03:52.06 | Tinman_gz | thx... softphones are easy for testing i have a granstream 4 line too that is just for testing its good to have a dev and a production box (its easy to dick up your dialplan) |
| 03:52.21 | jaytee | amen! |
| 03:52.37 | Tinman_gz | extensions.conf is picky |
| 03:52.53 | Nasra | Tinman_gz: I was told they not that great to opt for polycom |
| 03:53.26 | isamar | says "thank you" to coppice |
| 03:53.38 | Tinman_gz | Nasra huh? u mean grandstream vs polycom? |
| 03:53.46 | Nasra | yes sir |
| 03:54.19 | Tinman_gz | i have a bunch of stuff linksys poly a couple iaxys granstream bunch of softphones... its all pretty much the same to me |
| 03:54.44 | Nasra | everybody says diff things then.... |
| 03:55.13 | Tinman_gz | grandstream phones feel chincy and light but most of this stuff has a steel weight in the bottom to make it feel more "substantail" as long as it works i am good |
| 03:55.46 | Nasra | Tinman_gz: I went to telephonydepot.com and there are a whole bunch in there....(good prices).... |
| 03:55.49 | Tinman_gz | man that doesnt look right... substantial |
| 03:56.02 | coppice | I hate phones that impress people by having a chunk of steel in the handset |
| 03:56.18 | denon | "substantial" is right |
| 03:56.20 | Tinman_gz | coppice that and remote controls... |
| 03:56.39 | denon | don't go trying to spell it with an "sh" or somethin ;) |
| 03:56.45 | jaytee | I've chosen Polycom for the sound quality where it counts and Grandstream for the places where it doesn't. |
| 03:56.47 | Tinman_gz | i try to type fast... i tyr ot tpye fsat |
| 03:57.55 | Nasra | Tinman_gz: thanks alot |
| 03:58.02 | Nasra | time to dig it |
| 03:58.48 | Tinman_gz | Nasra keep it simple first my dial plans have gotten so ugly that i have given up and switching everything to php and perl |
| 03:59.00 | mackes | The Polycom 320 is so inexpensive that I'm not sure the Grandstream as a reason to exist |
| 03:59.29 | Nasra | Tinman_gz: since I am gonna be learning that is how gonna be " simple"...one more thing. |
| 03:59.45 | Nasra | do you use a good voip provider? |
| 03:59.50 | JT | Tinman_gz: using fast agi i hope :) |
| 04:00.59 | Nasra | mackes is that good to use for a newbie? |
| 04:01.24 | coppice | mackes: the 320 may be reasonable, but a grandstream is still half the price |
| 04:01.25 | mackes | Oh sure thing. It has a straight forward Web Interface |
| 04:01.32 | Tinman_gz | JT no just a local apache server... what is fast agi? |
| 04:01.38 | mackes | Hmmm Which one? |
| 04:01.56 | JT | Tinman_gz: oh, how do your dialplans work? curl? |
| 04:02.03 | Tinman_gz | JT yea |
| 04:02.04 | mackes | A Polycom 320 is $80 |
| 04:02.26 | mackes | And is an excellent phone. |
| 04:02.32 | JT | Tinman_gz: ah fair enough |
| 04:02.35 | Nasra | okay |
| 04:02.49 | JT | AGI can bog down when you have a few calls going |
| 04:02.53 | Tinman_gz | JT donno if fast is important i had a redundant loop that probly went 300 times in a second or two |
| 04:02.57 | mackes | I have had one of all of the grandtream products at they are all only "ok", sometimes |
| 04:02.58 | coppice | the BT102 is <$40 |
| 04:03.09 | mackes | Oh, The BT102 |
| 04:03.12 | mackes | That is true |
| 04:03.16 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.30 | JT | i'd rather an analogue phone and ATA than a BT102 |
| 04:03.31 | Tinman_gz | JT a php session makes tracking very very very easy |
| 04:03.46 | mackes | I dont mean to knock those who like grandstream. They are neat to play with and Modify |
| 04:04.01 | JT | heh, php, no comment :) |
| 04:04.03 | mackes | However, everyday use-- not so good. |
| 04:04.24 | Tinman_gz | my grandstream gxp2000 is sweet |
| 04:04.34 | Tinman_gz | sounds good lots of buttons |
| 04:04.47 | JT | compare to a polycom, sounds good? o_O |
| 04:05.12 | Tinman_gz | JT maybe my ear isnt that sensitive but yea... |
| 04:05.31 | *** join/#asterisk unstuck (n=gareth@202-21-136-49.cable4.acsdata.co.nz) |
| 04:05.31 | mackes | I guess it is a matter of taste. But, before anyone buys more then one Grandstream, please look at a Polycom 320 |
| 04:05.37 | drmessano | JT: GSM is GSM ;) |
| 04:05.41 | JT | you must've found a good firmware, make sure you have a copy of it before you try changing firmware |
| 04:06.15 | Tinman_gz | i took it out of the box and plugged it in... using iLBC |
| 04:06.43 | mackes | I have used grandstreams to modify for overhead paging. They do that very well. |
| 04:06.52 | mackes | Also for remote ringers. |
| 04:07.16 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 04:07.52 | Tinman_gz | we dont do much sip mostly all PTSN stuff forwarding... play messages (for realators) |
| 04:08.03 | Tinman_gz | sip is handy for dev though |
| 04:08.40 | Tinman_gz | oh but we do need a good T1 card... any suggestions? |
| 04:09.31 | *** join/#asterisk styelz (n=yoohoo@2001:388:f000:0:0:0:0:20b) |
| 04:09.50 | mackes | If you are installing Asterisk to replace a Nortel, Alcatel, or another professional telephone system, installing a nice handset is everything. Normal users dont care about what is in the back office (AKA Asterisk) They care about the phone on their desk. Its worth spending a few extra bucks for a decent, even if basic phone set. |
| 04:10.17 | JT | Tinman_gz: make sure you get a card with hardware echo cancellation |
| 04:11.20 | mackes | One mistake many people make is to buy a SIP phone with two many lines (registration points). A one "line" phone is almost aways fine, because Asterisk will handle routing multiple extensions to that one registration |
| 04:11.27 | jaytee | anyone running 1.6 beta? |
| 04:12.15 | Tinman_gz | mackes is sweet for dev though |
| 04:13.09 | mackes | grandstreams? For one on your desk to test with- Yep, they were the first phones I had... and they did the job. |
| 04:13.10 | jameswf-home | nope it has 0 downloads |
| 04:13.36 | Tinman_gz | my "extension" is my cell phone... |
| 04:13.46 | mackes | Just know that better, and easier is available for just a few dollars more |
| 04:13.50 | drmessano | Sure you want to use Asterisk 1.6.0 or Fonality FonCoresterisk 1.6.0? |
| 04:13.59 | jameswf-home | we use a ipod touch as an extension |
| 04:14.49 | jaytee | drmessano, FonCoreasterisk 1.6.0? |
| 04:14.55 | Tinman_gz | wierd... dial(blah|5) i didnt pick up and it goes to the next priority not to exten => t,1 |
| 04:15.08 | outtolunc | you got a mic for the itouch? |
| 04:16.13 | jameswf-home | the boss set his up i is too po for a regular ipod muchless a iphone jr |
| 04:16.33 | Tinman_gz | it took 8 more seconds to hit exten => t,1 am i doing something wrong |
| 04:16.35 | Tinman_gz | ? |
| 04:17.09 | outtolunc | ah there is a doc conn mic mod |
| 04:17.19 | outtolunc | er dock |
| 04:19.34 | jameswf-home | someone else doing it http://www.youtube.com/watch?v=oeOOdnm6vRs |
| 04:19.49 | jameswf-home | vid looks like crap |
| 04:22.54 | outtolunc | http://www.youtube.com/watch?v=GmUPRrrxoVc&NR=1 |
| 04:38.44 | mackes | Its Funny how chat slows as it gets later |
| 04:39.08 | mackes | However we are all over the world- so its a different time everywhere |
| 04:42.51 | mackes | mackes |
| 04:44.32 | mackes | ~book |
| 04:44.33 | 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 |
| 04:49.44 | jameswf-home | anyone play with adobe air... looks neat |
| 04:53.49 | mackes | ~mackes |
| 04:53.50 | jbot | i heard mackes is the man |
| 04:57.33 | mackes | Thanks jbot |
| 04:57.33 | jbot | bitte, mackes |
| 05:00.47 | *** join/#asterisk CCFL_Man2 (i=1aa54744@pool-71-241-86-162.scr.east.verizon.net) |
| 05:10.16 | mackes | ~noob |
| 05:11.36 | mackes | ~noob |
| 05:11.48 | mackes | What is a noob |
| 05:22.31 | Tinman_gz | yea... |
| 05:39.48 | jameswf-home | ~newbjuice |
| 05:39.49 | jbot | DUDE your spilling your red newb juice on the white carpet and we just had it cleaned... |
| 05:47.22 | V-chris | jbot can't spell ;( |
| 05:47.59 | V-chris | ~include => spell-checker |
| 05:56.36 | CCFL_Man2 | can cisco voice gateways get incoming sip calls on a port other than 5060? |
| 06:15.55 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:31.11 | *** join/#asterisk philip76 (n=philip@202.21.177.3) |
| 06:33.00 | TrentCreek | ~burn mackes |
| 06:33.00 | jbot | ACTION pours gasoline all over mackes, ignites the fire, and then enjoys some toasty marshmallows with the glorious blaze |
| 06:34.38 | jameswf-home | :)) http://www.break.com/index/new-level-of-crazy.html |
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| 07:14.26 | dkwiebe4 | Good Day, is asterisk 1.6 stable yet? I have a customer who very badly wants the "callback on failed blind transfer..." |
| 07:14.45 | drmessano | It's still beta.. |
| 07:14.52 | drmessano | Beta = beta |
| 07:15.25 | dkwiebe4 | sounds good. |
| 07:15.51 | dkwiebe4 | Beta sometimes = documentation not done yet |
| 07:16.00 | dkwiebe4 | I'll just bump them off. I can do most of it in the dialplan anyways |
| 07:24.53 | drmessano | Beta is never supposed to mean "documentation not done yet" |
| 07:24.58 | drmessano | Beta is beta |
| 07:37.13 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 07:40.32 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 07:42.04 | jameswf-home | wtf do people use alpha/beta software and complain that it seems incomplete in some fasion? |
| 07:44.32 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-187-7.hsd1.wa.comcast.net) |
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| 08:09.11 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:15.01 | mandh | Hi all , i use asterisk in call center the sound volume of inicoming calls from PSTN is weak but the other side hear very well how to fix this problem please |
| 08:17.18 | *** join/#asterisk redback (n=noname@mail.datadream.co.uk) |
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| 10:47.32 | SparFux | Does anybody use linux 2.6.25 with misdn? |
| 10:52.35 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 11:00.46 | coppice | i just found problems trying to use 2.6.25 with sangoma. it seems some names have changed in the kernel |
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| 11:32.41 | phix | smiley, smile |
| 11:37.23 | *** join/#asterisk ipkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 11:38.35 | ipkaf | hi |
| 11:38.35 | jbot | hello |
| 11:38.43 | ipkaf | hi jbot |
| 11:39.09 | matnel | :D |
| 11:39.57 | ipkaf | i got router asus wl500g premium |
| 11:40.21 | ipkaf | Ram 32 MB |
| 11:40.36 | ipkaf | 8MB |
| 11:40.43 | ipkaf | flash |
| 11:41.23 | ipkaf | frequency 240 mhz |
| 11:41.52 | ipkaf | these are the technical details of my router |
| 11:42.23 | ipkaf | how many simultanoues can do with this router ??? |
| 11:42.35 | ipkaf | how many simultanoues call can do with this router ??? |
| 11:44.00 | JT | to run asterisk on the router? |
| 11:44.19 | ipkaf | yes |
| 11:44.48 | ipkaf | don't worry about installing asterisk on my router |
| 11:45.03 | ipkaf | my question is how many simultanoues call can do ??? |
| 11:45.20 | ipkaf | if i install install asterisk on my router ? |
| 11:45.28 | ipkaf | how many ??, |
| 11:45.40 | ipkaf | can anyone give me an idea |
| 11:45.53 | smps | ipkaf, it depends how are you going to use it |
| 11:46.18 | smps | ipkaf, if it has to proccess many voide coding/decoding (ie. from one codec into another and vice versa) |
| 11:46.30 | smps | ipkaf, s/voide/voice/ |
| 11:46.57 | smps | ipkaf, then not much since those routers are not so powerfull like normal 240MHz intel cpu |
| 11:47.32 | ipkaf | i just want to remote my pstn line over the net |
| 11:47.36 | JT | ??????? ??? ? |
| 11:47.51 | ipkaf | for this purpose is it got idea |
| 11:47.56 | ipkaf | ??? |
| 11:48.04 | ipkaf | for this purpose is it good idea |
| 11:48.50 | smps | ipkaf, how are you planing to do that ? |
| 11:49.08 | ipkaf | ??? |
| 11:49.24 | smps | pstn is isdn, analog line or ? |
| 11:49.49 | ipkaf | yes analog line |
| 11:50.26 | smps | ipkaf, so you are planning to bus one ATA with FXO port , connect FXO <-> PSTN and from ATA to asterisk ? |
| 11:50.39 | ipkaf | yes |
| 11:50.39 | smps | ipkaf, s/bus/buy/ |
| 11:50.41 | ipkaf | correct |
| 11:50.49 | ipkaf | i got i allready ata |
| 11:50.53 | ipkaf | etc.. |
| 11:50.56 | smps | ipkaf, ok, it should work :) |
| 11:51.16 | tzafrir | ipkaf, you won't be able to run too many concurrent cals with Asterisk. CPU might not be the only problem. Memory might also be a problem |
| 11:51.28 | ipkaf | thx a lot |
| 11:51.35 | ipkaf | just only one its enough |
| 11:51.50 | ipkaf | i just want to remote my pstn line over net |
| 11:51.53 | ipkaf | that 's all |
| 11:52.21 | ipkaf | is it ok ??? |
| 11:52.24 | JT | ipkaf: enough with the question marks. |
| 11:52.41 | tzafrir | Asterisk has "channels". Each channel is basically half a call (a "leg" of a call) |
| 11:52.45 | smps | ipkaf, yes it is ok, go set it up |
| 11:52.51 | ipkaf | ok thx a lot |
| 11:53.00 | tzafrir | Each such channel is independent and uses a bit of memory |
| 11:53.05 | ipkaf | dream become true becauze of asterisk |
| 11:53.07 | ipkaf | bye to all |
| 11:53.39 | JT | too much speed or meth |
| 11:53.41 | JT | heh |
| 11:53.48 | rob0 | wipes a tear |
| 11:54.02 | rob0 | oh wait |
| 11:54.13 | rob0 | that's because I don't have enough coffee |
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| 12:12.07 | marl | hi folks, anyone tell me how stable 1.6 is at the moment? |
| 12:13.14 | file | that answer will vary depending on what 1.6 is in use for for the person |
| 12:13.41 | file | for example I run 1.6 in my apartment setup with 10 or so phone calls a day and it is perfectly happy but my usage patterns probably don't equal your usage patterns |
| 12:14.43 | marl | have 10 iax trunks, and its used for call recording/monitoring/answer machine, and iax routing, no phone hardware |
| 12:15.37 | file | the only way to know for sure is to put it into a test setup with your specific applications in use and test it |
| 12:18.51 | marl | ok, will give it a try |
| 12:23.13 | marl | is there a link someone could give me that gives the differances between the asterisk versions? 1.4.14 - 1.4.20 and 1.6 ? |
| 12:24.34 | file | there is a file in the source directory called... CHANGES |
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| 12:36.22 | marl | thanks |
| 12:36.22 | jbot | de rien, marl |
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| 13:49.20 | Shotygun | Hello. Using Asterisk 1.2.x, I was wondering how can I make a usage of queues strategy=leastrecent while still skipping to the next member in case the first member does not answer. Right now it keeps insisting on the same member over and over.. |
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| 14:25.30 | *** join/#asterisk atomspace (n=huh@adsl-226-14-149.bhm.bellsouth.net) |
| 14:25.43 | atomspace | can you shutdown/poweroff from asterisk cli? |
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| 14:56.16 | marl | ok, have install asterisk 1.6, there is no phone hardware on this server (am using iax only) do i need to install zaptel? |
| 14:59.21 | atomspace | I do not believe so. As you're hinting at, I believe the zaptel stuff is only if you have cards - digium/otherwise - installed |
| 15:00.31 | RypPn | nope, if you have no zaptel hw, you prolly still want ztdummy for timing |
| 15:00.34 | *** join/#asterisk Zuchmir (n=dddddd@ool-18bd3bfc.dyn.optonline.net) |
| 15:01.18 | Zuchmir | are there any ATAs that run * that can be modified? |
| 15:01.24 | marl | ok, any idea what would cause iax to not connect? i am using my iax.conf file from 1.4.4 (which worked fine on 1.4.4) and * doesnt seem to register to my iax provider, if i run iax2 show peers. it lists all the iax stuff but says that they are all unmonitored and iax2 show channels, show nothing |
| 15:07.20 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:07.20 | SparFux | Does anybody use linux 2.6.25 with misdn? |
| 15:07.55 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:08.13 | riddlebox | Zuchmir, not sure if I understand your question, ATA's dont run asterisk |
| 15:10.29 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
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| 15:13.48 | marl | ok, iax is working, 4 some reason it just took a while to register with my iax provider |
| 15:14.49 | atomspace | http://pastebin.ca/1021923 |
| 15:14.51 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 15:14.56 | atomspace | line 116 is the problem, I think |
| 15:15.32 | atomspace | "lacks" extension - but I have it setup as an extension (801). Any thoughts? |
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| 15:20.20 | Zuchmir | riddlebox: my guess is that linksys, as well as other ATAs *DO* run linux |
| 15:20.55 | Zuchmir | and my guess is, if it's running linux it's running * and/or *can* run * |
| 15:22.17 | riddlebox | Zuchmir, well you would have to be able access the firmware somehow |
| 15:23.10 | atomspace | is there a webgui for the device --- http://ip-of-device |
| 15:23.12 | atomspace | ? |
| 15:23.38 | marl | ok, now im tottaly confused :( i had 1.4.4 forking fine with the followme app, a wwek ago, (posibly after a power outage) the followme app, no longer accepts the called party pressing 1 or 2 to accept/reject the call, i thought this may have been a corupted file or something, re ran the mae install for 1.4.4 and it didnt fix the problem, so unintalled * and installed 1.6, the followme app still wont accept 1 or 2 for the accept/reject :( anyone any ideas? |
| 15:24.09 | marl | this is killing me, as i use the followme app for all my incoming calls :( |
| 15:26.09 | JT | i think the SPA4000 or something runs asterisk |
| 15:27.07 | atomspace | marl: have any extensions changed since it last was working? |
| 15:27.43 | marl | no |
| 15:28.26 | marl | the call comes intot he called persons phone, as usuall, it is just that * wont accept press 1 or 2, it just keeps repeating the press 1 to accept..... message until time out |
| 15:29.00 | marl | afk |
| 15:29.22 | atomspace | marl: are you sure that you have 2 set as an option within asterisk? |
| 15:42.15 | tzafrir | Is there any way to set the default language in Asterisk without changing some 22 different places in the config directory? |
| 15:51.07 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 15:58.57 | *** join/#asterisk DaneM (n=dane@adsl-76-236-27-148.dsl.chi2ca.sbcglobal.net) |
| 16:02.15 | DaneM | Hello, everybody. I'm having a problem with a Motorola Wildcard X100P card and zaptel drivers. When I modprobe the modules (wctdm, wcfxo, wct1xxp, zaptel), it only shows the following in /dev/zap: "1 channel ctl pseudo timer", but does not show the FXO port, which (as far as I can tell) should be "2". I've tried switching PCI slots, and disabling ACPI at boot (with acpi=off on the kernel line), to no avail. Any help would be |
| 16:06.59 | DaneM | Has anybody run into such a problem before? |
| 16:07.02 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:07.02 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:11.00 | Strom_C | DaneM: first off, you don't need wctdm or wct1xxp |
| 16:11.07 | riddlebox | man the forums are getting alot of spam these days |
| 16:11.22 | Strom_C | DaneM: second, have you configured the card correctly in zaptel.conf? |
| 16:11.31 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 16:11.48 | mackes | Hey Everyone |
| 16:12.19 | DaneM | Strom_C: thanks for the reply. I'll rmmod those and see what happens. I'll get the contents of zaptel.conf for you. One min. |
| 16:12.33 | Strom_C | ~pb |
| 16:12.34 | 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 |
| 16:12.44 | Strom_C | DaneM: rmmod isnt going to help you |
| 16:12.51 | DaneM | what should I do? |
| 16:12.51 | Strom_C | it's just not necessary to load those drivers |
| 16:13.01 | DaneM | Strom_C: ok |
| 16:14.38 | Zuchmir | jt: i assume u meant SPA400 |
| 16:15.01 | DaneM | Strom_C: In zaptel.conf I have the following (lines separated by ";") fxsks=1 ; fxoks=2 ; loadzone = us ; defaultzone = us |
| 16:15.33 | Strom_C | what happens when you run ztcfg? |
| 16:15.51 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:17.43 | *** join/#asterisk atomspace (n=huh@adsl-242-0-24.bhm.bellsouth.net) |
| 16:17.47 | atomspace | outgoing calls just hangup ("click") after about 20 seconds. http://pastebin.ca/1021962 |
| 16:18.30 | atomspace | is line 117 the problem? |
| 16:18.44 | *** join/#asterisk kash (n=nibblet@24.182.112.112) |
| 16:19.16 | kash | hi, i'm having errors with les.net termination |
| 16:19.16 | kash | i get 503 SERVICE UNAVAILABLE with outgoing; incoming works fine. |
| 16:19.24 | DaneM | Strom_C: www.pastebin.com/m2369314b (I think I typed that right) |
| 16:19.25 | Strom_C | atomspace: jesus, where did this overgrown macro come from? |
| 16:19.36 | Strom_C | DaneM: don't you have copy and paste? :P |
| 16:19.39 | DaneM | (I'm using a different computer for chatting) |
| 16:19.54 | DaneM | kvm switch :-p |
| 16:19.55 | Strom_C | DaneM: ssh, my friend |
| 16:20.02 | DaneM | heh ok...I'll do that. |
| 16:20.54 | Strom_C | DaneM: |
| 16:20.54 | Strom_C | "Changing signalling on channel 1 from Unused to FXS Kewlstart" <-- this is a good thing |
| 16:20.56 | atomspace | Strom: I'm not sure. |
| 16:20.56 | Strom_C | it means it worked |
| 16:21.12 | DaneM | Strom_C: that's good :-) |
| 16:21.14 | Strom_C | atomspace: you're not running freepbx/trixbox, are you? |
| 16:21.16 | atomspace | Strom: I'm using freepbx, if that may help answer |
| 16:21.18 | atomspace | yes |
| 16:21.20 | Strom_C | sigh |
| 16:21.24 | Strom_C | please read the channel topic |
| 16:21.46 | Strom_C | ~freepbx |
| 16:21.47 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:21.55 | atomspace | so you're saying that this error is specific to freepbx then? freepbx issue? |
| 16:22.03 | atomspace | my mistake, I thought this was an asterisk problem |
| 16:22.19 | DaneM | Strom_C: OK...ssh'd in :-) |
| 16:23.18 | DaneM | Strom_C: any ideas on why channel 2 is having problems? I tested the card on another machine only a day earlier and it worked. :-( |
| 16:23.31 | Strom_C | DaneM: "x100p" is only a one-port card |
| 16:24.30 | DaneM | Hmmm...OK. I has 2 ports in the back, which I don't understand. *scratches head* Anyway, I need FXO to work, not FXS. Any ideas on that? |
| 16:24.33 | rob0 | haha, but it has two rj-45 jacks, it being up to the lucky winner to figure out which one to use. |
| 16:24.54 | Strom_C | DaneM: the one port is a passthrough |
| 16:24.54 | rob0 | My idea would be to get a magnifying glass and look at them. |
| 16:24.55 | atomspace | rj45 or rj11? |
| 16:25.02 | rob0 | 11 yes sorry |
| 16:25.10 | *** part/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 16:25.12 | *** join/#asterisk kash (n=nibblet@24.182.112.112) |
| 16:25.12 | atomspace | np, just making sure I knew what was goin on |
| 16:25.20 | kash | sorry. |
| 16:25.28 | kash | did you see my question about les.net |
| 16:25.35 | DaneM | Strom_C: ok. Makes sense. Can I just change the line in zaptel.conf to make it fxoks then? |
| 16:25.37 | atomspace | so is this not a x100, but a tdm400 type card? |
| 16:25.59 | DaneM | atomspace: no, it's a x100p. only 2 rj-11 ports in the back. |
| 16:26.04 | Strom_C | DaneM: it's a ONE PORT CARD |
| 16:26.07 | atomspace | well, if its a fxs card, you cannot set it to work as fxo, you'll have to buy a fxo "daughter" card |
| 16:26.12 | atomspace | oh |
| 16:26.15 | rob0 | x100p has an "in" jack and a "thru" jack. |
| 16:26.26 | atomspace | OH, I see now |
| 16:26.31 | DaneM | Strom_C: ok. thanks. |
| 16:26.44 | atomspace | well, one port should say "line" and the other say "phone" or something to that magnitude, yes? |
| 16:26.52 | DaneM | atomspace: that's correct. |
| 16:27.11 | atomspace | fxo = line, fxs = phone |
| 16:27.25 | atomspace | ...if that is what you're asking |
| 16:27.50 | DaneM | atomspace: that's what I thought, but for some reason it only shows the fxs port in /dev/zap :-( |
| 16:28.09 | DaneM | (it only lists a "1", but no "2", like it did in the other computer I tried it in. |
| 16:28.12 | DaneM | ) |
| 16:28.27 | DaneM | ...but like Strom_C said, that could be the hardware's fault. |
| 16:28.49 | atomspace | cli - "zap show channels" ---- is that where it only shows "1" and not "2" ? |
| 16:29.21 | DaneM | atomspace: no, I'll try that. (I haven't gotten around to configuring * yet) It says that in "ls /dev/zap" |
| 16:29.56 | atomspace | oh, well if its not listed in the zap config, then I doubt it will show in 'zap show channels" |
| 16:30.04 | DaneM | ok |
| 16:30.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:31.00 | *** join/#asterisk Zimmy (n=zimmy@atheme/member/zimmy) |
| 16:31.01 | atomspace | hmmm, and once you get into the zap config, i'm no help. When I added my tdm400 card, I just ran one command and it installed the card and everything for me. Something like "zap-genconfig" or something like that ---- Strom, or anyone else, does that ring a bell? |
| 16:31.31 | DaneM | atomspace: perhaps genzaptelconf? |
| 16:31.35 | Strom_C | beats me; I always just configure it by hand |
| 16:31.39 | rob0 | yes, genzaptelconf |
| 16:31.49 | DaneM | just ran it...checking now... |
| 16:32.33 | rob0 | You could poke around /proc/zaptel and do the same thing |
| 16:33.56 | DaneM | rob0: I just changed fxsks=1 to fxoks=1 in zaptel.conf and ran a "cat /proc/zaptel/1. Here's what I got: Span 1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) ; 1 WCFXO/0/0 |
| 16:34.07 | DaneM | (lines separated by a ;) |
| 16:34.11 | DaneM | ";" |
| 16:37.42 | *** join/#asterisk kash (n=nibblet@24.182.112.112) |
| 16:37.54 | kash | does anyone else here use les.net? |
| 16:38.10 | DaneM | "zap show channels" reveals the following: Chan Extension/pseudo Context/default MOH Interpret/default |
| 16:38.42 | DaneM | ...but I don't know if I have * configured properly for zapata yet. (there is a section in zapata-channels.conf, though) |
| 16:41.11 | *** join/#asterisk DarnoQ (n=d@chello089076192243.chello.pl) |
| 16:41.52 | DarnoQ | anyone tried using g729 via iaxlite ? |
| 16:42.20 | DarnoQ | am i being legal using it ? |
| 16:50.58 | Qwell | are you transcoding to/from g729, or is it just passthrough? |
| 16:51.10 | atomspace | I have a tdm400 card with an fxo port. For me to use that fxo port as a trunk to the pstn, must that fxo port have it's own extension? |
| 16:53.03 | marl | atomspace,sorry had to dive out there :( |
| 16:55.48 | marl | ok, have removed my followme.conf file, installed the default 1.6 file and added one section, still wont accept my options 1 or 2 |
| 16:56.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:57.00 | DaneM | that's odd...suddenly "zap show channels" is giving me a "no such command 'zap show'" error. :-( |
| 16:57.44 | DarnoQ | it |
| 16:58.08 | DarnoQ | it's only passthrough on asterisk but i assume iaxlite encodes somehow to send it |
| 16:58.48 | marl | have even tried changing the accept button to 9, but still not working :( |
| 17:00.45 | rob0 | atom, no, but it would be used in Dial() commands for contexts in your dialplan which are allowed to dial out on it. |
| 17:02.14 | atomspace | ok, so are you saying that in the "Dial()" dialplans the extension is interchangeable with some other reference to the zap channel like "Zapg0" or "Zap1" or something like that? |
| 17:02.21 | atomspace | sorry, that was to rob0 |
| 17:03.55 | drmessano | atomspace: are you using FreePBX? |
| 17:04.14 | rob0 | um, "core show application dial" and some of the examples in the extensions.conf sample might help. |
| 17:04.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:04.31 | atomspace | Drmessano: yes I am |
| 17:04.56 | drmessano | atomspace: Then you need to be asking in #freepbx and keeping the setup in that context, as it's going to be different |
| 17:05.43 | atomspace | I'm asking regarding asterisk in general, not freepbx |
| 17:06.25 | atomspace | ...dialplans in general |
| 17:07.09 | drmessano | I am not gonna argue with you, but FreePBX _IS_ your dialplan, and how a card is addressed within FreePBX is different |
| 17:07.25 | rob0 | Sure, but you can break your freepbx if you aren't careful. |
| 17:08.41 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:09.36 | atomspace | what I'm trying to do is learn about asterisk by itself, so I can move away from freepbx. I'm not doing anything to any textfiles directly in my freepbx setup. I'm learning, thats all. |
| 17:09.49 | atomspace | ...mom |
| 17:10.46 | rob0 | It might be better to build it up from scratch. |
| 17:11.53 | atomspace | dumb question, but with asterisk (not freepbx/asterisknow/any other asterisk+gui setup), you only deal with text files, correct? just typing stuff to set everything up, no clicking? |
| 17:12.52 | russellb | correct |
| 17:15.24 | mackes | atomspace: And it is the best way to go |
| 17:15.55 | mackes | All of the GUI versions look good at first, but they are a huge pain in the ass over time |
| 17:17.29 | atomspace | mackes: because they use resources? or why do you say that? |
| 17:18.09 | mackes | Raw Asterisk is almost limitless in the number of VoIP things it can do. |
| 17:18.24 | mackes | The Gui version are limited by the interface |
| 17:18.34 | errr | thats not totally true with freepbx |
| 17:19.16 | errr | you can do anything with it asterisk can.. you can use the _custom.conf files to override anything the default dialplan it creates does |
| 17:19.57 | mackes | But then you have a system that is non standard- for the Gui distro, and Non Standard asterisk. |
| 17:20.06 | tzafrir | errr, how can you rename a number with freepbx? |
| 17:20.18 | errr | tzafrir: rename a number? |
| 17:20.28 | tzafrir | that is: change the number assigned to a phone (or user) |
| 17:20.31 | errr | mackes: and what is standard anyway? |
| 17:21.14 | errr | tzafrir: with out using the gui? |
| 17:21.25 | tzafrir | with or without |
| 17:21.55 | tzafrir | your choice |
| 17:21.58 | atomspace | FIGHT! FIGHT! FIGHT! circle around, everyone |
| 17:22.02 | errr | yes with, and with more effort without |
| 17:22.20 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 17:22.23 | tzafrir | with: remove and recreate |
| 17:22.45 | errr | the thing is, anything you change from custom conf files cant be changed from the gui |
| 17:23.06 | tzafrir | And don't forget to write over all the definitions you set there |
| 17:23.34 | errr | tzafrir: I didnt say it would be easy, Im just saying it is possible |
| 17:23.55 | tzafrir | It's also possible to run Asterisk on a turing machine |
| 17:23.57 | mackes | It is all a matter of opinion. I do know that freepbx didn't include all of the options i required. Switchvox is closer, however I do not want a database involved in my configuration. |
| 17:24.34 | mackes | I love flat files, that I can easily backup and transfer in case of a failure. |
| 17:24.41 | tzafrir | Swichvox is closed, indeed |
| 17:25.07 | tzafrir | ;-) |
| 17:25.23 | atomspace | speaking of backing up stuff - what do you guys use to backup asterisk? |
| 17:25.46 | tzafrir | What is "asterisk"? Configuration? spare system? |
| 17:26.04 | mackes | What do you mean tzafrir |
| 17:26.14 | atomspace | I'd like to just image the whole drive, so in the event of drive failure I can just load that on another drive, but I've never done any sort of imaging on servers |
| 17:26.43 | tzafrir | atomspace, rsync is usually better for a live sync |
| 17:27.14 | mackes | You just need the /etc/asterisk directory and your zap files on /ect to restore an asterisk install in case of failure |
| 17:27.33 | tzafrir | "image" means copying a whole lots of data. To copy this at a partition level the system must be down for quite a while |
| 17:28.09 | tzafrir | Or even if not down (e.g: using mondo rescue) it will do quite some disk IO, and Asterisk may not be able to function correctly |
| 17:28.09 | mackes | I tar these files once an hour and stick them on another disk, and another share on another machine |
| 17:28.31 | mackes | I would not image.... |
| 17:28.57 | mackes | I recommend just backing up your configs. |
| 17:29.07 | errr | if you use lvm on your disks you could take lvm snap shots and dump the snapshots |
| 17:29.08 | mackes | This is another issue with the GUI Asterisk distros |
| 17:29.30 | mackes | It is hard to know just what files you need with the GUI Distros |
| 17:29.43 | mackes | Straight Asterisk- just the text files |
| 17:30.00 | errr | freepbx has a backup module with it that will cron schedule backups and gets all needed files |
| 17:30.14 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 17:30.24 | errr | mackes: you would need the asteriskdb too even for straight asterisk |
| 17:32.35 | tzafrir | errr, do you actually have information worth saving in the astdb? |
| 17:32.45 | errr | tzafrir: yes tons |
| 17:32.59 | tzafrir | something that will not get re-generated? |
| 17:33.11 | tzafrir | There's also tons of junk there, not worth saving |
| 17:33.24 | errr | I guess it depends on what you store in yours |
| 17:34.04 | tzafrir | Well, writing a backup/restore for it isn't difficult |
| 17:34.19 | errr | keeping a copy is trivial too |
| 17:34.22 | mackes | You dont need to keep asteriskdb ? |
| 17:34.25 | mackes | Why? |
| 17:35.14 | *** part/#asterisk Zuchmir (n=dddddd@ool-18bd3bfc.dyn.optonline.net) |
| 17:35.33 | mackes | Just the text files. Use the same version of the Asterisk- pop the files in the directories, set your machine for the same IP address as the old, and start it up |
| 17:36.21 | mackes | The only other things you might want to backup is your Voicemail directory, and your sounds directory if you created custom prompts. |
| 17:36.45 | mackes | I also have my TFTP server running on my Asterisk system- so I back up that directory as well |
| 17:38.00 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:38.16 | mackes | Dont get me wrong. I think the Gui Distros are very cool- Great for home use systems, or small offices.... 25 extensions lets say. |
| 17:38.29 | mackes | They are very good, and very inexpensive |
| 17:39.32 | mackes | However- If you would like to really install enterprise, large systems- company wide, hundreds of extensions, including all types of users- straight Asterisk is the way to go. |
| 17:39.33 | errr | mackes: I use freepbx at our offices. We have 125 extensions spread over 3 locations and have a central voicemail server all the locations share. It has its uses beyond home and small offices |
| 17:39.58 | drmessano | I don't see why a GUI matters.. |
| 17:40.14 | drmessano | All you're doing is writing config files that asterisk reads |
| 17:40.25 | drmessano | Performance issues? How? |
| 17:40.49 | drmessano | Asterisk slows down based on how its configs are written? Someone file a bug report. |
| 17:41.04 | atomspace | mackes: you said "They are very good, and very inexpensive" ---- please elaborate on how expense comes into the software side |
| 17:41.31 | atomspace | are you referring to support? |
| 17:41.34 | mackes | I'm glad it works for you. I hope it continues to keep your enterprise serviced |
| 17:41.49 | DaneM | (belatedly) thanks for all the help, guys. I figured out that the card I have is actually an FSX card (per Strom_C's suggestion). I have ordered an FXO card. you guys are great! |
| 17:41.49 | mackes | Im sorry? Why is Asterisk inexpensive? |
| 17:42.05 | atomspace | well thats good |
| 17:42.31 | *** part/#asterisk DaneM (n=dane@adsl-76-236-27-148.dsl.chi2ca.sbcglobal.net) |
| 17:42.36 | drmessano | Other than the fact that config files written by some of the Asterisk GUIs are VERY hard to debug, hence the lack of support in here, you can't even begin to tell me the performance is an issue |
| 17:42.37 | atomspace | mackes: I understood that you meant "gui distro's are very good, and very inexpensive" |
| 17:42.47 | *** join/#asterisk Darnoq (n=d@chello089076192243.chello.pl) |
| 17:42.54 | mackes | Bigger Picture. A GUI asterisk systems is much less expensive then installing a Nortel Meridan System |
| 17:43.06 | atomspace | ah, I gotcha |
| 17:43.08 | mackes | That is what I meant. They meet a need |
| 17:43.36 | mackes | We have 28 retail locations, and 2 HQ locations |
| 17:44.01 | mackes | the 28 retail had a mix of Nortel Meridian systems, and Alcatel Hybrid systems |
| 17:44.21 | mackes | They were expensive to maintain- service, and handset replacement |
| 17:44.30 | atomspace | is Nortel Meridan a commercial pbx with a gui? |
| 17:45.47 | mackes | almost none of the commercial PBX systems on the market have GUI inherently. They have expensive Add-ons that write configs at the command line |
| 17:45.58 | atomspace | I see |
| 17:46.16 | mackes | Most PBX systems are alot like Cisco Routers. All CLI |
| 17:46.41 | mackes | Nortel is one of the largest manufactures out there. |
| 17:47.10 | mackes | every doctors office, government building, office building runs different size Nortel Systems |
| 17:47.50 | mackes | Anyways, GUI Asterisk systems are a good replacement for the smaller 25 extension systems. |
| 17:48.40 | atomspace | so is a GUI asterisk system better than straight asterisk for the smaller than 25 extension systems? |
| 17:48.41 | mackes | If you are going to have a call center, receptionists, office workers, help desks, et all on one system- then the Gui system isnt going to do it all (From the Gui) |
| 17:49.03 | drmessano | So they don't do Queues and IVRs? |
| 17:49.15 | drmessano | Ringgroups.. |
| 17:49.17 | atomspace | I have freepbx and use an ivr |
| 17:49.28 | atomspace | oh, didn't realize that was you dr |
| 17:49.58 | mackes | I would always run the straight Asterisk. But, that is because I am comfortable with it. It is faster and more powerful for me. However, if you would like a fast, and stable straight forward PBX, the Gui versions are very good |
| 17:51.21 | mackes | They Do- Ring groups, and IVR. Its all about scale. |
| 17:51.59 | atomspace | well, correct me if I'm wrong, but the gui distro's just make it so one doesn't have to touch raw text files. I am well versed in neither linux, nor straight asterisk, so a gui distro is for someone like me. If I did know linux/asterisk well, then you could just do without the gui? |
| 17:52.05 | drmessano | So theres a limit to the size of the IVRs and ring groups, and queues? |
| 17:52.24 | mackes | You can make straight Asterisk do almost anything- Every extension is a set of rules, that are followed. The gui versions are what the author defined them to be. Thats all |
| 17:52.24 | atomspace | basically, what I'm asking is "does the gui just replace having to edit files directly" ? |
| 17:52.49 | mackes | yes |
| 17:52.57 | mackes | and it sound like what you need. |
| 17:53.02 | atomspace | oh, perfect |
| 17:53.03 | mackes | They are very very good systems |
| 17:53.11 | mackes | I would install one |
| 17:53.27 | mackes | How many users/ extensions will you have? |
| 17:53.28 | atomspace | ....but the underlying "asterisk" is the exact same, right? |
| 17:53.44 | atomspace | 6 extensions |
| 17:53.49 | atomspace | very small business |
| 17:53.51 | mackes | Ohhhh. Yes |
| 17:54.09 | mackes | I am sorry we even debated this |
| 17:54.14 | atomspace | haha |
| 17:54.21 | mackes | Which version are you going to use? |
| 17:55.18 | atomspace | freepbx --- if thats what you're asking. I've got it completely setup, and it was super easy ---- again, this is coming from someone who has only used ubuntu desktop |
| 17:55.27 | atomspace | ... the only linux I've used .... |
| 17:55.27 | mackes | I think "PBX in a Flash" is a very good version. The community at large seems to smile on this distro |
| 17:55.30 | mackes | http://pbxinaflash.net/ |
| 17:55.33 | mackes | No Setup |
| 17:55.44 | mackes | Just install on a blank machine |
| 17:55.46 | atomspace | I'm sorry, yes that what I have |
| 17:56.06 | mackes | a $300 new dell would work nicely |
| 17:56.09 | atomspace | *I've got it completely setup except for outbound calling |
| 17:56.25 | atomspace | I went the route of the ebay special -- $30 dell opteron |
| 17:56.32 | atomspace | woops, optiplex |
| 17:56.32 | mackes | Cool. I think you will be happy. Set it up, and forget it. |
| 17:56.40 | mackes | That should be more then fine |
| 17:57.18 | atomspace | was anyone here at the Atlanta Asterisk user group's "install fest" last month? |
| 17:57.53 | mackes | I'm in Buffalo NY. We have a large Toronto Users Group which is awesome |
| 17:58.07 | *** join/#asterisk ikevin (n=kevin@2001:5c0:99b9:0:20f:b0ff:fe4a:4589) |
| 17:58.28 | atomspace | mackes you aren't affiliated with a voip hardware vendor up there are you? |
| 17:58.43 | atomspace | I've bought a few things from a place in NY - can't remember what city though |
| 17:58.46 | mackes | Who VoIp Supply? |
| 17:58.55 | mackes | They are in Buffalo |
| 17:59.31 | mackes | No, I work for a Car Wash/ Detail Shop/ CStore/ Deli/ Repair Shop chain. |
| 17:59.39 | mackes | www.deltasoniccarwash.com |
| 18:00.14 | atomspace | yep, that is it - voip supply |
| 18:00.38 | atomspace | wow, thats a mouthful |
| 18:00.50 | atomspace | ---the description, not the url |
| 18:01.29 | mackes | Yeah. We are owned by a wealthy private company, that likes to expand- |
| 18:02.45 | mackes | If you would like to do VoIP outbound calling- (Like Vonage) I recommend www.vitelity.com |
| 18:03.20 | mackes | Very inexpensive- VERY, and we routed 60,000 minutes out them last month, no issues |
| 18:03.57 | *** join/#asterisk frieze (n=frieze@pool-71-251-13-242.nycmny.fios.verizon.net) |
| 18:04.26 | frieze | okay so I installed asterisk via apt-get (i.e. the coward's way out). Now how do I add modules (specifically chan_mobile) |
| 18:04.58 | mackes | Sorry frieze, I have never installed that way |
| 18:05.14 | russellb | chan_mobile is only in asterisk 1.6 |
| 18:05.23 | russellb | i'm not sure if there are any 1.4 versions floating around |
| 18:05.33 | mackes | What does chan_mobil do? |
| 18:06.34 | frieze | looks like it's apt-get remove for me |
| 18:06.54 | drmessano | Im using it on 1.4 |
| 18:07.26 | drmessano | I got lazy and used the compiled module from the Trixbox 2.4 distro.. and it's worked VERY well |
| 18:11.37 | frieze | screw it. I'll download and build like my forefathers before me |
| 18:12.25 | atomspace | I think I have some sort of voltage issue: when I try to dial out using a voip phone, the call dials, then is quite for roughly 20 seconds, then the call ends. If I pick up the same line on an analog phone (and just hear dialtone), and THEN dial the number on the voip-phone, the call goes through. So, my question is should I pursue this as a hardware issue, or software issue? |
| 18:14.01 | atomspace | if software, then yes, I will give it a go in the freepbx channel on monday, but I'm not sure if it IS a software issue |
| 18:14.40 | atomspace | *I'm using POTS into a tdm400 card |
| 18:14.50 | Qwell | when running freepbx, always assume software/configuration issue |
| 18:15.48 | atomspace | ok qwell. I'll try my hand there then. thanks |
| 18:18.40 | atomspace | hmmm I think it is just dialing too quickly. I'll try adding a pause before dialing. ....just thinking with my fingers |
| 18:20.24 | atomspace | as far as dial rules in asterisk, is there a "pause" --- like X = 0-9, Z= 1-9, N = 2-9, ??? = "wait one second" |
| 18:21.13 | atomspace | maybe like a "hard pause" or "soft pause"....or is what I'm saying just rediculous |
| 18:23.11 | atomspace | i'm checking the book, sorry |
| 18:23.20 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 18:26.21 | tzafrir | atomspace, you can always bypass freepbx |
| 18:26.43 | tzafrir | For starters: originate SIP/whatever application Echo |
| 18:27.18 | tzafrir | This will skip the huge junk log freepbx generates at dialing |
| 18:27.38 | tzafrir | You also failed to mention the name of that specific voip device |
| 18:27.44 | vee8 | atomspace: Maybe you want to go in FreePBX under trunks put a w in outboud dial prefix |
| 18:28.04 | tzafrir | actually: |
| 18:28.14 | tzafrir | originate SIP/whatever/number application Echo |
| 18:28.40 | atomspace | vee8: I'm in "the future of telephony", under "Dialplan syntax" I don't see "w" --- what is that? |
| 18:30.09 | tzafrir | isn't w zaptel-specific? |
| 18:30.10 | vee8 | it means wait to give the POTS line a chance to get dialtone |
| 18:30.39 | ManxPower | atomspace: "w" will give you a .5 second pause. Use it as part of the dialed number on FXO ports Dial(Zap/1/5551212ww1234) or use it as part of the D() option for all other ports, like Dial(SIP/fff1212@mypeer,,D(ww1234)) |
| 18:30.43 | atomspace | oh jeese, thats exactly what I'm looking for... trying now |
| 18:30.51 | tzafrir | hmm.... oops, why did I get the impression this was sip? |
| 18:31.06 | atomspace | I'm sorry, I may not have been specific enough |
| 18:31.15 | atomspace | tza:^ |
| 18:31.20 | ManxPower | ww gives you a 1 seconds pause, of course. |
| 18:31.22 | tzafrir | you have |
| 18:32.31 | ManxPower | Of course, I CANNOT and WILL NOT help with GUI Asterisks. |
| 18:32.37 | *** join/#asterisk RoyK (n=roy@ti211310a080-6545.bb.online.no) |
| 18:32.37 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:33.53 | Nugget | Yeah, MaxPower isn't a GUI guy. He's her to help people with their grandstream phones. |
| 18:34.57 | frieze | grr |
| 18:35.05 | frieze | anyone up for some build debugging? |
| 18:35.32 | Nugget | just delete lines until it compiles, that's what I always do. |
| 18:36.23 | frieze | Nugget: tempting... |
| 18:36.24 | Nugget | hey, if they don't compile then obviously they're not important. |
| 18:38.56 | frieze | Nugget: we shall see |
| 18:41.25 | frieze | okay, on the other end of the spectrum, anyone else used one of those sata-molex adapters to power a digium card in an sata-only system? |
| 18:42.16 | drmessano | Oh, so you're the one |
| 18:42.54 | Qwell | frieze: yes |
| 18:43.09 | Qwell | there is also the external brick you can get for them |
| 18:43.57 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-126-26-36.bflony.east.verizon.net) |
| 18:44.15 | russellb | Qwell: we really need to just start shipping those with cards ... they're cheap enough |
| 18:44.21 | *** join/#asterisk ManxPower (n=manxpowe@242.sub-75-202-20.myvzw.com) |
| 18:49.26 | *** join/#asterisk pacmanfan (n=Andy@12-218-141-42.client.mchsi.com) |
| 18:50.42 | riddlebox | have you guys tried out oslec yet? |
| 18:53.13 | RoyK | riddlebox: yeah - beats zaptel quite well |
| 18:54.09 | riddlebox | yeah I just got it up and tested, I had a problem with like the first 10-15 seconds of an incoming call I could hear myself talk, but I used oslec and fxotune and no more echo! |
| 18:54.41 | riddlebox | I will have to go out to a customer site next week and install it, they have the same issue |
| 18:54.49 | RoyK | uses A104D cards these days - HWEC.... |
| 18:55.09 | riddlebox | yeah I will have to start doing that |
| 18:55.11 | pacmanfan | i'm puzzled... i can't access asterisk-gui on my server. |
| 18:55.21 | pacmanfan | make checkconfig says my config files look fine. |
| 18:55.35 | pacmanfan | i don't need Apache configured, do i? |
| 18:55.39 | riddlebox | pacmanfan, can you pastebin the link? I think I know what the problem is |
| 18:55.47 | pacmanfan | btw, i'm running Centos 5.1 |
| 18:55.49 | riddlebox | from make checkconfig |
| 18:56.36 | pacmanfan | http://pastebin.com/m81f3d4e |
| 18:56.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:56.59 | *** join/#asterisk aarcane (n=aarcane@ppp-71-128-108-41.dsl.scrm01.pacbell.net) |
| 18:57.20 | riddlebox | pacmanfan, what is the ip address of the server |
| 18:57.47 | aarcane | is asterisk suited for running voicemail and transparent call recording for a home ? |
| 18:57.57 | pacmanfan | 192.168.1.27... and yes, i replaced "asterisk" in the URL with the server ip. |
| 18:58.29 | pacmanfan | i didn't change the default port. |
| 18:58.52 | riddlebox | is this the link you tried http://192.168.1.27:8088/asterisk/static/config/cfgbasic.html |
| 18:58.59 | riddlebox | and can you ping 192.168.1.27 |
| 18:59.05 | pacmanfan | yep and yep |
| 18:59.16 | *** join/#asterisk deltaray (n=deltaray@66.244.79.1) |
| 18:59.17 | pacmanfan | i can log into the asterisk cli |
| 18:59.27 | pacmanfan | asterisk-gui just seems dead on the water |
| 19:00.08 | deltaray | Ok, I'm still trying to get my phone from home connecting to my asterisk server. Its working, but the asterisk server at the office is sending out its internal IP address instead of the external one. Is there a way to configure it to use the external IP? |
| 19:00.22 | riddlebox | pacmanfan, did you get rid of the cache on firefox? |
| 19:01.16 | pacmanfan | yep |
| 19:01.31 | riddlebox | hrmm restart asterisk? |
| 19:02.07 | pacmanfan | oh, wait... now, i AM trying to access asterisk-gui from another system on my network. |
| 19:02.33 | ManxPower | Off topic, please respond via /msg. Does anyone have recommendations for inexpensive hosting services? Virtualized is fine. |
| 19:02.41 | pacmanfan | i haven't tried to access it from localhost... i don't have any browsers on the asterisk server, nor do i have X installed. |
| 19:02.57 | ManxPower | pacmanfan: you might find better help of you went to the correct channel. |
| 19:03.23 | pacmanfan | hmm, good point. i didn't realize asterisk-gui had its own channel. |
| 19:03.30 | ManxPower | deltaray: you mean like externip= option? |
| 19:03.49 | deltaray | ManxPower, does that go in the sip.conf file? |
| 19:04.13 | ManxPower | pacmanfan: we don't use GUIs here, so trying to get support here is like trying to get linux support on a windows channel, or better yet, trying to get Unbuntu support on a Debian channel. |
| 19:04.16 | ManxPower | ~sipnat |
| 19:04.17 | 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 |
| 19:04.38 | ManxPower | deltaray: you need externip= and localnet= on the server side that is behind nat in sip.conf [general] |
| 19:05.59 | atomspace | sorry it took so long for the response --- thank you guys soooo much for your help, specifically vee8 and manx. |
| 19:06.11 | deltaray | ManxPower, but only for the extension that is on the outside, right? MOst of my extensions are on the local network that the asterisk server is on. |
| 19:06.25 | atomspace | adding the "w" fixed my problem. the pbx is completely setup now - thank goodness, we go live at our new offices monday! |
| 19:06.31 | ManxPower | atomspace: fee free to send large amounts of money via paypal to eric@fnords.org |
| 19:06.48 | atomspace | do you accept $3 bills? |
| 19:07.03 | ManxPower | deltaray: no. localnet= lets asterisk know what devices are local and which are remote. |
| 19:07.21 | ManxPower | if the device matches the net/mask of localnet= then it's local, if not then it's remote |
| 19:07.39 | ManxPower | deltaray: go read the sipnat info |
| 19:08.20 | deltaray | ManxPower, ok, thanks for your help. |
| 19:10.09 | ManxPower | atomspace: Your innocence in endearing. Once you go live with the system you should expect massive issues because of stuff you didn't know you should test and because users will start screaming because "its different". |
| 19:10.31 | ManxPower | Oh and because users will be demanding features that they never told you about until now. |
| 19:11.18 | ManxPower | "So, exactly why did you not tell us the most importnat feature you need is for the phone system to call you on your cell phone and read you your new messages!?!?!" |
| 19:11.44 | ManxPower | ^^^ yes, this actually happened |
| 19:14.52 | TrentCreek | ~burn mackes |
| 19:14.53 | jbot | ACTION pours gasoline all over mackes, ignites the fire, and then enjoys some toasty marshmallows with the glorious blaze |
| 19:15.21 | mackes | Ahmmmmm Why?? |
| 19:15.39 | TrentCreek | buhwahahaha |
| 19:17.29 | mackes | ~mackes |
| 19:17.29 | jbot | i heard mackes is the man |
| 19:18.13 | mackes | ~TrentCreek |
| 19:18.14 | jbot | i guess trentcreek is a tool |
| 19:18.52 | russellb | ~thwack mackes |
| 19:18.53 | jbot | ACTION beats mackes on the knee with a Cisco Manual |
| 19:19.14 | mackes | Hmmm |
| 19:19.16 | mackes | ok |
| 19:19.24 | mackes | My poor knee |
| 19:19.27 | russellb | heh |
| 19:19.30 | russellb | random attack! |
| 19:21.23 | aarcane | is asterisk suited for running voicemail and transparent call recording for a home ? |
| 19:22.27 | mackes | Got any thing else TrentCreek? |
| 19:23.49 | russellb | aarcane: yes |
| 19:24.15 | aarcane | russellb, do you know of a decent guide for that setup ? |
| 19:24.31 | russellb | can't say that I do |
| 19:24.38 | russellb | the book is the best general documentation out there IMO |
| 19:24.40 | russellb | ~thebook |
| 19:24.40 | 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 |
| 19:25.15 | russellb | if you read that, you'll learn how to do much cooler stuff on your home system than that :) |
| 19:26.09 | *** join/#asterisk mindCrime (n=chatzill@cpe-075-177-141-190.nc.res.rr.com) |
| 19:26.36 | frieze | ok, so when I try to start the asterisk service I get an error message saying that it's unable to open /var/run/asterisk.pid permission denied |
| 19:26.42 | frieze | anyone have any theories? |
| 19:27.38 | ManxPower | frieze: try being root |
| 19:27.50 | frieze | it also created this weird alt.asterisk.canary.tweet.tweet.tweet entry |
| 19:27.53 | frieze | i sudo'ed it |
| 19:28.21 | ManxPower | "it" being the /etc/rc.d/init.d/asterisk script? |
| 19:28.29 | frieze | yes |
| 19:28.31 | tzafrir | or setting astvarrundir to /var/run/asterisk |
| 19:29.04 | ManxPower | you must have a hacked script because it should not create a directory called "alt.asterisk.canary.tweet.tweet.tweet" |
| 19:29.09 | tzafrir | frieze, as for the canary: search a bit on it. |
| 19:29.25 | tzafrir | it's a file, not a directory, IIRC |
| 19:29.31 | frieze | it is |
| 19:29.32 | ManxPower | tzafrir: he's running a GUI isn't he? |
| 19:29.37 | smps | one asterisk book too: http://www.the-asterisk-book.com/ |
| 19:29.39 | frieze | can't figure out what the deal is |
| 19:29.44 | frieze | two google hits |
| 19:29.47 | russellb | nah, the canary thing is part of asterisk ... |
| 19:29.49 | russellb | you can ignore it |
| 19:30.21 | frieze | no canary in the book and unhelpful google returns |
| 19:30.41 | russellb | you can blame Corydon76-dig if it's not documented anywhere :) |
| 19:31.22 | frieze | what's weird is the init.d script shows status as running afterward |
| 19:31.28 | tzafrir | yeah, isn't the asterisk source mapped by search engines somehow? |
| 19:31.36 | ManxPower | frieze: what happens when you "su -l root" then "service asterisk start"? |
| 19:31.56 | ManxPower | make sure to stop or kill asterisk first, of course. |
| 19:32.23 | frieze | ManxPower: give me a minute while I enable root login |
| 19:33.52 | frieze | ManxPower: same deal |
| 19:34.01 | frieze | permission denied |
| 19:34.28 | frieze | where do you set the asterisk uid? |
| 19:34.51 | ManxPower | frieze: "ps -axwww | grep asterisk" |
| 19:34.55 | ManxPower | does it show asterisk running? |
| 19:35.30 | tzafrir | frieze, could you please pastebin your /etc/asterisk/asterisk.conf ? |
| 19:35.57 | tzafrir | also, do you have /var/log/asterisk/messages created? anything in it? |
| 19:36.11 | deltaray | Cool, it works. Thanks ManxPower and all the other people who helped. |
| 19:36.12 | frieze | I have 6 instances of that astcanary thing running as well as /usr/sbin/asterisk |
| 19:36.25 | frieze | so it looks like it may just be the script that's broken |
| 19:36.47 | ManxPower | frieze: looks like you forgot to stop asterisk first. |
| 19:37.03 | frieze | i started it again before checking |
| 19:37.07 | ManxPower | I would expect a problem if you tried to start asterisk if it was already running |
| 19:37.27 | ManxPower | stop asterisk, use ps to make SURE it's REALLY not running |
| 19:37.46 | frieze | when I kill it that astcanary thing keeps running, btu asterisk exits correctly (apparently) |
| 19:38.52 | ManxPower | then kill astcanary |
| 19:38.55 | frieze | I did |
| 19:39.03 | tzafrir | deltaray, I suspect 5 of them are runaways. But I'm not well-familiar with asterisk 1.6 |
| 19:39.09 | ManxPower | so you have nothing in anyway related to asterisk running, right? |
| 19:39.23 | frieze | when I start again I get the same error message and while asterisk does start running another copy of astcanary starts as well |
| 19:39.29 | ManxPower | tzafrir: apparently 1.6 is for the birds 8-) |
| 19:39.43 | ManxPower | frieze: have you considered a non-beta version of Asterisk? |
| 19:39.52 | frieze | astcanary is not in my startup script so I'm assuming that the version of 1.6-current calls it somewhere |
| 19:39.53 | frieze | I have |
| 19:40.00 | tzafrir | ManxPower, this is a plain and simple permissions issue |
| 19:40.07 | atomspace | If I have two PSTN trunks (2 fxo's on 400 card), is it ok for them both to be referenced as ZAP/g0, or should I reference the second as ZAP/g1 ? |
| 19:40.14 | tzafrir | now if he'd pastebin-ed errors, we could get somewhere |
| 19:40.15 | *** join/#asterisk b1shop (n=b1shop@c-76-16-229-8.hsd1.il.comcast.net) |
| 19:40.18 | ManxPower | tzafrir: one would think that if you are root you have all permission |
| 19:40.28 | frieze | I am just wondering if it's a weird call somewhere that is using an asterisk user or something |
| 19:40.41 | ManxPower | atomspace: you reference them as however you configure the group= in zapata.conf |
| 19:40.42 | tzafrir | ManxPower, I'm still waiting for his asterisk.conf |
| 19:41.01 | frieze | sorry, my asterisk.conf is totally stock |
| 19:41.01 | atomspace | great, thanks Manx |
| 19:41.07 | frieze | I haven't edited it at all |
| 19:41.17 | frieze | astrundir is /var/run/asterisk |
| 19:41.20 | tzafrir | pastebin it anyway |
| 19:41.22 | ManxPower | frieze: doesn't matter, he still needs to see it. |
| 19:41.47 | ManxPower | frieze: "stock" in 1.6 is not the same as "stock" in the version of Asterisk most of us run. |
| 19:42.16 | frieze | ah |
| 19:42.32 | frieze | okay now I have to figure out how to get it from my x11 clipboard to my mac one |
| 19:42.33 | frieze | grr |
| 19:42.37 | frieze | I hate computers |
| 19:46.28 | ManxPower | You just need to figure out what is different about your setup as compared to the tens or hundreds of thousands other Asterisk installs out there. |
| 19:46.36 | tzafrir | http://paste.debian.net/ , allows uploading |
| 19:46.47 | tzafrir | alternatively: paste into a text editor |
| 19:47.08 | tzafrir | (shift-insert in xterm/vi ?) |
| 19:48.56 | frieze | http://paste.debian.net/3468 |
| 19:49.30 | frieze | I have yet to find an elegant x-11 to mac clipboard solution |
| 19:50.08 | frieze | the emacs copy works sometimes |
| 19:51.08 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 19:55.28 | frieze | hmm |
| 19:55.50 | frieze | the script tries to create the pid in /var/run/asterisk, but the error message says it is looking in /var/run |
| 19:56.27 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 19:57.09 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 19:57.24 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:57.34 | atomspace | In your phonebooks, in your pbx, do you have the prefix to get an outside line (commonly "9") as part of the phone numbers in your phone book? |
| 19:58.22 | magic_hat | howdy everyone. I recently switched routers on our office network and had to redo the NAT settings. Now, everything in our * setup is working... except I get no ringback when I call from an outside line. any idea what's wrong here? |
| 19:58.27 | ManxPower | atomspace: yes |
| 19:58.33 | atomspace | iow, do all outside numbers in your phonebook begin with "(" |
| 19:58.36 | atomspace | 9 |
| 19:58.42 | atomspace | oh, ok, great. Thanks again Manx |
| 19:58.46 | ManxPower | I don't have a phone book. |
| 19:58.52 | rob0 | dialplan? |
| 19:58.59 | ManxPower | I have a dialplan! |
| 19:59.22 | atomspace | you have every number that you might possibly dial in your dialplan? |
| 19:59.39 | atomspace | or just every area code? |
| 19:59.49 | rob0 | Well no, that's why $DEITY created pattern matching. |
| 19:59.50 | atomspace | ...followed by "." ? |
| 19:59.55 | ManxPower | I have pattern matches that would match every number that might be dialed. |
| 20:00.33 | atomspace | ok, I'll look into that. Thank you guys |
| 20:00.43 | ManxPower | of course, the only numbers that might be dialed are 9-1-NXX-NXX-XXXX, NXX-XXXX, N11, and 011. |
| 20:00.52 | ManxPower | oh and XXXX for extensions |
| 20:01.33 | atomspace | aha, I didn't think about pattern matching for number of digits, thanks for the hint Manx |
| 20:01.47 | rob0 | 7425 ... Rick Astley; 8463 ... time; 6665 ... tt-monkeys |
| 20:02.11 | ManxPower | our extensions are actually [2-8]XXX |
| 20:02.34 | atomspace | Manx, so then with pattern matching, can you completely get rid of having to dial 9 for an outside line? |
| 20:03.08 | atomspace | ---and still be able to dial extensions |
| 20:04.50 | magic_hat | anyone got thoughts re my problem with ringback? |
| 20:08.34 | ManxPower | atomspace: 9 is never REQUIRED, but if you have overlapping patterns you will have delays when dialing. |
| 20:08.54 | ManxPower | you can avoid that by requireing all outside calls to be dialed as 1 + areacode+number and never have an extension starting with a 1 |
| 20:09.28 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 20:12.08 | atomspace | Manx: I see, I've been thinking that 9 was required for some reason. Great, it makes sense now. |
| 20:12.35 | ManxPower | We use 9 because it makes the rest of our dial plan easier. |
| 20:14.03 | magic_hat | ringback? anyone? help? |
| 20:20.48 | atomspace | when using a pstn trunk, is it normal for asterisk to pause after the 6th digit is dialed? |
| 20:21.40 | atomspace | whether doing 7 or 10 digit local dialing, for example...7: dials 123456 pause 7. 10 digits: 123456 pause 78910 |
| 20:22.37 | atomspace | no actually, correction, it just pauses before the last digit ---- 7: 123456 pause 7. 10: 123456789 pause 10 |
| 20:23.19 | ManxPower | atomspace: yes |
| 20:23.38 | magic_hat | more info on my ringback problem... log is showing that it's playing the ringer when the call comes in. But I hear no ringing... http://pastie.caboo.se/199186 |
| 20:23.41 | ManxPower | I believe it's a bug, but it's never caused a problem as far as I know. |
| 20:23.56 | ManxPower | magic_hat: Do you get two-way audio? |
| 20:25.12 | magic_hat | manx: I know i do on outbound. Let me test on inbound as well. |
| 20:25.35 | ManxPower | you are answering the call. If you want ringback after answer you need to have /etc/asterisk/indications.conf |
| 20:26.43 | magic_hat | this won't do it? exten => s,2,Ringing |
| 20:27.32 | ManxPower | not after an Answer unless you have /etc/asterisk/indications.conf |
| 20:27.48 | magic_hat | huh. So what goes in indications.conf? |
| 20:28.02 | ManxPower | You realize that you almost never need Playtones(ring), Ringing, or "r" option to Dial, right? |
| 20:28.19 | ManxPower | magic_hat: cp /path/to/src/asterisk/configs/indications.conf.sample /etc/asterisk/indications.conf" |
| 20:28.29 | Corydon76-dig | frieze: There's an XChat for OSX native. I'd suggest it. |
| 20:28.30 | magic_hat | yeah, this is strictly on inbound calls. |
| 20:28.33 | ManxPower | then restart Asterisk (a reload might do it) |
| 20:28.47 | ManxPower | exactly WHY are you Answer'ing the call anyway? |
| 20:29.01 | Corydon76-dig | frieze: and astcanary only runs if you run Asterisk with the -p flag |
| 20:29.55 | magic_hat | Manx: The call goes to a greeter that answers it, plays company directory info, then waits for the user to dial an ext. |
| 20:31.26 | ManxPower | magic_hat: but you are answering, then ringing, then waiting 6 seconds. Makes no sense. Just do the Wait(6) like you do now, don't do the ringing or the answer. Background will BY DEFAULT answer the line if it needs to. |
| 20:31.54 | Corydon76-dig | frieze: additionally, if you use 'asterisk -rx "stop now"', then Asterisk will kill off the astcanary process all by itself |
| 20:31.59 | magic_hat | hrm, okay. that was code I snagged from somewhere on my first day with *. lol |
| 20:32.09 | ManxPower | Most of the code out there is crap |
| 20:32.28 | ManxPower | magic_hat: it's still a good idea to copy over the indications.conf |
| 20:33.50 | magic_hat | Manx: Yeah, I have indications.conf in my directory, with us set as the default. |
| 20:39.13 | magic_hat | ManxPower: I'm not getting any ringback with this... http://pastie.caboo.se/199191 |
| 20:42.07 | magic_hat | so... looks like my VOIP provider isn't providing ringback, eh? |
| 20:46.02 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 20:47.40 | magic_hat | Manx? |
| 20:49.42 | *** join/#asterisk marc7 (n=marc@S0106001c10243803.gv.shawcable.net) |
| 20:51.53 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 20:54.26 | ManxPower | magic_hat: other than lack of ringback you have no problems, right? |
| 20:56.35 | *** join/#asterisk dFence (n=chatzill@ings-c34712df.pool.mediaWays.net) |
| 20:56.46 | *** join/#asterisk Slash7719 (i=aadmin@orbital.dreamhost.com) |
| 20:57.54 | Slash7719 | hello - anybody listening? |
| 20:57.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:58.09 | Slash7719 | I got a couple of problem/questions regarding my asterisk. |
| 20:58.52 | Slash7719 | question one is regarding call transfers between sip (grandstream products) and zap (digium products) channels |
| 20:59.17 | Slash7719 | is there anyone here who might be willing to listen? |
| 20:59.18 | mackes | ok |
| 20:59.23 | mackes | Hey |
| 20:59.27 | mackes | Whats Happening |
| 20:59.39 | Slash7719 | I have been running asterisk for several years with only zap channels |
| 20:59.45 | mackes | cool |
| 20:59.59 | Slash7719 | I recently updated to the most recent stable... dot 19 something release |
| 21:00.07 | mackes | 1.4? |
| 21:00.21 | Slash7719 | then added a grandstream 386 module and a grandstream 1200 two line phone. |
| 21:00.35 | mackes | ok |
| 21:00.45 | Slash7719 | everything works great, but my ability to transfer calls appears to be wacked out. |
| 21:01.01 | Slash7719 | I can still transfer using the hookflash from zap to zap |
| 21:01.07 | mackes | OK, What happens |
| 21:01.28 | Slash7719 | but from a sip phone to zap, it gets messed up along the way. |
| 21:01.41 | Slash7719 | I can put the call on hold with a hookflash |
| 21:01.49 | Slash7719 | then dial an extension... ok |
| 21:02.12 | Slash7719 | if I attempt a blind xfer, by hanging up.. the xfer fails. |
| 21:02.34 | mackes | Have you attempted Sip to Sip on your 2 line phone? |
| 21:02.49 | Slash7719 | if I wait until the other phone picks up, and then hang up, it appears to go thru |
| 21:03.04 | Slash7719 | I have tried sip to sip, and I get mixed results there depending on the phone. |
| 21:03.09 | mackes | that is an attended transfer- |
| 21:03.24 | mackes | So, Blind transfers are the issue |
| 21:03.33 | mackes | are these involving your 386 or not? |
| 21:03.44 | Slash7719 | I have read that significant changes are being made to the xfer code, where it is becoming more traditional like |
| 21:03.55 | Slash7719 | yes, these do involve the 386 |
| 21:04.11 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 21:04.32 | Slash7719 | are there known difficulties with transferring between technologies like sip to zap? |
| 21:05.24 | Slash7719 | thanks mackes... yes, blind transfers are a problem. Also attended xfer also fails sometimes with both sides geting MOH |
| 21:05.24 | jbot | pas de quoi, Slash7719 |
| 21:05.45 | *** join/#asterisk RoyK (n=roy@ti211310a080-6545.bb.online.no) |
| 21:06.15 | Slash7719 | sorry jbot, I do not understand french (is it?) |
| 21:06.34 | mackes | Hmmm |
| 21:06.52 | mackes | My best guys is that the issue lies in the 386 setup somewhere |
| 21:07.00 | mackes | they ar flaky |
| 21:07.17 | mackes | How are you doing ZAP? Digium card? |
| 21:07.44 | Slash7719 | yes, I have suspected this.. and read some too. But so far, I am having pretty good results except for xfer. |
| 21:08.10 | Slash7719 | zap is via digium TD400, IIRC |
| 21:08.12 | mackes | Oh, they do work- they just have a wacky GUI, and alot of options. |
| 21:08.42 | mackes | Ok, so, I would take the 386 out of the mix, and experent between the your ZAP channels and a true SIP phone |
| 21:08.58 | Slash7719 | You mean like the GS 1200? |
| 21:09.19 | Slash7719 | OK, so here is my story on the GS 1200. |
| 21:09.35 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 21:09.39 | Slash7719 | I am somewhat new with these phones, so give me a shot. |
| 21:09.50 | mackes | Yep |
| 21:09.52 | Slash7719 | the gs 1200 does not have a hookflash |
| 21:10.02 | mackes | It has a transfer button |
| 21:10.15 | Slash7719 | I hit the transfer button, and get a message to enter extension, |
| 21:10.20 | mackes | And with the 386, you dont use hook flash, because its sip. |
| 21:10.37 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 21:10.43 | Slash7719 | I enter a single digit and I get a message that the extension does not exist. |
| 21:11.00 | mackes | when does that happen? |
| 21:11.23 | Slash7719 | w/rt 386... a hookflash on the attached handset does do the job. So go figgure. |
| 21:11.37 | mackes | Hmmmm |
| 21:11.51 | Slash7719 | I call the 1200 from a zap phone. |
| 21:11.52 | magic_hat | hey all. I figured things out so that I now hear ringing on inbound calls. However, the first part of my greeting message is being cut off somehow. ideas? http://pastie.caboo.se/199202 |
| 21:12.18 | Slash7719 | I answer the call and can talk |
| 21:12.20 | mackes | I would setup your features.conf for your transfers |
| 21:12.55 | Slash7719 | then I hit transfer and a single digit extension and get invalid extension message |
| 21:14.01 | Slash7719 | I have looked at features.conf, but the defaults are working for me elsewhere ... ie zap channels. what to change? |
| 21:14.19 | mackes | Yeah, The phone should not attempt to transfer that quickly. |
| 21:14.44 | Slash7719 | OK |
| 21:14.54 | Slash7719 | I got another question.... |
| 21:15.11 | Slash7719 | When I have an incoming call. |
| 21:15.44 | Slash7719 | If the person on the incoming call sends a DTMF tone during the conversation |
| 21:16.00 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 21:16.11 | Slash7719 | I get a screaming DTMF tone on my phone, and the conversation is essentially over. |
| 21:16.20 | mackes | yep |
| 21:16.26 | Slash7719 | the debug logs show that the tone starts, but it never stops. |
| 21:17.03 | Slash7719 | I got two installations, with different releases and both are doing it |
| 21:17.18 | mackes | Your dial command for that exenention has the switch enabled to allow the callee to send commands to the feature.conf |
| 21:18.01 | Slash7719 | the switch in the dial command is the "t" tee, right? |
| 21:18.08 | Slash7719 | got it. |
| 21:18.46 | Slash7719 | callee switch is a "T" and that is absent. |
| 21:18.59 | Slash7719 | let me check. |
| 21:20.02 | *** join/#asterisk SomethingISODD (n=dancole@S010600a0d1757bfb.cg.shawcable.net) |
| 21:20.09 | SomethingISODD | hello all |
| 21:20.21 | *** join/#asterisk Dossy (i=dossy@aolserver/dossy) |
| 21:20.21 | SomethingISODD | can you run chan_h323 and gnugk on the same server? |
| 21:21.38 | Slash7719 | the dial command that dials the extension has a lower case "t" included. |
| 21:22.35 | Slash7719 | this should allow the called extension to enter digits, not the callee |
| 21:24.32 | Slash7719 | Thanks for listening mackes... anybody else got a clue what might be going on? |
| 21:24.32 | jbot | Slash7719: gern geschehen |
| 21:27.51 | RoyK | ~lart kram |
| 21:27.51 | jbot | gives kram an extra strength ACME sleeping pill, sending kram to sleep for 150 years, and awakening to seven strange dwarfs and a large apple |
| 21:31.46 | mackes | Sorry Slash |
| 21:31.49 | mackes | Later on |
| 21:36.29 | *** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz) |
| 21:46.16 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:51.05 | mackes | ~mackes |
| 21:51.06 | jbot | from memory, mackes is the man |
| 21:51.46 | drmessano | That |
| 21:51.52 | drmessano | is weak |
| 21:52.42 | mvanbaak | sorry, no microwave |
| 21:53.24 | mvanbaak | hhmm, my mailstore is pretty clean |
| 21:53.38 | drmessano | lol |
| 21:53.39 | mvanbaak | since 1999 I archived 6356 mails |
| 21:53.45 | mvanbaak | that's it |
| 21:53.54 | mvanbaak | http://michiel.vanbaak.info/temp/mailboxes |
| 21:54.02 | mvanbaak | that's my mutt mailboxes config file |
| 21:54.07 | drmessano | I had 11,000 e-mails in my deleted items once |
| 21:54.20 | mvanbaak | I dont have a deleted items |
| 21:54.26 | mvanbaak | in mutt, when I delete a mail it's gone |
| 21:54.55 | drmessano | I don't save anything anymore.. except some sent items |
| 21:55.35 | mvanbaak | my =Sent is 2158 items |
| 21:57.41 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:01.30 | mvanbaak | oops |
| 22:01.43 | mvanbaak | just hit 'Enter' on a mail with 2 dvd images |
| 22:02.06 | mvanbaak | Fetching message: 480MB/12GB |
| 22:02.33 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 22:03.31 | drmessano | lol |
| 22:05.18 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 22:05.36 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 22:07.57 | jameswf-home | wtf is DAHDI ? |
| 22:08.12 | jameswf-home | zaptel is finaly getting a name change? |
| 22:09.30 | mvanbaak | jameswf-home: gheh, I wondered as well |
| 22:09.31 | drmessano | Where do you see that? |
| 22:09.53 | mvanbaak | drmessano: svn-commits@lists.digium.com |
| 22:09.58 | drmessano | oh |
| 22:10.17 | mvanbaak | it was reverted 5 minutes later |
| 22:10.23 | jameswf-home | some secret stuff goin on? |
| 22:10.29 | mvanbaak | must be |
| 22:11.40 | drmessano | Digium Appliance Hardware Decoder Interface |
| 22:11.48 | drmessano | it was worth a shot |
| 22:12.04 | mvanbaak | gheh, nice one |
| 22:12.16 | drmessano | Digium Appliance something, I would bet |
| 22:12.42 | jameswf-home | ~wiki dahdi |
| 22:12.58 | jameswf-home | ~dahdi |
| 22:13.06 | mvanbaak | infobot is nice :) |
| 22:13.07 | jameswf-home | jbot doesnt know either |
| 22:13.09 | rob0 | ~daddy |
| 22:13.10 | jbot | i guess daddy is copperd |
| 22:13.52 | mvanbaak | I thought daddy was [kthreadd] |
| 22:14.03 | drmessano | Digium Appliance Hardware Dialing Interface |
| 22:14.19 | drmessano | Hmmm |
| 22:14.31 | mvanbaak | Digium Appliance Hardware Dying Inpeace |
| 22:14.42 | drmessano | Digium Appliance High Density Interface |
| 22:15.13 | mvanbaak | Hail Doomsday Interface |
| 22:16.03 | drmessano | OH NOOOO |
| 22:16.26 | drmessano | ~markster |
| 22:16.33 | drmessano | EEEEVIL |
| 22:16.47 | drmessano | ~seen markster |
| 22:16.51 | jbot | i haven't seen 'markster', drmessano |
| 22:16.51 | mvanbaak | ~mvanbaak |
| 22:16.55 | drmessano | I bet |
| 22:17.21 | drmessano | Because he's in the CAYMANS waiting for his LITTLE PLAN to be EXECUTED |
| 22:17.31 | mvanbaak | lol |
| 22:17.44 | mvanbaak | I bet it involves a lot of women and boose for da men |
| 22:17.50 | drmessano | waits as tens of thousands of asterisk boxes begin wardialing the pentagon and trying the launch codes |
| 22:18.06 | drmessano | CPE...1704....TK..... |
| 22:18.09 | drmessano | .... |
| 22:18.09 | drmessano | S |
| 22:18.13 | drmessano | OMG THEY GOT THE CODES |
| 22:18.17 | mvanbaak | lol |
| 22:18.22 | mvanbaak | that would be great |
| 22:18.37 | drmessano | ~wiki CPE1704TKS |
| 22:18.44 | drmessano | Hmmm |
| 22:18.54 | mvanbaak | ~42 |
| 22:18.54 | jbot | i heard 42 is the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything |
| 22:19.14 | mvanbaak | I bet the code is 42 |
| 22:19.32 | mvanbaak | I know my root ssh password is |
| 22:19.41 | mvanbaak | ;) |
| 22:19.44 | drmessano | HA!! YES!!!! http://www.zazzle.com/pd/find/qs-CPE1704TKS/cg-103118908023908739 |
| 22:20.01 | rob0 | CPE? shouldn't that be OPE? |
| 22:20.16 | drmessano | no |
| 22:20.33 | rob0 | is stuck in a Strangelove mindset |
| 22:20.42 | mvanbaak | Client Something |
| 22:21.14 | rob0 | Peace on Earth, Purity of Essence, Power over Ethernet |
| 22:21.33 | mvanbaak | I'm off to bed |
| 22:21.36 | mvanbaak | latero all |
| 22:21.44 | drmessano | later |
| 22:21.46 | drmessano | Funny thing is |
| 22:22.01 | drmessano | The code is displayed as CPE1704TKS on the big screens |
| 22:22.14 | drmessano | But on the WOPR, when they cut over to it, it shows JPE1704TKS |
| 22:22.21 | drmessano | Small continuity error |
| 22:22.27 | rob0 | Wing Attack, Plan R |
| 22:23.04 | *** join/#asterisk RoyK (n=roy@ip-229-22-149-91.dialup.ice.no) |
| 22:23.07 | drmessano | After very careful consideration sir, I've come to |
| 22:23.07 | drmessano | <PROTECTED> |
| 22:23.16 | drmessano | One of the best lines in the movie |
| 22:26.33 | jameswf-home | wow neat someone on Dialup |
| 22:33.32 | jameswf-home | Maybe Who's your DahDi |
| 22:35.44 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 22:55.28 | *** part/#asterisk clickonce (n=bar@unaffiliated/clickonce) |
| 23:03.56 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 23:16.26 | dFence | is it generally possible to have a sip-username with like "Firstname, Lastname"? |
| 23:16.42 | dFence | (with a space in the username) |
| 23:18.45 | drmessano | Why would you want to do that? |
| 23:20.27 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 23:21.24 | jeffspeff | i'm can't one-touch recording to work on one part... I have "exten => *9,1,Dial(${jeffclay}/mycellphone#,30,mwW)" defined, and it works. I use it for emergency contact from my ivr menus. the call goes through fine, but it won't let me use the one-touch recording.... how can i fix this? |
| 23:22.04 | *** join/#asterisk Bananaskin (n=mike@78-105-246-34.zone3.bethere.co.uk) |
| 23:23.47 | dFence | drmessano: would facilitate a lot of things. |
| 23:26.57 | jeffspeff | does anybody have any ideas? |
| 23:27.08 | *** join/#asterisk dFence (n=chatzill@ings-c34712df.pool.mediaWays.net) |
| 23:27.44 | jeffspeff | i can't get one-touch recording to work on one part... I have "exten => *9,1,Dial(${jeffclay}/mycellphone#,30,mwW)" defined, and it works. I use it for emergency contact from my ivr menus. the call goes through fine, but it won't let me use the one-touch recording.... how can i fix this? the recording works fine everywhere else |
| 23:32.05 | jameswf-home | I have an idea for a super fast race car that runs on cooked tuna |
| 23:34.24 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 23:37.06 | jeffspeff | jameswf-home, i was referring to my question. |
| 23:38.21 | jameswf-home | oic// |
| 23:39.06 | drmessano | I have an idea |
| 23:39.21 | drmessano | Donuts shaped like pencils |
| 23:44.26 | jameswf-home | omfg dude krisy kream just reopened in Arizona it is insane... I went in to a sugar induced colma |
| 23:44.44 | drmessano | lol |
| 23:44.45 | *** join/#asterisk Zorix (n=Brandon@c-98-208-144-134.hsd1.fl.comcast.net) |
| 23:44.55 | JT | what is it, 20% sugar? |
| 23:44.57 | drmessano | Krispy Kreme donuts are nuts |
| 23:45.00 | drmessano | No |
| 23:45.04 | drmessano | 85% or so |
| 23:45.23 | jameswf-home | It's like crack for fat people.... dare I say better than Bacon |
| 23:45.41 | Zorix | hi guys, i just installed asterisknow after i have been using the beta a year ago.. relearning everything but I got my system up and tested calling the conference bridge and i get this long silence and very choppy music.. i seem to remember something about a sound library issue last time |
| 23:45.45 | Zorix | any ideas? |
| 23:45.51 | jameswf-home | ~asterisknow |
| 23:45.52 | jbot | somebody said asterisknow was based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 23:45.55 | drmessano | Have you ever seen the "Krispy Kreme $6 Burger"? |
| 23:45.56 | *** join/#asterisk bobbym (n=bob@unaffiliated/bobbym) |
| 23:45.56 | JT | i doubt it's 85% |
| 23:46.08 | jameswf-home | sadly yes |
| 23:46.09 | bobbym | how can i send the sip debug output to an file? |
| 23:46.13 | drmessano | LOL |
| 23:46.15 | drmessano | I love that |
| 23:46.21 | Zorix | ok fine if you want me to ask somewhere else |
| 23:46.26 | drmessano | Hardees burger, two krispy kreme donuts |
| 23:46.40 | jeffspeff | Zorix, #asterisknow |
| 23:46.57 | jameswf-home | I ate a double six dollar burger once... I thought I would be sick |
| 23:46.59 | Zorix | i am there but i think the core problem is with the asterisk service on that distribution |
| 23:47.07 | drmessano | A glazed donut is 50% sugar to start with |
| 23:47.16 | jameswf-home | Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 23:47.16 | drmessano | So cover it with a Sugar glaze |
| 23:47.19 | drmessano | ok, 60% |
| 23:48.56 | bobbym | how can i save the sip debug informations into a file? |
| 23:49.26 | JT | an original glazed doughnut only has 20% sugar |
| 23:49.44 | drmessano | ~humor |
| 23:49.44 | jbot | humor is, like, Q: Why are the streets of Paris lined with trees? A: Because Germans like to march in the shade. |
| 23:49.53 | drmessano | It's called "joking" |
| 23:49.57 | JT | the highest is like 34% |
| 23:50.01 | JT | okay |
| 23:50.17 | JT | a can of coke has as much sugar as an original glazed |
| 23:50.22 | jameswf-home | bobbym: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+logger.conf |
| 23:50.39 | bobbym | jameswf-home: thanks |
| 23:50.40 | JT | actually |
| 23:50.52 | JT | a can of cake has about 3 times more sugar than an original glazed |
| 23:51.12 | bobbym | jameswf-home: look i tried to create a new logger.conf entry file, but those debug informations that appers in console seems not be saved into the file |
| 23:51.23 | JT | s/cake/coke/ |
| 23:51.27 | bobbym | i'm creating the file using notice,warning,error,verbose,debug |
| 23:51.30 | JT | can of cake would be interesting |
| 23:52.25 | jeffspeff | is one-touch recording supposed to work on things other than the registered phones (in my case, the sip phones)? |
| 23:52.39 | jeffspeff | like from a cell phone |
| 23:53.50 | drmessano | Don't even expect anything DTMF to ever work from a cell phone |
| 23:54.31 | jeffspeff | SomethingISODD, in reference to my question from earlier... the answer is not possible? |
| 23:54.39 | jeffspeff | *so |
| 23:54.45 | JT | dtmf works fine from a cell phone for me |
| 23:54.46 | jeffspeff | i can't get one-touch recording to work on one part... I have "exten => *9,1,Dial(${jeffclay}/mycellphone#,30,mwW)" defined, and it works. I use it for emergency contact from my ivr menus. the call goes through fine, but it won't let me use the one-touch recording.... how can i fix this? the recording works fine everywhere else |
| 23:55.32 | *** join/#asterisk dachary (n=loic@magenta.dachary.org) |
| 23:56.15 | *** join/#asterisk RoyK (n=roy@ip-170-58-149-91.dialup.ice.no) |
| 23:56.42 | dachary | hi, can someone recommend a robust wifi / SIP phone ? |
| 23:56.58 | jeffspeff | ~sip phone |
| 23:56.59 | jbot | i guess sip phone is http://www.voip-info.org/tiki-index.php?page=Asterisk+phones |
| 23:57.02 | JT | ~wifisip |
| 23:57.03 | jbot | [~wifisip] Please refer to ~wifivoip for info on this. |
| 23:57.10 | JT | ~wifivoip |
| 23:57.10 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 23:57.38 | mackes | You know... that isnt true. |
| 23:57.49 | dachary | i've seen all this, thanks. I'd like a recommendation to help me chose ? |
| 23:57.50 | mackes | Lots of WiFi Sip phones are in use ALL of the time |
| 23:58.47 | jameswf-home | # w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 23:58.47 | jameswf-home | # W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 23:58.52 | mackes | Hitachi makes a Very good SIP phone |
| 23:59.06 | JT | just because people use them, doesn't make it a good idea |
| 23:59.10 | JT | dect is a better option |
| 23:59.21 | jameswf-home | prefers dect over wifi |
| 23:59.51 | mackes | ok JT |