| 00:00.30 | TommyTheKid | it was a test conference, we asked everyone who had some free time to dial in in our IT org |
| 00:01.27 | TommyTheKid | we are trying to figure out how to roll this out company wide, but there are problems with H.323, PRI cards are an expensive alternative, and we still had an issue that I would like to squash |
| 00:02.16 | TommyTheKid | I think the scalability would be better with the talker detection thing, as opposed to mixing in 77-(2 or 3) silent streams |
| 00:02.56 | quid246 | Yeah... definitely silence suppression would help |
| 00:02.58 | TommyTheKid | We really need it to be able to handle 650-ish people for our All Hands (uh maybe less after the RIF) |
| 00:03.09 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 00:03.29 | TommyTheKid | I am notseeing that happen unless we can link server to server (meetme clustering?) |
| 00:03.34 | quid246 | with that many calls... what is your CPU running at? |
| 00:03.59 | TommyTheKid | well, we would probably run it in a Sun Fire x4600 (or whatever the 8 way Opteron 280 is) |
| 00:04.44 | hoytbowUE | Can anyone please tell me how to get the mysql cdr support going in 1.2.x, I have downloaded latest tarball for asterisk-addons and latest trunk svn and when I compile, res_config_mysql.so is not there... Menuselect doesnt have a mysql option under the res... Mysql-server 4.1.11 and the dev pkg are installed |
| 00:04.46 | TommyTheKid | dual core opteron with 8 physical chips = 16 way |
| 00:05.17 | quid246 | I'm trying to figure out what kind of server to run... the info is all over the place... I want capability to do a mix of SIP/IAX (400 channels/200 calls) |
| 00:05.50 | TommyTheKid | I suppose that depends a lot on the codecs more than anything |
| 00:05.55 | quid246 | hoy: I dunno off hand, but I followed the WIKI and I got it to work for me |
| 00:06.03 | quid246 | Tommy: Yeah.. I only plan on running ULAW |
| 00:06.09 | TommyTheKid | we were using mostly PRI lines (96) and a few SIP people on ulaw |
| 00:06.10 | quid246 | so no transcoding |
| 00:06.45 | TommyTheKid | quid246: then it depends on what you plan to do.. lots of IVR type functions might take more than simple call routing |
| 00:07.08 | TommyTheKid | I can't claim to be an expert cause 77 people on my server today was the most we have ever had |
| 00:07.34 | quid246 | Tommy: Nope, just pure bouncing of calls |
| 00:07.40 | quid246 | hehe |
| 00:07.51 | quid246 | but * will stay in the media stream, no reinvites |
| 00:08.02 | TommyTheKid | I would think you could get away with something like a X2100 (one dual core opteron CPU) |
| 00:08.14 | TommyTheKid | but like I said, no expert here |
| 00:08.21 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:09.48 | quid246 | yeah I was thinking Opteron... regardless I am going to put mySQL on another machine, |
| 00:10.25 | TommyTheKid | yeah, that gives you the ability to scale up another server quite easily |
| 00:11.19 | quid246 | yeah, I think for the mySQL though..> I can probably go "bottom of the barrel" and run something like an older P4. |
| 00:11.39 | TommyTheKid | hehe |
| 00:12.04 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 00:12.13 | TommyTheKid | I think I'd get something that I knew was reliable, cause if you use it for call routing (for example) it will be down hard when mysql is down |
| 00:12.59 | quid246 | true enough. |
| 00:13.28 | TommyTheKid | the "lower end" X2100 is only like $750 I think (retail) |
| 00:14.01 | TommyTheKid | it would be pretty cool to have the SP for remote powercycles and/or console access in case of unforseen events |
| 00:14.19 | TommyTheKid | aka "remote lights out management" |
| 00:14.30 | *** join/#asterisk hfb (n=hfb@pool-71-106-220-165.lsanca.dsl-w.verizon.net) |
| 00:15.00 | quid246 | yeah, I've never seen KVMoIP... but it sounds pretty coo. |
| 00:15.01 | quid246 | cool |
| 00:18.02 | *** join/#asterisk w32 (n=w32@c-71-193-124-77.hsd1.il.comcast.net) |
| 00:24.38 | *** join/#asterisk henk (n=marius@s5593c2e9.adsl.wanadoo.nl) |
| 00:25.26 | henk | Hi is it possible to use asterisk to send sms though a sip registrar like sipdiscount? |
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| 00:28.33 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 00:44.17 | Tbb | Thanks Guys!... You all kick a$$! |
| 00:44.43 | Tbb | l8r |
| 00:44.45 | *** part/#asterisk Tbb (n=ccorbin@bravo.telebarbados.com) |
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| 00:55.56 | TommyTheKid | so, the "-- Playing 'some-sound' (language 'en')" could *really* use a channel name in it.. is it as simple as adding it on the end, or do I need to jump thru flaming hoops to get the channel name there? |
| 00:56.09 | TommyTheKid | .. or better stated, is there some reason its not already there :) |
| 00:56.18 | *** join/#asterisk draco_710 (n=tlambeth@74-34-82-173.dsl1.chi.il.frontiernet.net) |
| 00:57.00 | ManxPower | MANY of those sorts of messages could use a channel name on them |
| 00:57.03 | De_mon | how do i make sure all my sound files are the same volume using sox? |
| 01:00.19 | ariel_ | Hello everyone |
| 01:04.51 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
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| 01:19.55 | [TK]D-Fender | Hey I've got a funny problem I could use a hand pinning down : using GotoIfTime it doesn't seem to be working |
| 01:19.57 | [TK]D-Fender | I do : -- Executing GotoIfTime("Zap/1-1", "08:00-17:00|mon-fri|*|*?5:10") in new stack |
| 01:20.44 | [TK]D-Fender | And the time on the server (per "date") = Wed Aug 9 20:14:24 CDT 2006 |
| 01:21.07 | [TK]D-Fender | Double checked my syntax and something is still off and I'm missing it... |
| 01:26.58 | xheliox | I'm using the exact same syntax, and it's working. |
| 01:27.09 | [TK]D-Fender | :/ |
| 01:27.30 | xheliox | Sorry, not quite... but it should be basically the same... exten => s,3,GotoIfTime(8:00-18:00|mon-fri|*|*?mainmenu-day,s,1) |
| 01:27.42 | [TK]D-Fender | maybe the leading 0 is bad... |
| 01:27.52 | xheliox | Hmm, perhaps. |
| 01:28.06 | [TK]D-Fender | nope |
| 01:29.14 | *** join/#asterisk grabeez (n=Owner@141.152.252.82) |
| 01:29.29 | [TK]D-Fender | UGH... heres the ${TIMESTAMP} NoOp'd - 20060809-202244 |
| 01:29.35 | [TK]D-Fender | so it looks 100% legit.... |
| 01:30.47 | [TK]D-Fender | Wait a sec.... |
| 01:30.58 | [TK]D-Fender | there doesnt' seem to be an "else" clause..... |
| 01:31.01 | [TK]D-Fender | I may have goofed |
| 01:32.19 | [TK]D-Fender | Yup... I = silly :) |
| 01:32.23 | [TK]D-Fender | *sigh* |
| 01:32.37 | file | you're not silly... you're uh... |
| 01:32.40 | file | okay, maybe silly |
| 01:33.03 | [TK]D-Fender | :P |
| 01:33.08 | xheliox | lol |
| 01:33.09 | [TK]D-Fender | least I'm not a troll! |
| 01:33.17 | file | or are you? |
| 01:33.38 | [TK]D-Fender | that'll learn me... for assuming apps were built uniformly and SANELY. |
| 01:33.40 | *** join/#asterisk Nuwave (n=nuwave@CPE00131075a581-CM0013718c292c.cpe.net.cable.rogers.com) |
| 01:33.44 | xheliox | silly boy. |
| 01:33.46 | file | assumptions kill, mmmk? |
| 01:33.54 | file | a doctor could just *assume* you were healthy |
| 01:34.06 | [TK]D-Fender | file : I jsut need to fine tune my targets ;) |
| 01:34.08 | Qwell | I've been meaning to fix the *If* apps |
| 01:34.09 | file | and then boom, you're a VoIP provider - and that's unhealthy |
| 01:34.26 | [TK]D-Fender | lol |
| 01:34.47 | xheliox | Ugh, no wonder I've been so worn down lately. |
| 01:34.59 | Qwell | xheliox: unhealthy? |
| 01:34.59 | [TK]D-Fender | I keep trying to change my laser printers from "stun" to "kill" but haven't quite gotten that down pat yet... |
| 01:35.37 | *** join/#asterisk LoneShadow (n=duh@59.92.131.157) |
| 01:38.50 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 01:39.16 | [TK]D-Fender | somewhat |
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| 01:48.12 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:48.36 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 01:48.37 | Snake-Eyes | Any one know why type=peer (sip.conf) isn't looked at for incoming calls? |
| 01:50.22 | Snake-Eyes | I have two trunks to the same machine, one is friend, other is peer yet asterisk seems to choose which trunk to use by the order they are set out by and doesn't look at the type |
| 01:53.16 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-88.cybersurf.com) |
| 01:55.48 | *** join/#asterisk omal (n=omal@cpe-24-164-111-184.neo.res.rr.com) |
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| 02:02.15 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
| 02:02.19 | dos000 | howdy |
| 02:02.45 | dos000 | how do you get rid on asterisk once it is installed ? there is no uninstall ? |
| 02:02.54 | De_mon | Snake-Eyes peers are only for outbound calls |
| 02:03.33 | De_mon | dos000 how did you 'install' it? |
| 02:04.09 | *** join/#asterisk Zorix (n=Brandon@226-124.34-65.swfla.res.rr.com) |
| 02:05.24 | dos000 | De_mon, this is version 1.2.10 and i installed it using the src make install .. winth no istall prefix |
| 02:06.02 | Snake-Eyes | De_mon, then how come, when i swap the order of the trunks in sip.conf the peer trunk will have calls coming in on it instead of the friend trunk |
| 02:06.53 | De_mon | dos000 look at the makefile and see if there is a remove option... or just see what intall does and delete the files manually |
| 02:07.51 | dos000 | De_mon, there is no install nor remove ... but it seems there is a patch http://bugs.digium.com/file_download.php?file_id=8805&type=bug |
| 02:08.09 | De_mon | huh? make install |
| 02:08.19 | De_mon | Snake-Eyes are you reloading sip.conf? |
| 02:08.45 | De_mon | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+type |
| 02:09.23 | Snake-Eyes | De_mon, reloading everything :) |
| 02:09.42 | Snake-Eyes | De_mon, been there ages ago |
| 02:09.56 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 02:10.09 | De_mon | well, like it says peers are for outgoing calls, so what makes you think it's being used for an incoming call |
| 02:10.48 | De_mon | either you a) didn't configure it right, b) are jumping to the wrong conclusion or c) need to file a bug report. I'm leaning towards b right now |
| 02:11.44 | Snake-Eyes | De_mon, cause its says it in the cli, i gave both trunks a different account, I can see which account it is trying to use |
| 02:11.59 | De_mon | paste your sip.conf and whatever CLI output that you're drawing conclusions from on pastebin.ca and go from there |
| 02:12.14 | De_mon | (don't forget to blank out acct info) |
| 02:13.13 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:13.40 | *** join/#asterisk deb_user (n=Hypnotis@70-59-108-105.albq.qwest.net) |
| 02:13.54 | deb_user | hello all |
| 02:14.13 | deb_user | i'm having some trouble dialing out on a zaptel interface |
| 02:14.33 | deb_user | sometimes the call goes out, and others it hangs and I get an operator "your call did not go through" message |
| 02:14.47 | deb_user | I'm thinking this is codec related |
| 02:15.00 | deb_user | anybody have any experience with this sort of thing can give me some tips? |
| 02:16.18 | *** join/#asterisk s0lid (n=jlq@124.106.157.190) |
| 02:17.51 | De_mon | deb_user did you try adding W+? |
| 02:17.55 | De_mon | http://www.voip-info.org/wiki/view/TDM400P |
| 02:18.09 | deb_user | de_mon: i'll check that out |
| 02:18.12 | deb_user | thanks for the tip |
| 02:18.40 | De_mon | ^^ just googled exactly what you asked... |
| 02:18.56 | deb_user | hehe |
| 02:19.17 | dos000 | De_mon, so nice of you ... to answer these ! |
| 02:20.00 | De_mon | oh yeah.. I was looking for Makefile source to see if there was a remove/uninstall option |
| 02:20.24 | Snake-Eyes | De_mon, http://pastebin.ca/124445 |
| 02:21.01 | dos000 | De_mon, no .noo .. i meant the help you are providing. there was no tongue in cheek |
| 02:21.33 | deb_user | ummm... |
| 02:21.37 | De_mon | dos000 trunk has uninstall support btw |
| 02:21.50 | deb_user | ok, sounds like the guy was having my same problem |
| 02:22.05 | deb_user | but, I don't know what adding W+ to AMP is |
| 02:22.18 | dos000 | De_mon, not ssure why this did not make it to release 1.2.10 ... |
| 02:25.18 | De_mon | Snake-Eyes I'll assume the @x.x.x.x is to the wrong host... |
| 02:25.54 | De_mon | Snake-Eyes line 34, is using [back-trunk-ulaw] ? |
| 02:27.03 | Snake-Eyes | De_mon, its replying to back-trunk, the @x.x.x.x is another server sitting behind back-trunk |
| 02:27.18 | Snake-Eyes | De_mon, back-trunk is a plain ser setup |
| 02:27.45 | Snake-Eyes | De_mon, it just forwards to asterisk |
| 02:29.00 | Snake-Eyes | De_mon, if i swap the order of the two trunks in sip.cfg it works perfectly and line 33 changes to 8880006111 |
| 02:29.38 | De_mon | are both contexts using the same host? |
| 02:29.43 | Snake-Eyes | yes |
| 02:30.10 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 02:30.23 | De_mon | I don't think you can receive calls from the same ip on 2 different accounts |
| 02:30.24 | deb_user | anybody know where to add the w+ to get asterisk to wait a half second before dialing out? |
| 02:30.31 | deb_user | in extensions.conf |
| 02:30.34 | deb_user | or in zapata.conf? |
| 02:30.42 | Qwell | deb_user: extensions.conf |
| 02:30.49 | Qwell | Dial(Zap/1/123w456) |
| 02:31.13 | deb_user | I can put the w anywhere within the number? |
| 02:31.19 | deb_user | before the 1? |
| 02:31.23 | Qwell | anywhere |
| 02:31.25 | deb_user | and what about the plus? |
| 02:31.26 | Snake-Eyes | De_mon, I dont want back-trunk-ulaw (type peer) to recieve calls only make outgoing calls to x.x.x.x |
| 02:31.31 | Qwell | + does nothing, afaik |
| 02:31.44 | deb_user | ok |
| 02:31.50 | deb_user | thanks qwell |
| 02:31.53 | deb_user | much obliged |
| 02:31.56 | dos000 | de_mon isnt there a way to tell asterisk where to look for ?? it looks like the make file is in need of some autoconf love |
| 02:32.15 | De_mon | oh yeah.. forgot the problem |
| 02:32.39 | dos000 | De_mon, where to look for zaptel, libpri headers ... |
| 02:32.49 | deb_user | qwell: what if i'm using an expression? like exten => _6NXXXXXX,1,Dial(${OUTBOUND6}/${EXTEN:1}) |
| 02:32.59 | Qwell | deb_user: w${EXTEN:1} |
| 02:33.01 | dos000 | De_mon, i am on to other problems .. i fixed that using the patch |
| 02:33.06 | deb_user | qwell: thanks again |
| 02:33.51 | dos000 | De_mon, the make file is kind of ... retarded |
| 02:34.26 | dos000 | De_mon, but i know autoconf is a nightmare |
| 02:35.59 | De_mon | Snake-Eyes duno, i'm stumped. |
| 02:36.18 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:36.37 | deb_user | looks like its working fine now |
| 02:36.39 | deb_user | thanks guys |
| 02:36.40 | Snake-Eyes | De_mon, ah ok :( |
| 02:37.25 | Snake-Eyes | if i keep the order it will be fine but should it ever be changed..... |
| 02:37.36 | deb_user | can anybody recommend a good sip softphone besides x-ten lite? |
| 02:37.47 | Snake-Eyes | deb_user, sjphone |
| 02:38.58 | Snake-Eyes | De_mon, maybe I'll email the mailing list and see what comes back, thanks for the help |
| 02:40.04 | deb_user | how bout a good iax softphone snake eyes? I like to keep my options open :) |
| 02:40.56 | Snake-Eyes | deb_user, some one was pushing a iax phone in here the other day but i forget the name ;( |
| 02:42.41 | LoneShadow | Idefisk is a good iax soft phone |
| 02:42.57 | LoneShadow | I think its on asteriskguru.com |
| 02:43.07 | deb_user | oh, sjphone is cool |
| 02:43.22 | deb_user | I hate the contact manager in x-ten |
| 02:43.25 | deb_user | its awful |
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| 02:48.12 | dos000 | deb_user, very awful |
| 02:49.24 | dos000 | deb_user, did you try openwengo ? |
| 02:49.41 | dos000 | hmmm .. its not iax maybe |
| 02:52.35 | deb_user | didn't try openwengo |
| 02:52.38 | deb_user | i'll check it out now |
| 02:53.21 | deb_user | i've also been using kiax |
| 02:53.26 | deb_user | nice little iax softphone |
| 02:53.35 | deb_user | a little limited in terms of available codecs though |
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| 03:02.20 | derrick_ | beep |
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| 03:17.58 | omal | ok...so i'm trying to decide if my asterisk issues are bad asterisk config or firewall/routing issues |
| 03:18.04 | omal | my situation: |
| 03:18.43 | omal | i've got asterisk running on a server on my LAN. I can dial into it from a SIP phone on my laptop, or from a traditional analog phone plugged into a sipura-3000 thats registered with the server |
| 03:19.13 | omal | i've got my firewall configured to port forward both the data and control ports for SIP to my asterisk server |
| 03:19.30 | omal | but i can't seem to dial into it from outside my LAN, and nmapping it shows the port as closed |
| 03:20.45 | omal | i thought perhaps it was just my ISP filtering the traffic, but i have the same results running it on a different port |
| 03:21.11 | omal | i doubt they go to the trouble of having some sort of heavy duty packet analyzing firewall in place just to keep me from running my own VOIP |
| 03:21.19 | [TK]D-Fender | omal : What ports precisely and please pastebin your sip.conf |
| 03:21.22 | [TK]D-Fender | ~pb |
| 03:21.26 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 03:22.02 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 03:22.06 | TommyTheKid | ~pb |
| 03:22.13 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 03:22.18 | omal | i have UDP on 5060 going |
| 03:22.31 | omal | and 10000-10100 udp |
| 03:23.37 | [TK]D-Fender | omal : Does 10000-10100 match your rtp.conf? |
| 03:23.58 | omal | doh |
| 03:24.08 | omal | rtpstart=10000rnrtpend=20000 |
| 03:24.40 | [TK]D-Fender | omal : May be important...... also please show us the [general] section of yo sip.conf |
| 03:25.23 | logicwrath | Is sip.conf the only file that affects registering with the voip provider? seems like when I start calling myself over and over to test changes i become unregistered and the peer becomes unreachable |
| 03:25.38 | logicwrath | and im not changing the sip.conf |
| 03:25.56 | *** part/#asterisk terrymr (n=terrymr@209.181.42.92) |
| 03:26.24 | omal | i don't see myself in the asterisk console at all |
| 03:26.36 | omal | when i'm on the LAN, i see myself register, and the calls going through |
| 03:26.39 | omal | outside, nothing |
| 03:27.49 | *** join/#asterisk MacWeenie (n=chatzill@cpe-74-65-228-134.nyc.res.rr.com) |
| 03:28.27 | [TK]D-Fender | omal : Can you please provide what I asked you twice for... |
| 03:28.27 | TommyTheKid | Hello, I had a crash that I think was cased by meetme.. http://pastebin.ca/124535 has the bt.. does anyone have any advice? |
| 03:28.47 | omal | [TK]D-Fender, absolutely, pastebin is lagging on me |
| 03:29.18 | [TK]D-Fender | omal : use .ca |
| 03:29.45 | omal | http://pastebin.ca/124540 |
| 03:29.52 | omal | there we go. yeah .com was slammed |
| 03:30.08 | omal | its practically the stock config |
| 03:30.09 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 03:30.31 | [TK]D-Fender | (possibly) Polycom specific SIP question : On an INVITE it seem they convert # to %23 (hex for #), and * does not seem to like this very much. is there a workaround for this? |
| 03:30.31 | omal | i left out my two entries at the end for registering the two SIP devices |
| 03:31.32 | omal | hmmm, i have the roaming user set to "nat=no" |
| 03:31.33 | [TK]D-Fender | omal : You are missing nat=yes in [general] as well as externip entry |
| 03:31.44 | omal | that makes sense on the lan, but not roaming clearly |
| 03:31.54 | [TK]D-Fender | omal : you should also set canreinvite=on gloablly |
| 03:32.05 | omal | what does that do? |
| 03:32.20 | file | please sir, I want to reinvite |
| 03:32.33 | [TK]D-Fender | omal : It prevents enpoints in your lan from attempting to reconnect direct on a bridged call. |
| 03:32.49 | [TK]D-Fender | file : Care to venture a guess on my previous question? |
| 03:32.55 | *** join/#asterisk tempest1 (n=asf@c-68-58-187-78.hsd1.sc.comcast.net) |
| 03:33.18 | file | [TK]D-Fender: erm, you can try turning on pedantic |
| 03:33.50 | omal | hm, no i do have an externip entry |
| 03:33.54 | omal | but its wrong :D |
| 03:34.35 | [TK]D-Fender | omal : Spelling counts ;) |
| 03:34.53 | omal | spelling, and the fact that i've moved since first setting this up |
| 03:34.54 | [TK]D-Fender | omal : I was waiting for you to realize that or spout out "yeah I did that!" ;) |
| 03:34.59 | omal | new ISP = new ip |
| 03:35.14 | [TK]D-Fender | that too :p |
| 03:35.49 | omal | should setting nat to yes make any difference for local users that aren't natted? |
| 03:35.53 | mog | heh |
| 03:37.11 | [TK]D-Fender | file : Most excellet! Party on! |
| 03:37.38 | [TK]D-Fender | Learned something new today..... |
| 03:38.00 | [TK]D-Fender | Trying to improve my outlook by counting the number of new things I can learn.... |
| 03:38.11 | file | pedantic makes chan_sip more... careful |
| 03:38.17 | omal | oooh, its showing as "filtered" instead of closed. although i'm still not connecting |
| 03:38.28 | [TK]D-Fender | file : I'll take that as "sane". |
| 03:39.07 | file | [TK]D-Fender: sane might not be the right word |
| 03:40.08 | [TK]D-Fender | file : More like "not lazy". I don't think this should necessarily be taken as a negative option. I see no documented downside. know of something? |
| 03:40.23 | file | it makes it more slower, but not a lot |
| 03:40.41 | [TK]D-Fender | What will I do with my newfound ms?!?! Oh noes! |
| 03:41.13 | crochat | Hello ! |
| 03:41.38 | *** join/#asterisk erwinism (n=erwinpog@203.115.172.243) |
| 03:42.15 | crochat | I have problems with the Monitor application ! With the parameter "m", there are still two files : in and out ! Asterisk does not mix anything ! Why ? |
| 03:42.42 | [TK]D-Fender | crochat : Use Mixmonitor instead. |
| 03:42.59 | [TK]D-Fender | crochat : that option you use depends on sox which you might not have. |
| 03:43.02 | erwinism | hello, I have a t1 line, I already have an Asterisk running. What should else should i need in order to accpept 24 simultaneous calls from PSTN ? |
| 03:43.30 | derrick_ | you need top score on centipede |
| 03:44.03 | crochat | [TK]D-Fender: No, I have sox and soxmix installed |
| 03:44.25 | [TK]D-Fender | crochat : Tell you what, try that other app I mentioned. Very effective. |
| 03:44.37 | erwinism | I have a plan to get TE412P. what else should i need? |
| 03:44.54 | [TK]D-Fender | ericsmythe : you already have enough with that 1 card.... |
| 03:45.12 | *** join/#asterisk RF_MIA (n=Administ@68-235-157-35.miamfl.adelphia.net) |
| 03:45.15 | crochat | [TK]D-Fender: But in /etc/passwd file, asterisk's shell is /bin/false ! Could that be a problem ? |
| 03:45.26 | crochat | [TK]D-Fender: Ok, I'll try MixMonitor |
| 03:45.55 | [TK]D-Fender | crochat : No clue. No real experience with * as non-root. can be very tricky. |
| 03:46.20 | erwinism | [TK]D-Fender: Do I need a CSU/DSU to connect my T1 line to TE412P card? |
| 03:47.04 | [TK]D-Fender | ericsmythe : nope. |
| 03:47.19 | [TK]D-Fender | ericsmythe : Striaght Cat5 fromt he smartjack to the card, thats it |
| 03:47.20 | docelmo | hehee |
| 03:47.32 | docelmo | cdu/dsu.. do they even exist anymore? |
| 03:47.39 | [TK]D-Fender | ericsmythe : Digium/Sangoma cards have a CSU onboard. |
| 03:48.09 | derrick_ | crochat no, /bin/false is fine |
| 03:48.17 | [TK]D-Fender | docelmo : Yeah I had a DS0 leased data circuit not too long ago in my company..... used an Adtran for that spitting out V.35 to a Cisco router :) |
| 03:48.31 | crochat | [TK]D-Fender: Thanks a lot ! MixMonitor works fine ;-) |
| 03:48.51 | [TK]D-Fender | crochat : Quite welcome |
| 03:48.55 | *** part/#asterisk RF_MIA (n=Administ@68-235-157-35.miamfl.adelphia.net) |
| 03:49.16 | docelmo | ouch.. I havent used a csu in a LONG time |
| 03:49.19 | docelmo | MIAMI! |
| 03:49.30 | [TK]D-Fender | docelmo : I was paying over 1500$/mo for a 56k DS0. Psychotcally stupid, no? :) |
| 03:49.39 | erwinism | [TK]D-Fender: how about setting up one Telephone number to the callers.. i mean call routing? where can i setup this? |
| 03:49.40 | docelmo | The land where a 2nd language is required.. |
| 03:49.53 | docelmo | Well that was back in the day |
| 03:50.00 | docelmo | I can get a DS3 for 1500 now |
| 03:50.01 | docelmo | :) |
| 03:50.03 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 03:50.26 | [TK]D-Fender | docelmo : No we had cheap DSL and VPN options... head office is manned by a bunch of people living it up Jurassic-style ;) |
| 03:50.39 | docelmo | hqhq |
| 03:50.41 | docelmo | haha |
| 03:50.57 | [TK]D-Fender | docelmo : And worse.... they're NEW-ENGLANDERS ;0 |
| 03:51.03 | docelmo | haha |
| 03:51.12 | docelmo | Im working on Fios for my office in newark |
| 03:51.16 | docelmo | I want the 30/5 |
| 03:53.00 | [TK]D-Fender | docelmo : It took me a long time beating into their head that VPN over DSl would be just fine... they phased it in only because all out e-mail was being pumped through that pipe in addition ot 5250 TE. |
| 03:53.10 | [TK]D-Fender | And was choking the shit out of our performance :) |
| 03:53.29 | [TK]D-Fender | docelmo : So when they brought it in the ONLY thing they did was route e-mail through it! |
| 03:54.01 | docelmo | sigh |
| 03:54.05 | docelmo | what a waste |
| 03:54.17 | erwinism | [TK]D-Fender: do you have any idea how to route calls? because I am only giving one telephone to the callers. |
| 03:54.41 | docelmo | Route calls? |
| 03:55.23 | [TK]D-Fender | docelmo : yeah I told them to wake to ^%$# up and kept sending them analysis sheets showing $1500+/mo savings potential if they got off their dumb asses and jsut went with the flow... |
| 03:55.44 | erwinism | I mean, i have T1, i will give One telephone number to the callers. can the Asterisk handle it so that the number will not always be busy? |
| 03:55.45 | [TK]D-Fender | erwinism : huh? |
| 03:55.52 | *** join/#asterisk michaelo (n=michaelo@adsl-147-45-179.gsp.bellsouth.net) |
| 03:56.31 | CunningPike | erwinism: If you have a PRI, yes - you can handle up to 23 incoming calls at once to the same number |
| 03:56.36 | [TK]D-Fender | erwinism : You mean DID. Your T1 will have likely either 23 or 24 channels depending, and if should pick the first available channel till your circuit is full. |
| 03:57.00 | erwinism | the server is expecting 24 simultaneous calls. And i onlu give one telephone number to the callers |
| 03:57.10 | [TK]D-Fender | erwinism : And since you seem rgey on theat do make very sure of the signalling you are going to have on that line. If its anything but PRI CHANGE IT. |
| 03:57.38 | [TK]D-Fender | erwinism : Server doesn't "expect" anything. calls come in on available channels. Thats it thats all. |
| 03:58.41 | *** join/#asterisk s0lid (n=jlq@124.106.157.190) |
| 03:58.54 | docelmo | haha |
| 04:01.36 | [TK]D-Fender | blarg.. I can't type worth shit tonight.... |
| 04:04.17 | TommyTheKid | hah, in 3 lines of code, I made it so that an admin can do *# (like on ATT concalls) on a app_meetme conference to get the number of users |
| 04:04.26 | TommyTheKid | mostly copied too :) |
| 04:12.45 | erwinism | [TK]D-Fender: thanks |
| 04:19.54 | *** join/#asterisk Trazz (n=traderz@c-67-163-92-37.hsd1.il.comcast.net) |
| 04:22.33 | TommyTheKid | ok i lied, it was 5 lines (with the case and the break;) and I had to put it in twice (once for users and once for admins menu) to make to work for everyone.. the hardest part would be recording the menu prompt again :) |
| 04:25.41 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 04:25.44 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 04:28.38 | [TK]D-Fender | TommyTheKid : I'm sure you could remove all tht unnecessary white-space and do it in 1 line :) |
| 04:29.40 | *** join/#asterisk inventor_ (n=spam@c-24-22-146-195.hsd1.wa.comcast.net) |
| 04:29.57 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-213-102.hsd1.il.comcast.net) |
| 04:30.13 | *** part/#asterisk rephorm (n=rephorm@cpe-24-27-8-18.austin.res.rr.com) |
| 04:30.15 | kuku5 | hey all |
| 04:32.41 | Flauto | hey kuku5 |
| 04:33.06 | kuku5 | Any large changes since 1.2 ? |
| 04:33.22 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 04:36.59 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:37.10 | logicwrath | does asterisk need 69-69 UDP open? |
| 04:37.18 | logicwrath | err 60-69 |
| 04:37.55 | ionix | no |
| 04:38.02 | ionix | 10000-20000 + 5060 |
| 04:38.15 | logicwrath | what do i lose if 10000-20000 is not open? |
| 04:38.23 | ionix | voice |
| 04:38.25 | ionix | not a big deal |
| 04:38.53 | ionix | you can specify the range in rtp.conf |
| 04:39.03 | logicwrath | yea i saw that |
| 04:39.14 | logicwrath | im trying to figure out why i keep losing my broadvoice proxy |
| 04:39.51 | Flauto | how to adjust time under linux? |
| 04:40.09 | Flauto | my linuxbox is one hour ahead of my time |
| 04:40.39 | Flauto | broadvoice sometimes is not stable |
| 04:40.46 | logicwrath | To set the system clock under Linux, use the date command. As an example, to set the current time and date to July 31, 11:16pm, type ``date 07312316'' |
| 04:41.05 | kuku5 | logicwrath: I have the same problem |
| 04:41.21 | logicwrath | •kuku5• are you down right now? |
| 04:41.24 | Flauto | one or a few of their proxies would not work well |
| 04:41.25 | *** join/#asterisk tempest1 (n=asf@c-68-58-187-78.hsd1.sc.comcast.net) |
| 04:41.32 | kuku5 | <PROTECTED> |
| 04:41.36 | logicwrath | same |
| 04:41.44 | kuku5 | its a nat problem |
| 04:42.05 | logicwrath | if i was DMZ would i be ok? |
| 04:42.11 | kuku5 | i dont kwno |
| 04:43.54 | logicwrath | i would switch providers if i knew it was a BV issue |
| 04:44.05 | kuku5 | i dont think it is |
| 04:44.10 | kuku5 | i had it working fine |
| 04:44.10 | Flauto | logicwrath, did you setup hosts for bv proxy? |
| 04:44.14 | kuku5 | then i changed routers |
| 04:44.14 | logicwrath | yes |
| 04:44.15 | De_mon | logicwrath what is qualify set to? |
| 04:44.19 | logicwrath | yes |
| 04:44.26 | logicwrath | qualify=yes |
| 04:44.31 | De_mon | nat=yes? |
| 04:44.37 | Flauto | which proxy you are using |
| 04:44.38 | logicwrath | no nat line |
| 04:44.40 | logicwrath | chicago |
| 04:44.48 | Flauto | are you in chicago? |
| 04:44.56 | logicwrath | im on the chi backbone |
| 04:44.58 | logicwrath | with comcast |
| 04:45.12 | Flauto | i was using teir dc proxy |
| 04:45.16 | logicwrath | i get 15-20 ms pings to chi |
| 04:45.19 | Flauto | that one was more stable |
| 04:45.19 | De_mon | well set nat=yes |
| 04:45.34 | logicwrath | ive tried nat=yes but i still dont register |
| 04:45.36 | Flauto | logic, you should set nat=yes |
| 04:45.38 | logicwrath | i change it and reload |
| 04:45.49 | logicwrath | ive got the ports routed to my asterisk box |
| 04:45.53 | logicwrath | is that considered nat/ |
| 04:45.58 | Flauto | and open ports 10000-20000 |
| 04:46.00 | De_mon | logicwrath do you have a register => line? |
| 04:46.04 | logicwrath | yes |
| 04:46.21 | Flauto | you should be all good |
| 04:46.22 | De_mon | you can do sip debug peer <peer> to see why the registrations arn't working |
| 04:46.30 | Flauto | use dca proxy |
| 04:46.33 | logicwrath | ive tried that but it doesnt make sence to me |
| 04:46.38 | logicwrath | sense |
| 04:46.39 | Flauto | that one is further away but more stable |
| 04:46.40 | De_mon | pastebin.ca |
| 04:46.50 | logicwrath | i can only sip debug peer when registered |
| 04:46.54 | De_mon | mesa going to bed |
| 04:47.08 | De_mon | eh? well to sip debug ip then |
| 04:47.13 | De_mon | s/to/do/ |
| 04:47.13 | logicwrath | 1 sec |
| 04:47.16 | Flauto | good night, de_mon |
| 04:47.43 | Flauto | i used bv for almost a year |
| 04:47.54 | Flauto | i quit on them about half year ago |
| 04:48.13 | Flauto | i now, use a cheap ass service voipstunt |
| 04:50.15 | logicwrath | were you losing registration with BV? |
| 04:50.22 | Flauto | no |
| 04:51.00 | Flauto | but i went through the tough time when they lost some contracting provider and for a while the service was almost not working |
| 04:51.20 | Flauto | i just followed their setting and i had no problem |
| 04:51.24 | [TK]D-Fender | ok, I'm off, back tomorrow |
| 04:51.51 | logicwrath | http://pastebin.ca/124649 |
| 04:52.02 | hads | Grr. mantis needs a date format setting |
| 04:52.49 | Juggie | its php i'm sure it can be edited. |
| 04:53.37 | Flauto | change it to dca |
| 04:53.42 | Flauto | and see what you will get |
| 04:56.15 | *** join/#asterisk s0lid (n=jlq@ded-153-4.eglobalreach.net) |
| 04:56.20 | Flauto | when i used bv, i sometimes had to change proxy from one to another |
| 04:56.35 | Flauto | which plan are you using? |
| 04:56.41 | Flauto | 19.99? |
| 04:57.44 | logicwrath | meh |
| 04:57.48 | logicwrath | i switched and it works now |
| 04:57.54 | logicwrath | no the 9.99 plan |
| 04:57.56 | Flauto | yes |
| 04:58.00 | Flauto | as i was telling you |
| 04:58.07 | Flauto | the dca proxy is the stable one |
| 04:58.21 | Flauto | even though, it is a little bit further away from you |
| 04:58.38 | Flauto | even the nyc and bos are more stable than chi |
| 04:58.43 | Flauto | chi is the unstable one |
| 05:01.00 | logicwrath | ya but im still not able to dial out |
| 05:01.08 | logicwrath | im just registered now |
| 05:01.17 | Flauto | really? |
| 05:01.32 | Flauto | maybe bv is fucked up now |
| 05:01.56 | Flauto | you want to call them to ask them? |
| 05:02.03 | logicwrath | it might be my dial plan |
| 05:02.13 | Flauto | do you have fwd or something? |
| 05:03.01 | logicwrath | i have nothing defined in ext-local right now |
| 05:03.01 | logicwrath | as i stopped using amp and i want to find an example before i rewrite it |
| 05:03.34 | logicwrath | i created my extensions in sip.conf and from-internal |
| 05:04.04 | logicwrath | still need to do something with ext-local before my ring group will work right i think |
| 05:04.28 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 05:04.38 | logicwrath | now its dialing |
| 05:04.43 | logicwrath | i didnt change anything |
| 05:05.03 | logicwrath | im starting to think its a BV issue |
| 05:05.10 | ManxPower | logicwrath, that's the way VoIP works. |
| 05:05.18 | ManxPower | That's why I try to avoid it 8-) |
| 05:05.58 | logicwrath | need some kind of redundant DID system |
| 05:06.03 | logicwrath | like tiered MX records |
| 05:06.12 | logicwrath | 10, 20, 30 |
| 05:11.59 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 05:11.59 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 05:12.03 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 05:12.10 | *** join/#asterisk rene1 (n=rene1@200.93.193.198) |
| 05:12.16 | rene1 | hello |
| 05:12.18 | rene1 | g nite |
| 05:12.34 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-246-132-82.aubnin.fios.verizon.net) |
| 05:12.40 | rene1 | i have been looking at the queues and queue_status asterisk manager actions |
| 05:14.07 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 05:15.08 | rene1 | well they seem the same except that the first is formatted to be read by humans (i.e. in the same way as CLI show queues) and the later is formatted to be parsed by computers (one key,value per line) anyways were Queues show events like (Unavailable, Available) Queue_status shows status (1,5), and then of course there are the status reported via show agents, whew |
| 05:16.58 | hads|home | Anyone here use the pickupexten feature (*8) with SIP phones? I'm getting some weird voice distortion on calls when using this. |
| 05:18.57 | *** join/#asterisk junbug (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 05:19.22 | *** join/#asterisk omal (n=omal@cpe-24-164-111-184.neo.res.rr.com) |
| 05:19.24 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 05:19.31 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 05:19.43 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 05:19.43 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:20.33 | *** join/#asterisk apardo (n=apardo@87.217.144.160) |
| 05:20.35 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 05:21.01 | *** join/#asterisk budmang (n=budman@12.206.134.162) |
| 05:22.12 | rene1 | question: can apps like app_machine_detect or NV machine detect app be used to reliably detect calls answered by PBX systems? |
| 05:22.55 | rene1 | and second: what do people that are doing auto dialing do when they encounter a PBX. does the agent navigates tru the IVR and try to reach the contact? |
| 05:23.44 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:28.46 | erwinism | What is the good computer specification to handle a T1 line accepting 24 calls at the same time not included the IP based clients? |
| 05:29.15 | rene1 | t1 line is 23 calls |
| 05:29.37 | erwinism | rene, sorry 23 youre correct |
| 05:29.37 | rene1 | it depends, will you be doing recording? and if so with compression? |
| 05:29.50 | erwinism | rene yes, recording is a must |
| 05:30.15 | rene1 | you could get a way with a P4 3 Ghz 1 G RAM |
| 05:30.21 | erwinism | but i have another linux box to handle the data files |
| 05:30.21 | Qwell | isn't a T1 24 channels? a PRI is 23 + D |
| 05:30.37 | JT | T1 PRI |
| 05:30.40 | Qwell | I'm fairly certain |
| 05:30.45 | erwinism | I will be requestion 23B + 1D |
| 05:30.48 | JT | can't say PRI without saying the line type |
| 05:30.49 | rene1 | Qwell you are right |
| 05:30.54 | erwinism | I will be requesting 23B + 1D |
| 05:30.59 | JT | it could be an E1 PRI |
| 05:31.07 | erwinism | yes :) |
| 05:31.17 | Qwell | JT: That becomes just semantics |
| 05:31.25 | Qwell | but, saying that a T1 only has 23 channels is incorrect |
| 05:31.27 | rene1 | can run 24 calls on a PRI t1 tho |
| 05:31.32 | rene1 | cant |
| 05:31.40 | Qwell | rene1: sure you can, with something like NFAS |
| 05:31.52 | JT | Qwell: saying it has 23 traffic channels when used in CCS mode is correct though |
| 05:31.58 | rene1 | with a single T1? |
| 05:32.02 | JT | you can with CAS |
| 05:32.07 | rene1 | JT is right |
| 05:32.22 | Qwell | <rene1> t1 line is 23 calls |
| 05:32.27 | Qwell | That, unqualified, is incorrect :) |
| 05:32.43 | erwinism | rene, so P4 3GHZ and 1GB of ram will be enough for my Asterisk to handle right? |
| 05:32.43 | rene1 | it was correct in the context |
| 05:32.44 | Qwell | it's important to specify exactly what you mean |
| 05:32.55 | rene1 | yes erwinism |
| 05:32.56 | budmang | Qwell and his BS :-) |
| 05:33.16 | rene1 | it will do the job nicely just dont buy an el cheapo system |
| 05:33.40 | JT | i'd say it's better to get a slightly slower machine IF it means more server grade hardware |
| 05:33.49 | erwinism | rene thanks.. hehehe i already told my boss i need opteron server which is overkill. hehe |
| 05:33.52 | JT | eg redundant power supplies and RAID1 or 5 |
| 05:34.04 | erwinism | i will cancel the request |
| 05:34.21 | SkramX | what needs to be done after installing astetisk-addons? I need the mysql application... do i need to recompile the main asterisk code? |
| 05:34.24 | JT | if you can pull it off, faster is just great |
| 05:34.29 | JT | but make sure it's server grade |
| 05:34.59 | Un1x | erwinism dont cancel send the opteron server to me :D |
| 05:35.07 | rene1 | heheh |
| 05:35.13 | Un1x | btw are you concerned about bandwidth, and upload usage |
| 05:35.19 | erwinism | lol Un1x :) |
| 05:36.10 | rene1 | Qwell: remember your suggestion of using astmanproxy? |
| 05:36.13 | rene1 | it has helped me a lot |
| 05:36.22 | rene1 | i even ended up using two of them LOL |
| 05:36.25 | SkramX | ? |
| 05:36.25 | rene1 | in the same box |
| 05:36.37 | rene1 | i needed both http and standard i/o |
| 05:36.46 | budmang | why would my que all of a sudden just start saying "later" |
| 05:36.47 | erwinism | we already have good bandwidth here. I am requesting a good codec thou.. if the company will accept then it wpuld be great. I am still looking forward to implement speex |
| 05:36.58 | rene1 | it is very cool and it eats very little resourcs |
| 05:37.22 | rene1 | speex is cool it just that there aint not much gear that talks speex |
| 05:37.24 | Un1x | erwinism is the company big and make more then excess of 1million dollars in profit if so send me that server lol |
| 05:37.31 | Un1x | i think it's about time i got something back from companys |
| 05:37.36 | Un1x | they always take money and never give :( |
| 05:38.09 | SkramX | ? |
| 05:38.27 | rene1 | servers are made by companies, if you send them a check they might send you a server, you just need good credit |
| 05:38.31 | rene1 | jk |
| 05:38.42 | rene1 | SkramX: after compiling mysql addons |
| 05:38.45 | erwinism | lol un1x, i dont have any idea on the revenue, i just do the implementation of making this voice blogging/podcasting project |
| 05:38.48 | budmang | why would my que all of a sudden just start saying "later" |
| 05:38.51 | rene1 | you need to configure res_confi_mysql |
| 05:38.51 | SkramX | rene1: what about it |
| 05:38.55 | SkramX | oh |
| 05:38.58 | rene1 | yep |
| 05:38.59 | SkramX | forgot abou tthat part |
| 05:39.02 | SkramX | in? |
| 05:39.05 | SkramX | oh, ill find it |
| 05:39.18 | rene1 | in '/etc/asterisk' |
| 05:39.22 | Un1x | SkramX: do you need to type 3 word lines? |
| 05:39.23 | rene1 | you did issue make install? |
| 05:39.34 | SkramX | rene1: Indee |
| 05:39.35 | SkramX | +d |
| 05:39.37 | *** part/#asterisk w32 (n=w32@c-71-193-124-77.hsd1.il.comcast.net) |
| 05:39.42 | SkramX | damn, Sorry about that; |
| 05:39.50 | rene1 | budmang: saying later? |
| 05:39.56 | rene1 | what do u mean |
| 05:40.19 | Un1x | Anyway, i'll be back on in a bit going to go play some xbox and mayube get something to drink and then come back up, cya guy's in 5 minutes or so.. |
| 05:40.22 | SkramX | rene1: in modules.conf? |
| 05:40.38 | SkramX | or should there be a res_mysql.conf? |
| 05:40.49 | rene1 | SkramX: res_mysql.conf |
| 05:40.53 | *** join/#asterisk tempest1 (n=asf@c-68-58-187-78.hsd1.sc.comcast.net) |
| 05:40.53 | SkramX | I see none |
| 05:40.56 | SkramX | just a res_odbc.conf |
| 05:41.01 | rene1 | you may need to copy from your build directory by hand |
| 05:41.26 | rene1 | look into your asterisk-addons build directory |
| 05:41.36 | rene1 | it surely is there and copy it to /etc/asterisk |
| 05:41.43 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 05:41.49 | SkramX | rene1: will look |
| 05:41.54 | rene1 | then from within the CLI do |
| 05:42.06 | rene1 | realtime mysql status or mysql realtime status |
| 05:42.15 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 05:42.20 | rene1 | and that will show you if asterisk connected to the DB |
| 05:42.56 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 05:44.08 | rene1 | the format for queue_log is weird |
| 05:44.20 | rene1 | cant really make much sense of it |
| 05:45.38 | SkramX | rene1: will try all that shortly |
| 05:46.20 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 05:49.02 | SkramX | rene1: so.. i do make |
| 05:49.04 | SkramX | make install.. |
| 05:49.25 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 05:51.08 | rene1 | i have to fo |
| 05:51.09 | rene1 | go |
| 05:51.14 | SkramX | peace out |
| 05:51.35 | rene1 | make install and if that doesnt land you a res_mysql file in /etc/asterik cooy by hand |
| 05:51.38 | rene1 | copy* |
| 05:51.41 | rene1 | bye then |
| 05:58.11 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 05:58.38 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 06:02.13 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
| 06:03.06 | dos000 | anyone know what diff params they used to get this patch output http://bugs.digium.com/file_download.php?file_id=8805&type=bug |
| 06:03.17 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-42-80.cybersurf.com) |
| 06:03.29 | Qwell | dos000: svn diff |
| 06:03.49 | SkramX | bah |
| 06:03.57 | dos000 | tow ... there is no possible way to get the same output from diff standalone ? |
| 06:04.12 | Qwell | -u |
| 06:05.05 | dos000 | Qwell, i tried -urNX ./diff.exclude src dst .. it still list binary files |
| 06:05.25 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:07.53 | dos000 | Qwell, btw i think there should be a diff.exclude file for all the files that asterisk generates .. to make life easier on people doing patches .. is it already there |
| 06:08.23 | Qwell | dos000: no, but one could easily make one from the svn:exclude property |
| 06:08.29 | Qwell | erm, svm:ignore |
| 06:08.32 | Qwell | bah, svn:ignore |
| 06:10.49 | *** join/#asterisk digime (n=digime@70.230.196.197) |
| 06:11.02 | digime | has anyone here used "request tracker" ticketing software? |
| 06:12.37 | *** join/#asterisk jasloan (n=JSLOAN2@c-24-131-124-92.hsd1.oh.comcast.net) |
| 06:13.48 | *** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au) |
| 06:13.51 | jasloan | Anyone around that can help me troubleshoot a macro? |
| 06:14.46 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 06:14.49 | jasloan | It doesn't run the script if the caller hangs up from voicemail. http://pastebin.ca/123371 |
| 06:15.47 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 06:20.12 | *** join/#asterisk BugKham (i=BugKham@202.8.86.164) |
| 06:20.43 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 06:21.19 | BugKham | anyone running asterisk on Pentium D with x86_64? |
| 06:21.48 | BugKham | asterisk reload very slowly on mine |
| 06:26.04 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 06:29.45 | SkramX | do you do agi(script.agi,${variable}) |
| 06:29.47 | SkramX | is that it? |
| 06:34.03 | *** join/#asterisk daysmen3 (n=primus@host86-138-239-164.range86-138.btcentralplus.com) |
| 06:35.01 | jasloan | SkramX: is that to me? |
| 06:35.29 | SkramX | anyone |
| 06:38.04 | jasloan | i don't think you use the comma |
| 06:38.32 | jasloan | agi(script.agi ${ARG1}) |
| 06:38.55 | Qwell | Yes you do |
| 06:39.20 | SkramX | plau |
| 06:39.23 | SkramX | weird |
| 06:39.35 | SkramX | no comma between arg.. or is a comma okay? |
| 06:39.44 | Qwell | | is better |
| 06:39.56 | mog | omg its qwell |
| 06:39.58 | SkramX | exten => 3,1,AGI(mysql_blacklist.agi|${number}) |
| 06:40.03 | Qwell | omg mog! |
| 06:40.04 | SkramX | ${number} is set |
| 06:40.06 | mog | they are interpretted same |
| 06:40.11 | mog | Qwell, why you up so late |
| 06:40.13 | Qwell | SkramX: agi debug? |
| 06:40.16 | Qwell | mog: It's only 11:30 |
| 06:40.21 | SkramX | Qwell: okie dokies |
| 06:40.41 | mog | heh 140 here |
| 06:42.10 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 06:42.24 | SkramX | same here |
| 06:42.27 | *** join/#asterisk my007ms (n=noor@217.139.224.194) |
| 06:42.29 | SkramX | <-- Austin, TX |
| 06:43.08 | SkramX | <PROTECTED> |
| 06:43.15 | SkramX | didnt work though |
| 06:43.22 | SkramX | so... something inside the script? |
| 06:43.29 | SkramX | if i echo in it.. will it print to the console? |
| 06:44.03 | Qwell | mog: 2 days ;) |
| 06:44.08 | mog | word |
| 06:44.11 | mog | call in sick |
| 06:44.13 | mog | start early |
| 06:44.14 | Qwell | heh |
| 06:44.15 | SkramX | Qwell: so...? |
| 06:44.24 | L|NUX | mog : O_o |
| 06:44.30 | L|NUX | 11:44 AM here |
| 06:44.31 | Qwell | mog: I'm going to say my goodbyes at work tomorrow, at my old building |
| 06:44.37 | Qwell | then Friday is smooth sailing |
| 06:44.41 | mog | nice |
| 06:46.51 | SkramX | could i get a little help with agi? |
| 06:46.59 | SkramX | will the variable be shown in agi debug? |
| 06:48.45 | *** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au) |
| 06:50.42 | mog | still one more bug with this stuff |
| 06:50.49 | mog | and i need a machine for the test set up |
| 06:52.06 | threat2 | whats up mog ? |
| 06:52.21 | mog | eh stupid sla stuff |
| 06:52.38 | threat2 | no good |
| 06:52.43 | mog | you? |
| 06:53.03 | SkramX | :( |
| 06:53.49 | threat2 | Attemping to run windows XP within qemu under Linux :) |
| 06:53.54 | mog | ew |
| 06:54.00 | mog | reason? |
| 06:54.05 | threat2 | competely asterisks unrelated I know :P |
| 06:54.43 | threat2 | I need to use Visual Studio for a uni project, I refuse to install XP onto any of my hard drives so I am installing it in an image file |
| 06:55.01 | mog | lol |
| 06:55.11 | SkramX | how do i see the variables of the agi? |
| 06:55.16 | mog | i had similar problems in some of my cs projects |
| 06:55.23 | mog | i convinced teacher to let me use make though |
| 06:55.47 | threat2 | yes I "could" you wine, but I find that unstable / reliable compared to a CPU emulator type program |
| 06:55.50 | threat2 | mog, haha |
| 06:56.06 | threat2 | your teacher needed to be convinced for you to use make? wtf |
| 06:56.42 | *** join/#asterisk tempest1 (n=asf@c-68-58-187-78.hsd1.sc.comcast.net) |
| 06:56.43 | threat2 | what was your teachers experties (Icant spell)? VB ? |
| 06:56.50 | mog | lol |
| 06:56.59 | mog | heh some of em are vb heads |
| 06:57.03 | mog | thinking vb is future |
| 06:57.15 | threat2 | mog, did they insist you have a shit load of goto lines in your code? |
| 06:57.21 | mog | nah |
| 06:57.28 | mog | school is anti goto |
| 06:57.35 | threat2 | same at my uni |
| 06:57.50 | threat2 | but you should see some VB code :S |
| 06:58.45 | mog | i dont look at vb code |
| 06:59.33 | threat2 | heh, you point and click? or just stay completely away from it ? |
| 07:00.00 | mog | i try to stay away from windows |
| 07:00.05 | threat2 | good man |
| 07:00.19 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 07:00.21 | threat2 | and / or women |
| 07:00.52 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 07:00.59 | L|NUX | mog : still at work ? |
| 07:00.59 | L|NUX | :P |
| 07:01.15 | mog | yeah |
| 07:01.22 | mog | trying to think of way to fix this code properly |
| 07:01.51 | Snake-Eyes | Any one ever used softhangup? |
| 07:02.12 | mog | yes |
| 07:02.21 | Shaun2222 | anybody know where the correct table scheme for the mysql addon is for extensions? |
| 07:02.23 | *** join/#asterisk jhamlyn (i=jhamlyn@203.33.186.65) |
| 07:02.29 | jhamlyn | :-) |
| 07:02.31 | mog | voip-info does Shaun2222 |
| 07:02.36 | mog | i dont remember sorry |
| 07:02.41 | *** join/#asterisk tempest1 (n=asf@c-68-58-187-78.hsd1.sc.comcast.net) |
| 07:03.31 | Shaun2222 | mog: voip-info has about 10 versions |
| 07:03.32 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 07:03.49 | Snake-Eyes | mog, so come when I use SoftHangup(SIP/trunk-1) or SoftHangup(SIP/trunk|a) in a macro it doesnt hang up the other active call on the trunk ? |
| 07:04.12 | mog | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 07:04.19 | mog | three second look up Shaun2222 ..... |
| 07:04.48 | mog | a hangs up all sub channels for that single channel i believe |
| 07:04.53 | SkramX | in an agi.. how do i send txt to stderr so it shows up in the asterisk console |
| 07:06.09 | Snake-Eyes | mog, sub channels ? ? |
| 07:06.09 | Assid | SkramX: just use noop |
| 07:06.13 | SkramX | hmm |
| 07:06.36 | SkramX | i just need to insert someting into a db |
| 07:06.37 | Snake-Eyes | mog, you can only have one call per channel, right? |
| 07:06.41 | SkramX | and mysql() doesnt want to compile |
| 07:07.26 | mog | nope |
| 07:07.36 | mog | er one per channel but not per device |
| 07:07.59 | mog | so it hangs up all sip/myprovider |
| 07:08.02 | mog | not all sip |
| 07:08.19 | mog | or it should i think |
| 07:09.24 | mog | i wonder if i should take a nap |
| 07:09.24 | Qwell | mog: redbull |
| 07:09.24 | mog | none on this side of office |
| 07:09.24 | mog | and i dont have key to other side |
| 07:09.24 | Qwell | You're still in the office? |
| 07:09.24 | Qwell | dude, go home ;/ |
| 07:09.26 | Snake-Eyes | mog no you cant i still have questions :P |
| 07:09.27 | mog | yeah |
| 07:10.22 | Snake-Eyes | mog, so if 1 call is using a trunk and the 2nd call uses softhangup shouldn't the 1st call be hangup ? |
| 07:11.27 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 07:11.29 | Snake-Eyes | crap he's dropped dead already |
| 07:11.49 | Snake-Eyes | :P |
| 07:12.37 | SkramX | arg i really need help |
| 07:13.03 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 07:13.41 | SkramX | http://pastebin.ca/raw/124776 |
| 07:14.44 | Shaun2222 | how can i set a preferred codec for a iax user? |
| 07:14.54 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 07:15.37 | Assid | same way you set it for a sip user |
| 07:15.38 | Shaun2222 | hmm, actually looks like it goes in order of the allows |
| 07:16.40 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:17.10 | *** join/#asterisk theStargazer (n=stargr@ppp147-137.adsl.forthnet.gr) |
| 07:21.11 | *** join/#asterisk xxoxx (n=xxoxxx@tor/regular/xxoxx) |
| 07:21.31 | Snake-Eyes | Shaun2222, are you using g729 ? |
| 07:25.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:27.02 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:27.04 | Chris-NB | hi |
| 07:28.40 | Chris-NB | anyone tried out high availability and loadsharing with only two boxes ? |
| 07:28.49 | Chris-NB | linux HA and ultramonkey |
| 07:32.18 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 07:40.33 | *** join/#asterisk bmg505 (n=leon@c1-44-3.rndf.isadsl.co.za) |
| 07:42.20 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:42.34 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 07:43.09 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 07:48.55 | DarKnesS_WolF | Aug 10 10:45:08 NOTICE[14218]: chan_zap.c:4661 zt_read: Fax detected, but no fax extension any idea ? |
| 07:49.37 | x86 | you need a "fax" extension |
| 07:49.59 | x86 | exten => fax,s,1,Goto(recieve-fax|s|1) |
| 07:50.41 | DarKnesS_WolF | ahh not s? |
| 07:50.42 | DarKnesS_WolF | thx |
| 07:50.49 | x86 | Well, you dont have to create an entire context for it, but it's advisable :) |
| 07:51.06 | x86 | [recieve-fax] |
| 07:51.15 | x86 | s,1,RxFax() |
| 07:51.20 | x86 | h,1,Hangup |
| 07:51.47 | DarKnesS_WolF | x86: i have done that |
| 07:52.16 | *** join/#asterisk suma (n=kans@61.14.86.23) |
| 07:52.17 | DarKnesS_WolF | [fax-in] |
| 07:52.17 | DarKnesS_WolF | exten => fax,1,Set(FAXFILE=/var/spool/astersik/fax/i${CALLERIDNUM}-${TIMESTAMP}.tif) |
| 07:52.20 | DarKnesS_WolF | exten => fax,n,rxfax(${FAXFILE}) |
| 07:52.31 | x86 | looks good |
| 07:52.39 | suma | is T.38 supported in asterisk ? |
| 07:52.46 | x86 | suma: chan_t38 |
| 07:52.53 | x86 | suma: it's "beta" though |
| 07:52.59 | suma | x86: thanks, i c |
| 07:54.24 | mog | still breathing out here |
| 07:54.33 | mog | t38 pass through is support in trunk |
| 07:54.55 | Chris-NB | noone tried ultramonkey out ? |
| 07:56.33 | DarKnesS_WolF | x86: Aug 10 10:54:57 NOTICE[14555]: chan_zap.c:6057 ss_thread: Got event 18 (Ring Begin)... |
| 07:56.35 | DarKnesS_WolF | Aug 10 10:54:58 NOTICE[14555]: chan_zap.c:6057 ss_thread: Got event 2 (Ring/Answered)... == Starting Zap/2-1 at fax-in,s,1 failed so falling back to exten 's' == Starting Zap/2-1 at fax-in,s,1 still failed so falling back to context 'default' |
| 07:56.39 | DarKnesS_WolF | Aug 10 10:54:58 WARNING[14555]: pbx.c:2357 __ast_pbx_run: Channel 'Zap/2-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 07:56.52 | DarKnesS_WolF | :-s |
| 07:57.08 | hads|home | Create a dialplan. |
| 07:57.20 | DarKnesS_WolF | i have dialplan |
| 07:58.39 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 07:58.52 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 08:00.08 | Snake-Eyes | hmm so softhangup wont hangup another call on a trunk ? |
| 08:00.59 | DarKnesS_WolF | x86: what is wrong about that |
| 08:03.07 | hads|home | DarKnesS_WolF: Look at the messages, Zap/2 is initially going into the fax-in context, extension s, which doesn't exist. So Asterisk then tries extension s in the default context, which doesn't exist either. |
| 08:03.29 | DarKnesS_WolF | yes i created s in fax-in |
| 08:03.30 | DarKnesS_WolF | testing now |
| 08:04.51 | hads|home | Messages are good like that, they tell you what's going on. |
| 08:05.10 | x86 | DarKnesS_WolF: [fax-in] exten => s,1,Set() exten => s,n,rxfax() |
| 08:06.03 | DarKnesS_WolF | x86: u mean fax,s,set()? |
| 08:06.26 | DarKnesS_WolF | x86: that what i have in the1st and i got that fax extion dosn't existes |
| 08:06.51 | DarKnesS_WolF | ahh i got it ! |
| 08:08.11 | x86 | exten => fax,1,Goto(fax-in|s|1); [fax-in]; exten => s,1,Set(); exten => s,n,RxFax(); |
| 08:08.32 | DarKnesS_WolF | yep done that ;-) |
| 08:08.47 | DarKnesS_WolF | exten => fax,s,goto(fax-in|S|1) |
| 08:08.52 | DarKnesS_WolF | i didnt read this when u posted it |
| 08:08.53 | DarKnesS_WolF | sorry :P |
| 08:09.49 | DarKnesS_WolF | x86: i'll put them all in [fax-in] |
| 08:09.56 | x86 | DarKnesS_WolF: dumbass |
| 08:09.58 | x86 | ;) |
| 08:09.58 | DarKnesS_WolF | [fax-in] |
| 08:09.59 | DarKnesS_WolF | exten => fax,s,Goto(fax-in|s|1) |
| 08:09.59 | DarKnesS_WolF | exten => s,1,Set(FAXFILE=/var/spool/astersik/fax/i${CALLERIDNUM}-${TIMESTAMP}.tif) |
| 08:10.01 | DarKnesS_WolF | exten => s,n,rxfax(${FAXFILE}) |
| 08:10.01 | DarKnesS_WolF | x86: why ? |
| 08:10.09 | x86 | that wont work |
| 08:10.14 | x86 | think about the logic |
| 08:10.24 | x86 | tell me when you see the recursive loop there ;) |
| 08:10.31 | x86 | hmm |
| 08:10.32 | DarKnesS_WolF | lol |
| 08:10.34 | DarKnesS_WolF | correct ! |
| 08:10.50 | x86 | no recursive loop, but still it's wrong and dirty, and you should go jump off a bridge now |
| 08:10.55 | x86 | ;) |
| 08:11.14 | DarKnesS_WolF | lol |
| 08:11.15 | DarKnesS_WolF | ok :P |
| 08:11.19 | DarKnesS_WolF | i'll put it in default :D? |
| 08:11.48 | x86 | exten => fax,1,Goto(fax-in|s|1) |
| 08:11.52 | x86 | put that in default |
| 08:12.03 | DarKnesS_WolF | done :D |
| 08:12.05 | DarKnesS_WolF | testing :P |
| 08:12.06 | DarKnesS_WolF | lol |
| 08:13.27 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 08:15.33 | erwinism | is codec g729, $10 per channel? am i right? |
| 08:15.49 | DarKnesS_WolF | x86: same error :( |
| 08:17.02 | DarKnesS_WolF | [fax-r] |
| 08:17.02 | DarKnesS_WolF | exten => fax,s,Goto(fax-in|s|1) |
| 08:17.11 | DarKnesS_WolF | [fax-in] |
| 08:17.11 | DarKnesS_WolF | exten => s,1,Set(FAXFILE=/var/spool/astersik/fax/i${CALLERIDNUM}-${TIMESTAMP}.tif) |
| 08:17.14 | DarKnesS_WolF | exten => s,n,rxfax(${FAXFILE}) |
| 08:17.31 | DarKnesS_WolF | x86: and in zaptel this zap2-1 is context=fax-r |
| 08:17.40 | x86 | DOOD |
| 08:17.56 | x86 | PUT YOUR FUCKING exten => fax,1,Goto() IN YOUR DEFAULT CONTEXT |
| 08:18.01 | x86 | and use 1 not s! |
| 08:20.16 | DarKnesS_WolF | don't hate me :P |
| 08:20.32 | erwinism | haha |
| 08:21.27 | DarKnesS_WolF | x86: oky done that will test now if didn't work i'll kick u in the nuts :P |
| 08:21.38 | erwinism | DarKnesS_WolF: x86 has a good humor. |
| 08:21.38 | x86 | i think your turbin is too tight bro ;) |
| 08:21.56 | DarKnesS_WolF | turbin ? |
| 08:22.01 | DarKnesS_WolF | what doe turbin means? |
| 08:22.59 | x86 | towels worn on heads ;) |
| 08:23.29 | L|NUX | x86 : O_o |
| 08:24.20 | SkramX | night |
| 08:24.24 | x86 | night |
| 08:24.39 | DarKnesS_WolF | night |
| 08:24.59 | *** join/#asterisk daysmen3 (n=primus@host86-138-239-164.range86-138.btcentralplus.com) |
| 08:25.09 | DarKnesS_WolF | x86: will test it now |
| 08:25.44 | x86 | heh |
| 08:26.05 | DarKnesS_WolF | x86: see now all astersik gurus kicking me :P |
| 08:26.06 | erwinism | haha |
| 08:26.38 | DarKnesS_WolF | [fax-in] |
| 08:26.38 | DarKnesS_WolF | exten => s,1,Set(FAXFILE=/var/spool/astersik/fax/i${CALLERIDNUM}-${TIMESTAMP}.tif) |
| 08:26.41 | DarKnesS_WolF | exten => s,n,rxfax(${FAXFILE}) |
| 08:26.47 | DarKnesS_WolF | [default] |
| 08:26.47 | DarKnesS_WolF | exten => fax,1,Goto(fax-in|s|1) |
| 08:26.49 | L|NUX | DarKnesS_WolF : me not guru |
| 08:26.55 | L|NUX | DarKnesS_WolF : i am just n00b like you are |
| 08:27.00 | x86 | DarKnesS_WolF: looks good |
| 08:27.00 | DarKnesS_WolF | x86: correct ? |
| 08:27.05 | DarKnesS_WolF | ok th |
| 08:27.05 | DarKnesS_WolF | x |
| 08:27.51 | L|NUX | DarKnesS_WolF : ALLHUMDULLILLAH |
| 08:27.52 | L|NUX | ;) |
| 08:27.56 | L|NUX | DarKnesS_WolF : you understand that |
| 08:29.09 | DarKnesS_WolF | L|NUX: what do u think :P |
| 08:29.23 | L|NUX | DarKnesS_WolF : about |
| 08:29.26 | L|NUX | ^_^ |
| 08:29.32 | DarKnesS_WolF | yepp understand that |
| 08:29.42 | L|NUX | thanks to ALLAH |
| 08:30.10 | *** join/#asterisk ken___ (n=ken@125.212.103.40) |
| 08:31.20 | ken___ | quick question, i'm running 2.6.17-2-686-smp and i'm trying to compile & install zaptel -- compilation happens just fine, however when i try to go modprobe zaptel or modprobe wctdm i get a FATAL: module zaptel not found ... anyone know what this is off the top of their heads? |
| 08:31.57 | ken___ | the modules are being compiled, and are being installed in /lib/modules/2.6.16/ instead of /lib/modules/2.6.16-2-686-smp/ |
| 08:32.14 | DarKnesS_WolF | x86: Aug 10 11:30:32 NOTICE[16826]: chan_zap.c:6057 ss_thread: Got event 18 (Ring Begin)... |
| 08:32.17 | DarKnesS_WolF | <PROTECTED> |
| 08:32.19 | DarKnesS_WolF | <PROTECTED> |
| 08:32.22 | DarKnesS_WolF | Aug 10 11:30:32 WARNING[16826]: pbx.c:2357 __ast_pbx_run: Channel 'Zap/2-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 08:32.25 | DarKnesS_WolF | <PROTECTED> |
| 08:34.43 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 08:34.46 | Nobbie | heya =) |
| 08:36.00 | Nobbie | i'm having a problem getting my PRI/Digium/Zaptel setup to work properly. i have the channels defined, and inbound calls work correctly, but outbout calls don't work. The PRI Debug shows these Message Types: > SETUP < CALL PROCEEDING < DISCONNECT. any suggestions ? |
| 08:37.14 | *** join/#asterisk henk (n=marius@s5593c2e9.adsl.wanadoo.nl) |
| 08:37.31 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 08:37.46 | *** part/#asterisk Rawplayer (n=kevin@braadharing.oom-killer.org) |
| 08:37.51 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 08:38.02 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 08:38.10 | henk | Hi, what do i need to do to set up sms using asterisk and sipdiscount.com ? I see they offer sms now and i was wondering if it is possible to get that working from asterisk |
| 08:39.03 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:40.17 | ken___ | henk: why pay for sms? all sms providers provide email -> sms gateways ... just write an AGI script to handle it |
| 08:40.36 | ken___ | er, of course, that didn't handle your question |
| 08:41.33 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 08:41.49 | henk | ken no it did not but i'm interested. how would i set that up i've not heard of free email2sms that is capable of sending sms to 'unregistered' phones. |
| 08:42.11 | *** join/#asterisk apardo (n=apardo@87.217.146.205) |
| 08:42.17 | ken___ | henk: nawh, it's really simple, just get an NPA NXX database, and use that for the providers |
| 08:42.30 | ken___ | henk: you won't get 100% sms coverage, but you'll get like.. 95% coverage |
| 08:42.54 | ken___ | henk: so you write a script that looks up in the database for who owns the phone number you're trying to sms to |
| 08:43.00 | *** part/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 08:43.25 | ken___ | henk: then you just search in that for AT&T or Sprint or Rogers CA or whatever, and then have a mapping table for the owner |
| 08:44.02 | ken___ | henk: so you know, you just then send a text message to the <phonenumber>@messaging.sprintpcs.com or whatever |
| 08:44.29 | ken___ | er, not a text message, an email message |
| 08:44.34 | ken___ | works really well |
| 08:44.44 | ken___ | it's how i have my servers alert me if they go down or whatever |
| 08:44.52 | ken___ | i also use it for some sms list management stuff |
| 08:44.56 | ken___ | really easy shit |
| 08:45.41 | *** join/#asterisk Xen^ (n=linux@202.5.145.56) |
| 08:46.43 | henk | Ok but still the person that you are trying to reach must have an telefphone operator that offers that for all their customers (here in holland that is a service you need to dign up on) and you apprarently can get userinformation on a cellphone from the number? wow that kinda kills your privacy... I think that cannot be done here either |
| 08:47.01 | ken___ | henk: no ... i'm in the states |
| 08:47.07 | ken___ | what's one of those cut & paste places ? |
| 08:47.20 | ken___ | i'll drop a bunch of code off on you, so you can look at what i have |
| 08:47.43 | henk | bastebin you mean? |
| 08:47.57 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 08:47.58 | ken___ | yeah |
| 08:48.15 | henk | pastebin.ca is ok |
| 08:48.24 | henk | .com is too damn slow |
| 08:49.04 | xnon | hello friends my problem with codecs is fixed but the quality of calls is so bad! |
| 08:49.22 | xnon | im using g711 |
| 08:49.34 | websae | try g729 |
| 08:49.46 | xnon | with g729 or g723 or g729 is so much bether? |
| 08:50.03 | xnon | ok but g729 is a pay codec isnt it? |
| 08:50.56 | websae | not if you don't need to do transcoding |
| 08:51.15 | websae | as long as your endpoints (phones) and service provider support it |
| 08:53.09 | ken___ | dude |
| 08:53.13 | ken___ | can't compile zaptel correctly |
| 08:53.15 | ken___ | this is killing me |
| 08:53.26 | ken___ | i've been using asterisk for years now, and i can't figure this out ... horrible ! |
| 08:54.25 | henk | ken___: thanx |
| 08:54.29 | xnon | websae, is free for develop? |
| 08:54.43 | ken___ | henk: that stuff probably won't do you any good in europe tho |
| 08:54.59 | xnon | websae, g729 is free for develop? i need for a personal use |
| 08:55.07 | mog | whats wrong ken___ |
| 08:55.33 | xnon | i have a others probs with my asterisk! |
| 08:55.48 | xnon | i have a errors in my asterisk shell |
| 08:55.59 | xnon | about dns asterisk |
| 08:56.07 | ken___ | mog: i'm running kernel 2.6.16-2-686-smp when i do make linux26 && make install on zaptel, then a modprobe zaptel i get a "invalid format" message from insmod |
| 08:56.19 | xnon | for example: [unixODBC][Driver Manager]Data source name not found, andno default driver specified |
| 08:56.21 | mog | do a dmesg and pastebin it |
| 08:56.49 | xnon | ok |
| 08:57.05 | ken___ | zaptel: version magic '2.6.16 SMP 686 gcc-4.1' should be '2.6.16-2-686-smp SMP 686 gcc-4.0' |
| 08:57.07 | ken___ | oh |
| 08:57.08 | ken___ | haha |
| 08:57.12 | ken___ | la la la |
| 08:57.15 | ken___ | fucking debian |
| 08:57.24 | ken___ | ok, got the problem ! |
| 08:57.29 | mog | ^_^ |
| 08:57.32 | mog | no problem |
| 08:57.34 | mog | happy to help |
| 08:58.07 | Nobbie | *argh* i need PRI help =( |
| 08:58.46 | ken___ | mog: will doing CC=gcc-4.0 work on make linux26 ? |
| 08:59.11 | mog | or just change the sim link |
| 08:59.59 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 09:00.07 | henk | ken___: as far as i can tell european providers do not do that kind of stuff |
| 09:00.22 | ken___ | henk: heh ... well, complain ! |
| 09:01.07 | henk | ken___: it wont help. here sms makes them far too much money i think we are one of the most expensive countries for sms in the world |
| 09:01.24 | henk | not enough competition i guess |
| 09:01.27 | xnon | woao i cant send the i was paste in pastebin.ca :S |
| 09:08.57 | ken___ | henk: what kind of volumes of sms messages are you talking about sending ? |
| 09:09.27 | henk | no much, couple of hundred a month maybe |
| 09:09.48 | henk | sipdiscount.com is working pretty well for just 5 cents |
| 09:09.50 | ken___ | henk: ok, so ... you should be able to do that via a script and a bluetooth cellphone or a usb cellphone |
| 09:10.48 | henk | na too much hassle that way |
| 09:11.29 | ken___ | henk: so what do you want asterisk to do exactly ? |
| 09:11.57 | ken___ | BAM ! got zaptel running |
| 09:12.00 | ken___ | sons of bitches! |
| 09:12.48 | henk | use my sipdiscount account to send sms using the php agi scripts I already have. whould have been the easiest way to do it. But i can write some other lib too |
| 09:13.11 | ken___ | henk: i mean, where are the sms messages originating? on a VoIP phone ? |
| 09:13.47 | henk | servers, websites, email, anywhere... i dont care i like to intergrate everything ;) |
| 09:14.11 | ken___ | henk: ok, well, if it's not coming from a phone, you probably don't need to integrate it with asterisk |
| 09:14.33 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 09:15.24 | henk | not for in comming no. but for outgoing having my already working sipdiscount setup work with some sms tool in asterisk whould have been handy |
| 09:15.33 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 09:19.00 | ken___ | aiight, thanks mog! i appreciate it |
| 09:19.02 | ken___ | later everyone |
| 09:19.17 | *** part/#asterisk henk (n=marius@s5593c2e9.adsl.wanadoo.nl) |
| 09:19.49 | xnon | i cant send anything for pastebin :( |
| 09:20.17 | *** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 09:24.53 | xnon | i have a router is posible that my router is not allow me for send anything with pastebin |
| 09:25.55 | *** join/#asterisk Tommmo (n=tps@203.62.181.52) |
| 09:26.24 | Tommmo | is anyone here good with Realtime? I'm trying to find a way around having to define the realtime contexts in extensions.conf |
| 09:26.28 | Tommmo | eg : |
| 09:26.32 | Tommmo | [context] |
| 09:26.38 | Tommmo | switch => Realtime/mycontext@realtime_ext |
| 09:26.53 | Tommmo | is there a way I can tell it to use Realtime for ALL contexts? |
| 09:27.10 | *** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au) |
| 09:29.25 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 09:30.49 | xnon | http://pastebin.ca/124930 |
| 09:30.52 | xnon | hey friends |
| 09:30.57 | xnon | there is my problem |
| 09:31.13 | xnon | this is my error in my asterisk console |
| 09:31.22 | xnon | one of more |
| 09:35.35 | xnon | hello anybody here? |
| 09:36.54 | xnon | i have few errors in my asterisk console and i wish resolve it |
| 09:38.26 | *** join/#asterisk apardo (n=apardo@87.217.145.102) |
| 09:40.37 | xnon | the pastebin is http://pastebin.ca/124936 |
| 09:43.01 | Chris-NB | anyone tried out ultramonkey with two asterisk ? |
| 09:45.24 | xnon | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 09:46.44 | mog | please dont post links to warez in channel |
| 09:47.55 | DarKnesS_WolF | anyone manged to get rx_fax to work ? |
| 09:48.02 | _4d4m_ | xnon: looks like you have not specified a valid DSN in res_odbc.conf. I dont use odbc connections myself so doubt i can help much further.. |
| 09:48.04 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:48.07 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 09:48.12 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 09:48.31 | _4d4m_ | Chris-NB: what are you trying to accomplish? active/active? active/passive? |
| 09:48.58 | Chris-NB | _4d4m_, active/active with only two boxes |
| 09:50.08 | _4d4m_ | Chris-NB: i've played around with it a bit, but long ago dropped it in favour of load balancing solutions |
| 09:50.23 | _4d4m_ | for active/active setups that is |
| 09:50.44 | Chris-NB | _4d4m_, what load balancing solutions do you mean? a third box? |
| 09:50.48 | Chris-NB | infront of them |
| 09:50.57 | Chris-NB | doing the balancing thing? |
| 09:51.00 | _4d4m_ | you can do it that way |
| 09:51.02 | _4d4m_ | or you can use DNS |
| 09:51.15 | _4d4m_ | some people use dundi |
| 09:51.28 | mog | dundi is way to go |
| 09:52.10 | Chris-NB | _4d4m_, mhm, thought about DNS. but I've to do failover as well |
| 09:54.01 | _4d4m_ | then i'd either suggest dundi, or replcating proxy's that provide load distribution, the vovida load balancer (or even some active/passive SIP aware router/firewall set-up could distribute the load) |
| 09:54.03 | Chris-NB | _4d4m_, now I'm at that point, that I can share load on incoming tcp requests for the manager interface with rr and wrr. but only on 1 box, not on the other one : / |
| 09:54.35 | Chris-NB | _4d4m_, and had no success for sip requests until now : / |
| 09:54.42 | _4d4m_ | hmm.. |
| 09:57.09 | *** join/#asterisk inspired (n=mikael@85.221.0.46) |
| 10:03.49 | xnon | hey friends |
| 10:03.53 | *** join/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com) |
| 10:04.13 | DarKnesS_WolF | i hate faxing in asterisk ! |
| 10:04.15 | xnon | i want to install a g729 and g723.1 codecs free |
| 10:04.35 | benjk | installation is free |
| 10:04.39 | benjk | use is not |
| 10:05.08 | xnon | :( |
| 10:05.11 | benjk | DarKnesS_WolF, I hate faxing |
| 10:05.12 | xnon | what can i do? |
| 10:05.19 | benjk | purchase a license |
| 10:05.24 | xnon | i want to do these codecs |
| 10:05.40 | benjk | purchase a license or use codec hardware |
| 10:05.41 | xnon | take a look this http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 10:05.54 | benjk | its not free |
| 10:05.58 | DarKnesS_WolF | xnon: u can't have them for free only for education |
| 10:06.19 | benjk | its not legal, unless you are using it for educational purposes |
| 10:06.52 | benjk | I don't understand all those codec freaks |
| 10:06.55 | xnon | ummmm ok |
| 10:06.59 | benjk | I want a Mercedes for free |
| 10:07.09 | benjk | and I want a Jaguar for free, too |
| 10:07.14 | *** join/#asterisk cian (n=cian@cian.ws) |
| 10:07.33 | xnon | jejejejejje mercedes! woao i want a Mustang Shelby GT 550 |
| 10:07.36 | xnon | :P |
| 10:07.43 | benjk | yeah, right |
| 10:07.58 | benjk | see, how unreasonable your "I want free" is?! |
| 10:08.07 | benjk | its not free, period |
| 10:08.10 | xnon | ok i understand |
| 10:08.17 | xnon | i dont care if is necesary pay |
| 10:08.21 | benjk | why use those codecs anyway, they suck balls |
| 10:08.23 | DarKnesS_WolF | http://pastebin.ca/124957 any idea guys? |
| 10:08.27 | xnon | but how i can pay for this |
| 10:08.35 | benjk | there are good free codecs avaialble |
| 10:08.39 | benjk | Speex and ILBC |
| 10:08.44 | benjk | also GSM 06.10 |
| 10:09.12 | benjk | you can go to Digium's website and purchase a license for the g729 soft codec |
| 10:09.24 | xnon | yea but my IP phones only support g711 g729 and g723.1 but g711 is not so good friend! |
| 10:09.34 | benjk | or you can go to Digium's website and purchase a G729+G723 hardware codec (PCI card) |
| 10:09.52 | benjk | g711 is actually much better |
| 10:09.59 | docelmo | The card isnt out yet |
| 10:10.04 | benjk | its superior sound quality |
| 10:10.16 | benjk | well, then you have to wait a little |
| 10:10.20 | DarKnesS_WolF | xnon: why i'm using g711 here |
| 10:10.44 | DarKnesS_WolF | benjk: any idea http://pastebin.ca/124957 ? |
| 10:11.04 | xnon | yes g711 for local net is good but in extern red is not so good |
| 10:11.40 | xnon | i have a ALL7050 IP Hardware Phone |
| 10:12.10 | xnon | and my asterisk have some problems with these codecs |
| 10:12.19 | xnon | i dont know what |
| 10:12.28 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 10:12.39 | xnon | but ired that g729 is excelent and g723.1 too |
| 10:13.23 | xnon | i read sorry |
| 10:13.42 | xnon | DarKnesS_WolF, this screip is for send faxes? |
| 10:13.58 | DarKnesS_WolF | this what? |
| 10:14.14 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 10:14.23 | xnon | your pastebin code |
| 10:14.38 | xnon | is for recive fax? or send or what? |
| 10:14.55 | xnon | sorry my english is not the bether |
| 10:15.41 | DarKnesS_WolF | resive |
| 10:15.46 | DarKnesS_WolF | recive |
| 10:17.30 | benjk | I don't use fax |
| 10:17.38 | benjk | not through a PBX |
| 10:17.51 | xnon | cool and it work? |
| 10:17.54 | DarKnesS_WolF | okay :) |
| 10:17.59 | DarKnesS_WolF | xnon: read there is errors:P |
| 10:18.05 | xnon | ok |
| 10:18.11 | benjk | analog line used for ADSL, directly plugged in to an old fax machine |
| 10:18.18 | xnon | you have any hardware? |
| 10:18.44 | *** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
| 10:20.27 | xnon | in your extensions.conf are u including fax-in context? |
| 10:21.48 | *** join/#asterisk nounoursfr (n=nounours@core1.mesbox.net) |
| 10:21.52 | nounoursfr | hi all |
| 10:22.12 | xnon | hifriend |
| 10:22.34 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 10:22.48 | nounoursfr | do you have a test asterisk en contrak sip ? |
| 10:23.10 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 10:23.23 | nounoursfr | for loadbalanc asterisk on multi box |
| 10:24.38 | xnon | friends for use G729 Codec is necesary a Dual Xeon 2.8GHz processors????? |
| 10:25.20 | nounoursfr | no for use codec G729 I use 1 P4 2.8Ghz |
| 10:25.45 | nounoursfr | asterisk do not use Multi Processors |
| 10:26.09 | xnon | ok |
| 10:26.11 | erwinism | my asterisk server is P3 933Mhz and it works perfecly |
| 10:26.19 | xnon | i have a P4 2.0 Ghz |
| 10:26.54 | xnon | erwinism, i mean for use G729 codec no Asterisk |
| 10:27.22 | nounoursfr | erwinism: you control how much call? |
| 10:27.53 | erwinism | nounoursfr:two pstn and 10 voip |
| 10:28.14 | benjk | YAY |
| 10:29.04 | phearless | I try to upgrade my cisco voip phone 7960 to the last firmware, and this stupid phone request by TFTP the firmware "P003-07-.bin" .... why ? |
| 10:29.09 | xnon | hey friends take a look! |
| 10:29.10 | xnon | Note: Should you want to operate 5 G.729 channels on one machine |
| 10:29.39 | benjk | <PROTECTED> |
| 10:29.39 | benjk | <PROTECTED> |
| 10:29.39 | benjk | DEBUG: spawn extension (macro-DialExtension, s-DIAL-EXT, 4) returned '0' in macro 'DialExtension' |
| 10:29.40 | benjk | <PROTECTED> |
| 10:29.40 | benjk | <PROTECTED> |
| 10:29.50 | xnon | this mean that 1 license can used only for a 5 extensions number in a asterisk server? |
| 10:29.59 | benjk | finally app_macro knows how to go to h |
| 10:30.09 | erwinism | my cisco 7960 voip phone is useless too. |
| 10:30.09 | benjk | that was a tough one |
| 10:30.17 | xnon | benjk, used pastebin.ca for this |
| 10:30.33 | benjk | four lines is still ok |
| 10:30.49 | xnon | ummmm |
| 10:30.53 | xnon | ok |
| 10:30.59 | benjk | anyway, your Asterisk won't do this :) |
| 10:31.10 | benjk | cause its broken |
| 10:33.01 | xnon | benjk is with me? |
| 10:33.04 | xnon | http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=G729CODEC |
| 10:33.47 | benjk | yes, that's the softcodec |
| 10:34.34 | xnon | yea |
| 10:35.11 | xnon | but is onlye for a 5 extensions number for my asterisk or all my asterisk server? |
| 10:35.31 | benjk | its for one channel |
| 10:35.44 | benjk | means you can have one call at a time use G729 |
| 10:35.59 | xnon | ok 1 channel = 1 extension number in my asterisk server? |
| 10:36.06 | benjk | no |
| 10:36.13 | xnon | ok ok i understod |
| 10:36.21 | benjk | one channel means one call at a time |
| 10:36.32 | mutilator | think anyone else would have a use for an activex control to interface with manager? |
| 10:36.39 | benjk | you want two calls at the same time which can use g729, you need two licenses |
| 10:36.54 | xnon | houch! |
| 10:37.06 | xnon | i dont like it |
| 10:37.10 | xnon | jejeje |
| 10:37.11 | benjk | but it doesn't matter how many extensions you have |
| 10:37.24 | benjk | only how many CONCURRENT calls you want to do G729 |
| 10:38.13 | xnon | ok |
| 10:38.52 | xnon | and if i want to do calls for a sip provider channel |
| 10:38.56 | xnon | ???????? |
| 10:40.44 | xnon | adamvozip.es = ISP Provider, i have a 100251 SIP number with they, this number 100251 is asociated in my asterisk server how a SIP provider context in sip.conf |
| 10:40.51 | benjk | it doesn't matter where the calls come from |
| 10:40.59 | benjk | it doens't matter where the calls go to |
| 10:40.59 | xnon | sorry adamvozip.es is a SIP provider |
| 10:41.18 | benjk | the only thing that matters is how many calls need to use the G729 AT THE SAME TIME |
| 10:41.27 | *** part/#asterisk evol-emil (n=emile@landi.oddi.is) |
| 10:41.28 | xnon | ok |
| 10:41.33 | xnon | well |
| 10:42.22 | xnon | i have send calls and recieve calls for this numer SIP adamvozip.es but the quality for this calls is so bad |
| 10:42.45 | benjk | then use a different provider |
| 10:43.14 | xnon | if i do call for the extension 113 to a adamvozip numer 100251, the call quality is so bad |
| 10:43.40 | xnon | what sip provider are you use? |
| 10:44.28 | xnon | freeworlddialup.com for example'?? |
| 10:46.52 | xnon | softphone comatible with FWD? |
| 10:48.39 | *** join/#asterisk evol-emil (n=emile@landi.oddi.is) |
| 11:02.06 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 11:12.12 | ionix | X-Lite |
| 11:13.00 | erwinism | guys i have to go thanks for the help |
| 11:14.22 | *** join/#asterisk ronn (n=zakforev@87.112.6.129.bbplus.ptn-ag1.dyn.plus.net) |
| 11:17.55 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 11:19.08 | fourcheeze | how much overhead is involved in calling a sip client that doesn't exist? |
| 11:19.29 | fourcheeze | e.g. dial(sip/123) where 123 isn't registered |
| 11:19.49 | *** part/#asterisk daysmen3 (n=primus@host86-138-239-164.range86-138.btcentralplus.com) |
| 11:19.50 | fourcheeze | is it something * handles easily or is it something I should test for? |
| 11:19.55 | *** join/#asterisk daysmen3 (n=primus@host86-138-239-164.range86-138.btcentralplus.com) |
| 11:26.32 | fourcheeze | by extension of that, if I had 3 * boxes and didn't know which one a user was logged into would it be terrible to call the user at each one simultaneously, knowing that 2/3 would fall through? |
| 11:27.32 | Assid | why are you doing that? |
| 11:28.24 | Assid | if i were you.. i'd just use asterisk realtime.. and let asterisk automatically pickup |
| 11:30.28 | Assid | and it just checks if there is a registration in the astdb if not.. it sends back not registered |
| 11:30.46 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 11:32.56 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
| 11:34.53 | *** part/#asterisk my007ms (n=noor@217.139.224.194) |
| 11:37.34 | *** part/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 11:40.56 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 11:45.56 | fourcheeze | Assid: I'm using realtime |
| 11:45.59 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 11:46.02 | fourcheeze | how do I know where a user is logged in? |
| 11:48.19 | Assid | if your using realtime.. then you dont need to worry |
| 11:48.25 | *** join/#asterisk lsackette (n=lsackett@c-69-142-135-33.hsd1.nj.comcast.net) |
| 11:48.26 | Assid | just call that extension and it will work |
| 11:51.15 | fourcheeze | that's never worked for me before |
| 11:51.24 | Assid | weird |
| 11:51.36 | fourcheeze | which table is it using to store that? |
| 11:51.47 | Assid | whatever you setup for realtime sip |
| 11:51.51 | fourcheeze | ok |
| 11:51.55 | fourcheeze | I'll try again now |
| 11:52.09 | fourcheeze | which field? |
| 11:52.14 | fourcheeze | maybe I have an old schema |
| 11:52.22 | *** join/#asterisk vcon (n=ad4@e182076207.adsl.alicedsl.de) |
| 11:52.30 | Assid | not sure |
| 11:52.38 | fourcheeze | Assid: but you have this working? |
| 11:52.57 | Assid | nah.. i just got 1 box on realtime.. cause im playing with it |
| 11:52.58 | fourcheeze | what should my Dial() command look like? |
| 11:53.15 | Assid | Dial(SIP/sipuser|30) |
| 11:53.22 | fourcheeze | yeah, that doesn't work for me |
| 11:53.41 | Assid | sip show peer sipuser load |
| 11:54.11 | Assid | do it on the box that doesnt have the user registered |
| 11:54.25 | queuetue | How do I check if an extension actually exists? (Trying to automate extension setup) |
| 11:54.35 | vcon | hello, i look for a stun plugin for asterisk or something else. has someone an idea? |
| 11:54.56 | fourcheeze | Assid: Status : UNKNOWN |
| 11:54.59 | Assid | vcon: you need an stun server.. |
| 11:55.16 | queuetue | Sorry, if a SIP peer exists - not if it's connected, but if it exists. |
| 11:55.23 | Assid | fourcheeze: but do you get the basic info? |
| 11:55.23 | vcon | yes i know, but i need a tool which makes the stun request |
| 11:55.45 | Assid | queuetue: realtime??? |
| 11:56.02 | fourcheeze | Assid: yep |
| 11:56.18 | fourcheeze | Assid: I get everything apart from any indication that the user is available |
| 11:56.21 | Assid | vcon: nah.. not sure but as far as i know.. stun should work on its own.. as long as the server is ona public network.. i dotn see why there should be a problem |
| 11:56.29 | fourcheeze | vcon: http://www.vovida.org/applications/downloads/stun/ |
| 11:56.33 | fourcheeze | vcon: there's a stun client there |
| 11:56.34 | queuetue | Assid: Well, as real as possible. I'm not sure how you mean that question - unless you meant to ask about "runtime"... |
| 11:56.38 | Assid | do you have a regcontact field in your database? |
| 11:56.48 | DarKnesS_WolF | http://pastebin.ca/124957 any idea? |
| 11:56.57 | Assid | queuetue: do you know what realtime is? |
| 11:57.20 | fourcheeze | Assid: yes I have regcontact and Reg Contact shows on sip show peer for anyone logged into the server |
| 11:57.36 | Assid | DarKnesS_WolF: whats the problem.. the error says its wrong |
| 11:57.42 | queuetue | Assid: Yes. When a program's execution is assured to execute within specific parameters. |
| 11:57.51 | Assid | queuetue: no.. ! |
| 11:58.45 | vcon | Assid: my server is behind nat and i have random ip changes. now i look for a tool which makes stun request for asterisk. a tool which tells asterisk what the current public ip address is. |
| 11:58.46 | Assid | fourcheeze: i remember ssomewhere about the table definition being changed to add more ffields. dotn remember what tho.. |
| 11:59.03 | fourcheeze | Assid: can you list your fields and paste somewhere? |
| 11:59.09 | queuetue | Ok, can anyone else explain how to test for the existence of a SIP peer? |
| 11:59.21 | Assid | sure i fi can get to my box!.. nets all messed up cause of the rains here |
| 11:59.37 | Assid | fourcheeze: http://www.voip-info.org/wiki-Asterisk+RealTime+Sip |
| 11:59.45 | Assid | also you may want to rtcache and stuff |
| 11:59.49 | DarKnesS_WolF | Assid: why ut's wrong it's should be fine :-s |
| 12:00.24 | Assid | ? |
| 12:00.36 | Assid | DarKnesS_WolF: ug 10 12:02:55 WARNING[18906]: pbx.c:2357 __ast_pbx_run: Channel 'Zap/2-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 12:00.46 | Assid | i dont see an s extension in you default context |
| 12:00.54 | DarKnesS_WolF | yes what it should be? |
| 12:01.05 | DarKnesS_WolF | should i have s extension ? and why ? what it should be look like? |
| 12:01.05 | Assid | i see a fax extension in your default context |
| 12:01.10 | fourcheeze | Assid: yeah I have rtcache |
| 12:01.19 | Assid | exten => fax,1,Goto(fax-in,s,1) exten => s,1,Goto(fax-in,s,1) |
| 12:01.33 | DarKnesS_WolF | both ? |
| 12:01.38 | DarKnesS_WolF | i should have both ? |
| 12:01.44 | Assid | fourcheeze: theoretically it should work since the information is taken from realtime |
| 12:01.54 | Assid | unless.. asterisk needs astdb database to be seeded as well |
| 12:02.03 | Assid | that could be the reason why its not working |
| 12:02.05 | fourcheeze | hmm |
| 12:02.10 | DarKnesS_WolF | or u mean i remove the fax on ? |
| 12:02.17 | queuetue | Can macros be used in other files, such as sip.conf? |
| 12:02.19 | DarKnesS_WolF | if i did i get error about fax extension dose not exists |
| 12:02.21 | Assid | replace it .. or add it.. no issues.. |
| 12:02.22 | *** join/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:02.23 | *** join/#asterisk daysmen3 (n=primus@host86-138-239-164.range86-138.btcentralplus.com) |
| 12:02.34 | Assid | add it then |
| 12:02.42 | DarKnesS_WolF | Assid: if i removed it it will never work :-s |
| 12:02.47 | queuetue | Soy, can I define a sip peer via a macro? |
| 12:02.50 | DarKnesS_WolF | telling me fax extenesion dosn't exists |
| 12:02.54 | DarKnesS_WolF | [TK]D-Fender: wb ;-) |
| 12:02.55 | Assid | queuetue: macros are for dialplans.. |
| 12:02.59 | Assid | mornin tkd |
| 12:03.01 | [TK]D-Fender | queuetue: Nope. |
| 12:03.15 | [TK]D-Fender | DarKnesS_WolF, Assid : y0 |
| 12:03.20 | fourcheeze | Assid: what should happen? Should asterisk call the Reg Contact, or call the extension at the server where it is registered? |
| 12:03.32 | DarKnesS_WolF | [TK]D-Fender: what is wrong in this http://pastebin.ca/124957 any idea? |
| 12:03.52 | Assid | asterisk should call the sip registered user/device.. but apparently it seems astdb needs to be seeded |
| 12:03.53 | queuetue | Are there any efforts underway to abstract away the dialplan from asterisk? Maybe consider it a replaceable module? |
| 12:04.11 | Assid | dialplan away from asterisk ? |
| 12:04.29 | [TK]D-Fender | DarKnesS_WolF: just like it says.. there is no "s" in [default] |
| 12:04.32 | queuetue | It would be nice if the DP language was replaceable. |
| 12:04.56 | DarKnesS_WolF | [TK]D-Fender: what it should look like |
| 12:05.02 | Assid | queuetue: you could use ael |
| 12:05.25 | [TK]D-Fender | DarKnesS_WolF: "fax" only works while * is processing other stuff and THEN catches the tone. It does not imply that it will sit an WAIT for in when entering the context for any reason. |
| 12:05.51 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 12:05.55 | [TK]D-Fender | queuetue: Think of AGI. Though its implementation is "heavy". |
| 12:06.40 | [TK]D-Fender | DarKnesS_WolF: And you should never make a context so generic as to be called [default], that is a bad habit |
| 12:06.41 | DarKnesS_WolF | hum Assid told me to add exten => s,1,goto(fax-in,s,1) adding it in the default |
| 12:06.55 | queuetue | [TK]D-Fender: Could you really build a call manager, define sip/IAX/Zap interfaces, all through AGI? Haven't been through those docs yet, but I got the impression it was limited. |
| 12:06.58 | DarKnesS_WolF | [TK]D-Fender: i swear it was empty :D i was using fax-in only |
| 12:06.58 | [TK]D-Fender | DarKnesS_WolF: What else falls on that context? |
| 12:07.12 | DarKnesS_WolF | default is tottaly empty |
| 12:07.38 | fourcheeze | Assid: how would I seed that? |
| 12:07.43 | [TK]D-Fender | DarKnesS_WolF: is this from a dedicated Fax Zap channel? |
| 12:07.57 | [TK]D-Fender | DarKnesS_WolF: Or do you really need to detect it first? |
| 12:08.32 | Assid | fourcheeze: i have no clue!!!! :( |
| 12:08.36 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:08.36 | Assid | thinking......... |
| 12:08.54 | Assid | oh wait.. tkd is here.. he could dream up a code in a jiffy |
| 12:09.04 | Assid | tkd.. you up for a small mind bender? |
| 12:09.38 | [TK]D-Fender | Assid: As long as the bend doesn't set :) |
| 12:09.39 | DarKnesS_WolF | [TK]D-Fender: yes it is |
| 12:09.53 | DarKnesS_WolF | [TK]D-Fender: for the incoming no it's the fax only channel |
| 12:10.02 | Assid | couple of asterisk boxes.. no clue where the sip user registers.. using asterisk realtime.. how do we call back to the sip user from a box that doesnt have the user seeded |
| 12:10.10 | DarKnesS_WolF | but for the outgoing i use this channel to do normal phone calls |
| 12:10.22 | fourcheeze | [TK]D-Fender: this must be a FAQ by now surely? |
| 12:10.24 | [TK]D-Fender | DarKnesS_WolF: Then just chage "fax" to "s" in [default] and that will fix everything. you simply want to START RECEIVING assuming its a fax. |
| 12:10.52 | [TK]D-Fender | DarKnesS_WolF: You only use[fax] in your first level IVR so that you don't NEED a dedicated line. |
| 12:10.57 | DarKnesS_WolF | [TK]D-Fender: u done that already and i got this erro that there is no fax extension. |
| 12:11.19 | [TK]D-Fender | DarKnesS_WolF: Just do it. |
| 12:11.52 | [TK]D-Fender | DarKnesS_WolF: : Oh and remove that "include" |
| 12:12.00 | DarKnesS_WolF | it did it will told me fax detected but there is no fax extenstion then it's said no 't' in fax-in for timeout |
| 12:12.12 | [TK]D-Fender | DarKnesS_WolF: And the "t" exxetn while you're at it |
| 12:12.27 | [TK]D-Fender | DarKnesS_WolF: Just do it. |
| 12:12.39 | Assid | nike! |
| 12:12.44 | DarKnesS_WolF | exten t,1,hangup ? |
| 12:12.57 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 12:13.15 | [TK]D-Fender | Assid: You mean the client chooses which server to reg to and the config is mirrored between them? |
| 12:13.39 | fourcheeze | [TK]D-Fender: config is in realtime - 1 config for a number of servers |
| 12:13.43 | [TK]D-Fender | DarKnesS_WolF: Yes, no need. "s,3" could simply be "Hangup" |
| 12:14.06 | Assid | [TK]D-Fender: well. using realtime.. so the authentication isnt a problem.. but since the client connects to server 2 as opposed to server 1 and 3.. the seed is in 2 , how would 1 or 3 call the sip user.. he says it doesnt work |
| 12:14.41 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.ca/125101 good like this ? |
| 12:14.44 | [TK]D-Fender | fourcheeze: Nice try on trying to create a "cloud" but * is NOT a SIP proxy. Making roaming virtual users across server is a royal PITA. |
| 12:15.02 | fourcheeze | [TK]D-Fender: well all I need to know is where each user is |
| 12:15.06 | [TK]D-Fender | DarKnesS_WolF: Chang the "t" for "s,3" |
| 12:15.15 | fourcheeze | since each * is connected to the same db surely this can't be hard |
| 12:15.34 | [TK]D-Fender | DarKnesS_WolF: And I think you might want to do "Answer" first.... |
| 12:15.41 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 12:15.55 | Assid | oh crap your right theres no answer |
| 12:16.10 | fourcheeze | Assid: as I feared |
| 12:16.17 | [TK]D-Fender | fourcheeze: WRONG. When phone 123 reg's to server A, how does B know that? |
| 12:16.18 | fourcheeze | so assuming that I can't know where each user is |
| 12:16.36 | DarKnesS_WolF | [TK]D-Fender: ok added s,1, in fax-in as answer anything els? |
| 12:16.46 | [TK]D-Fender | fourcheeze: Forget assuming "you can't" and tell me how you think you CAN <- |
| 12:16.46 | Assid | DarKnesS_WolF: try it |
| 12:16.47 | fourcheeze | [TK]D-Fender: well it would be simple for a field in the realtime sip peers to have that information |
| 12:17.13 | Assid | query astdb? :P |
| 12:17.16 | DarKnesS_WolF | ok will ask a freind to send a test |
| 12:17.20 | [TK]D-Fender | fourcheeze: If you made your dialplan |
| 12:17.44 | [TK]D-Fender | hold on |
| 12:17.47 | [TK]D-Fender | ... |
| 12:19.31 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.ca/125107 |
| 12:19.33 | DarKnesS_WolF | :-s |
| 12:20.16 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:20.33 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 12:20.39 | Assid | exten => s,1,Goto(fax-in,s,1) ------ change s to fax |
| 12:20.44 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 12:20.51 | Assid | and in fax-in context change all s to fax |
| 12:20.58 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 12:21.17 | Assid | you may want to add a h extension as well |
| 12:21.22 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 12:21.24 | DarKnesS_WolF | Assid: if i have done i'll have the error of no S ext. |
| 12:21.28 | Assid | h,1,Hangup -- maybe redundant but okay |
| 12:21.38 | [TK]D-Fender | fourcheeze: Trying to see how you can share a reg DB..... doen't make sense |
| 12:21.54 | fourcheeze | [TK]D-Fender: what doesn't make sense? |
| 12:22.04 | Assid | DarKnesS_WolF: No you wont.. cause we are specifyig it to go to fax extension not s extension |
| 12:22.17 | DarKnesS_WolF | ok |
| 12:22.20 | fourcheeze | if there was a field in the DB where when the user registered it recorded the host they registered with |
| 12:22.26 | Assid | DarKnesS_WolF: reload your extensions and try it |
| 12:22.30 | fourcheeze | then surely I can dial(sip/user@host) |
| 12:22.37 | [TK]D-Fender | DarKnesS_WolF: Disable fax detect from that Zap channel; |
| 12:22.43 | sumasuma | Can asterisk work with decentralised networks ? |
| 12:22.48 | fourcheeze | of course there is no field, but I'm wondering if there's some other mechanism |
| 12:23.09 | fourcheeze | like just dialling the reg contact |
| 12:23.17 | sumasuma | is there any mechanism to make it decentralised ? |
| 12:23.18 | [TK]D-Fender | fourcheeze: You'd have to use only a pattern match for your extensions then and query the DB on each to try and determin if it was local. |
| 12:23.21 | fourcheeze | which I should be able to get to in realtime, but can't seem to |
| 12:23.35 | fourcheeze | [TK]D-Fender: that sounds like a lot of work |
| 12:23.36 | DarKnesS_WolF | [TK]D-Fender: hum ok |
| 12:23.38 | [TK]D-Fender | fourcheeze: Again, * is NOT a SIP prox. thats why people use SER. |
| 12:23.55 | Assid | i need to work with ser.. |
| 12:24.01 | fourcheeze | [TK]D-Fender: sure I understand, but ser doesn't do presence very well |
| 12:24.08 | fourcheeze | if at all at the moment |
| 12:24.22 | Assid | DarKnesS_WolF: did you do it? |
| 12:24.43 | Assid | i tried reading some openser code.. i thought i was reading C |
| 12:24.43 | fourcheeze | [TK]D-Fender: so here's a related question - how bad would it be to call all of the servers at once? |
| 12:24.57 | fourcheeze | [TK]D-Fender: knowing that all except one would fail |
| 12:25.07 | DarKnesS_WolF | Assid: wait |
| 12:25.09 | Assid | fourcheeze: do you give users voicemail? |
| 12:25.23 | fourcheeze | Assid: yes, but I'm planning to have that on a separate server |
| 12:25.28 | DarKnesS_WolF | Assid: [TK]D-Fender : http://pastebin.ca/125110 anything else\:? |
| 12:25.49 | Assid | DarKnesS_WolF: NOOOOOOOOOOOOOOOOOOOOO |
| 12:25.56 | Assid | exten => fax,1,Goto(fax-in,fax,1) NOOOO |
| 12:25.56 | DarKnesS_WolF | Assid: ? |
| 12:26.02 | Assid | exten => s,1,Goto(fax-in,fax,1) |
| 12:26.04 | Assid | there |
| 12:26.06 | DarKnesS_WolF | Assid: ahh ok |
| 12:26.10 | fourcheeze | if I could sort this out then * would just scale perfectly |
| 12:26.20 | fourcheeze | and even without SER |
| 12:26.26 | Assid | fourcheeze: if i were you.. i'd use SER |
| 12:26.41 | fourcheeze | Assid: as far as I can tell SER can't be highly available either |
| 12:26.56 | *** join/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 12:27.07 | fourcheeze | Assid: I want to have redundancy in all core functions |
| 12:27.07 | Assid | fourcheeze: err. i thought SER can handle thousands of users |
| 12:27.16 | fourcheeze | this is fine until server breaks |
| 12:27.25 | Assid | so add a round robin dns.. |
| 12:27.43 | fourcheeze | well then I can still be registered at either server |
| 12:27.56 | Assid | right.. also .. you could use heartbeat |
| 12:27.58 | fourcheeze | as far as I can tell SER is no better at doing this than asterisk |
| 12:28.04 | fourcheeze | heartbeat is a kludge |
| 12:28.05 | [TK]D-Fender | fourcheeze: I think I'm going to step away from this. |
| 12:28.10 | ionix | you are wrong fourcheeze |
| 12:28.13 | fourcheeze | [TK]D-Fender: ahhh coward |
| 12:28.17 | fourcheeze | ;-) |
| 12:28.18 | ionix | SER is much faster to link internal SIP |
| 12:28.23 | Assid | DarKnesS_WolF: did you get it up |
| 12:28.41 | ionix | so let say you'd have an organization with 40 phones, I would link them with SER. |
| 12:28.41 | Assid | yep.. its just that i have no clue how to work with SER |
| 12:28.44 | fourcheeze | ionix: you mean I can have 2 SERs and call across them automagically? |
| 12:29.05 | ionix | Asterisk takes care of the dialplan and such |
| 12:29.11 | [TK]D-Fender | fourcheeze: No, mearly not CRAZY trying to do things * wasn't designed to do. asking for a world of pain..... |
| 12:29.28 | fourcheeze | [TK]D-Fender: I'll happily accept a product B solution |
| 12:29.38 | Assid | fourcheeze: cant you use ser with realtime? |
| 12:29.44 | [TK]D-Fender | fourcheeze: Broadsoft? |
| 12:29.47 | fourcheeze | I don't think ser talks to realtime |
| 12:29.52 | ionix | if you have two phones on SER, SER will route the call between them and inform asterisk. |
| 12:29.54 | fourcheeze | [TK]D-Fender: never heard of that one |
| 12:30.05 | fourcheeze | ionix: what about 2 phones on 2 separate SERs |
| 12:30.26 | fourcheeze | ionix: where each phone could be registered with either |
| 12:30.57 | fourcheeze | [TK]D-Fender: is that FLOSS? |
| 12:31.04 | Assid | i wanna learn SER :| |
| 12:31.05 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 12:31.09 | [TK]D-Fender | fourcheeze: LOL!!!! $$$ |
| 12:31.19 | fourcheeze | Assid: SER isn't hard when you get your head around SIP |
| 12:31.27 | fourcheeze | I just don't think it does what I want |
| 12:31.39 | DarKnesS_WolF | Assid: didn't work the other end said he heard the tone |
| 12:31.45 | DarKnesS_WolF | and then it got d.c in the middle |
| 12:31.53 | Assid | heard the tone ? |
| 12:31.56 | DarKnesS_WolF | yes |
| 12:32.01 | DarKnesS_WolF | the fax tone |
| 12:32.01 | Assid | the fax tone? |
| 12:32.05 | DarKnesS_WolF | yes |
| 12:32.13 | *** join/#asterisk [koss] (i=koss@adsl-75-36-15-21.dsl.bcvloh.sbcglobal.net) |
| 12:32.28 | Assid | before that did he hear the fax tone? |
| 12:32.39 | DarKnesS_WolF | yes |
| 12:32.51 | Assid | try with faxdetect on.. and also paste the CLI output |
| 12:32.54 | fourcheeze | all I want is a cluster where any one server can go down and I don't need to worry about heartbeat |
| 12:33.06 | fourcheeze | in my experience servers die but can hang on to IP numbers |
| 12:33.14 | fourcheeze | which is why I don't trust heartbeat |
| 12:33.33 | Assid | fourcheeze: thats why you use heartbeat router.. and internal ips for the rest of the servers |
| 12:33.36 | fourcheeze | it should be possible to build-in such redundancy |
| 12:34.07 | fourcheeze | Assid: still not completely convinced |
| 12:34.10 | DarKnesS_WolF | Assid: the command line didn't show anything strange just the fax the file and then hangup in h |
| 12:34.27 | fourcheeze | also I don't think that SER is good for presence |
| 12:34.34 | fourcheeze | so I really need users to register with * |
| 12:34.43 | fourcheeze | even if they go through SER in some way |
| 12:34.43 | Assid | DarKnesS_WolF: did you try with faxdetect on ? |
| 12:35.05 | *** part/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 12:35.18 | Assid | [TK]D-Fender: they might be buying that sangoma card. |
| 12:35.29 | Assid | said they will let me know |
| 12:35.32 | DarKnesS_WolF | Aug 10 15:33:43 WARNING[24246]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'fax-in' |
| 12:35.35 | DarKnesS_WolF | <PROTECTED> |
| 12:35.38 | DarKnesS_WolF | <PROTECTED> |
| 12:35.40 | DarKnesS_WolF | <PROTECTED> |
| 12:35.44 | Assid | err.. you removed the t extension? |
| 12:35.48 | DarKnesS_WolF | Assid: it was always faxing detect incoming |
| 12:35.52 | DarKnesS_WolF | Assid: yep :-s |
| 12:35.59 | Assid | bah |
| 12:36.01 | Assid | okay weell |
| 12:36.09 | sumasuma | Darkness_wolf: i have been seeing you with this problem from the morning |
| 12:36.14 | Assid | set absolute timeout to 300 |
| 12:36.19 | sumasuma | Darkness_wolf: i appreciate your patience |
| 12:36.20 | Assid | and add a t extension |
| 12:36.43 | DarKnesS_WolF | sumasuma: lol i have too fix it !! i'll never give up ! even [TK]D-Fender x86 always kicking my ass |
| 12:36.48 | DarKnesS_WolF | Assid: ok |
| 12:36.58 | DarKnesS_WolF | but i can ask my friend to send again he sents like 40 fax :P |
| 12:37.10 | Assid | isnt fax free? |
| 12:37.40 | DarKnesS_WolF | Assid: nop :-) it costs phone call |
| 12:38.05 | DarKnesS_WolF | Assid: exten => t,1,hangup ? |
| 12:38.07 | Assid | Set(TIMEOUT(absolute) = 300) |
| 12:38.21 | DarKnesS_WolF | 300 sec? |
| 12:38.28 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 12:38.31 | Assid | 5 mins.. for getting all your faxes in |
| 12:38.31 | Assid | :P |
| 12:38.43 | Assid | if its hung.. it will timeout and hangup as per t extension |
| 12:39.59 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 12:41.53 | DarKnesS_WolF | Assid: http://pastebin.ca/125120 |
| 12:41.57 | Assid | hrmm.. i wonder if i can accept fax over ulaw instead of having a zaptel device ring |
| 12:41.57 | [TK]D-Fender | DarKnesS_WolF: Did you disable it in zapata? |
| 12:42.05 | DarKnesS_WolF | [TK]D-Fender: disabled |
| 12:42.07 | DarKnesS_WolF | enabled |
| 12:42.08 | DarKnesS_WolF | on |
| 12:42.09 | DarKnesS_WolF | no |
| 12:42.11 | DarKnesS_WolF | incoming |
| 12:42.14 | DarKnesS_WolF | all that didn't work :D |
| 12:42.28 | DarKnesS_WolF | now i need to find someone to send me faxes to a lcal phobe in eg:P |
| 12:42.44 | Assid | just call the number |
| 12:42.51 | Assid | see igf it stays on |
| 12:47.02 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 12:49.09 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 12:51.37 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 12:52.54 | xnon | if i want afiliated a FWD account in my asterisk server i must use IAX extensions in my asterisk server or i cant use SIP extensions protocol????????????? |
| 12:53.11 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:54.09 | sumasuma | xnon: what is your problem ? |
| 12:54.21 | sumasuma | you want to register with SIP to fwd ? |
| 12:55.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 12:55.37 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:57.23 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 12:59.28 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 12:59.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:59.41 | [TK]D-Fender | xnon: you can register to them either way you want. You can use IAX to talk to FWD and SIP to talk to your phones and * will translate, or just go direct and that will work as well. |
| 13:00.52 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 13:04.17 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 13:08.27 | jhamlyn | Help please - How do I tell asterisk to accept an incoming call from an anonymous sender using SIP .. I want to direct any incoming call to a special extension |
| 13:08.48 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 13:08.49 | jhamlyn | I have it working for authenticated calls .. but cant get the un authenticated calls to work |
| 13:08.59 | MattH | Hi.. when doing an 'iax2 show netstats' what does 'Lost' mean on the 'LOCAL' side? |
| 13:11.47 | [TK]D-Fender | jhamlyn: Set a context in [general] and "allowguest=yes" |
| 13:12.18 | [TK]D-Fender | jhamlyn: And of course be very careful what is permitted in that context |
| 13:18.38 | hmmhesays | ~seen russelb |
| 13:18.41 | jbot | i haven't seen 'russelb', hmmhesays |
| 13:18.47 | hmmhesays | ~seen russellb |
| 13:18.49 | jbot | russellb is currently on #asterisk, last said: 'Strom_C: knew abount what?'. |
| 13:21.11 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:21.23 | [TK]D-Fender | ~seen [TK]D-Fender Whats redundancy? |
| 13:21.25 | jbot | [TK]D-Fender: i haven't seen '[tk]d-fender whats redundancy' |
| 13:21.44 | [TK]D-Fender | :O |
| 13:25.04 | hmmhesays | if you set a variable inside a macro, can the context the macro was called from access it? |
| 13:25.05 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:25.16 | blitzrage | morning! |
| 13:25.58 | mog | MORNING |
| 13:26.10 | blitzrage | mog: !!! |
| 13:26.15 | blitzrage | what are you doing up so early? :) |
| 13:26.19 | mog | im still up |
| 13:26.21 | blitzrage | hahaha |
| 13:26.26 | blitzrage | sounds like me usually :) |
| 13:26.28 | mog | bought to clock out for the day |
| 13:26.39 | mog | i hope |
| 13:26.44 | blitzrage | nice, just woke up and already into the land of fires (and I'm on vacation :)) |
| 13:27.02 | [TK]D-Fender | blitzrage: I don't want to meet your mom! |
| 13:27.13 | mog | what are you doing online then? |
| 13:27.22 | blitzrage | open question: I have a Cisco 7960 calling from line 1 to line 2, and after 16 seconds, the 2nd line (after answer) drops the call -- does it with canreinvite=no or not... |
| 13:27.24 | [TK]D-Fender | hmmhesays: Sure |
| 13:27.49 | blitzrage | It's the same phone, so maybe it's a hairpin issue? Although it is two separate registrations |
| 13:28.15 | blitzrage | mog: because Mr. Smith is driving across the country, and I'm the only one to help with the issue unfortunately |
| 13:28.42 | blitzrage | trying to figure out why I can't perform a SIP transfer |
| 13:28.55 | blitzrage | and why trunk keeps crashing on me :) |
| 13:29.48 | hmmhesays | i need the set the epoch into a variable when a call is answered |
| 13:30.19 | blitzrage | isn't there an EPOCH var? |
| 13:30.27 | blitzrage | or something similar? |
| 13:30.28 | [TK]D-Fender | hmmhesays: OH you mean in a macro called by DIAL... |
| 13:30.35 | benjk | hmmhesays, were you one of those folks yesterday who were interested in dialstatus info after hangup? |
| 13:30.51 | hmmhesays | benjk: no |
| 13:30.55 | benjk | ok |
| 13:31.06 | hmmhesays | i was interested in writing the cdr before extension h was called |
| 13:31.14 | [TK]D-Fender | hmmhesays: I don't think reversin inheretance works. You'd need a more persistant storage like AstDB or something. |
| 13:31.22 | benjk | well, sort of what I meant |
| 13:31.31 | blitzrage | ok, so the 8.0 FW image for a 7960 is really crappy :) |
| 13:31.36 | hmmhesays | russellb wrote a patch to do so but... it is broken for 1.2.10 |
| 13:31.39 | benjk | I added support for h in app_macro |
| 13:31.44 | hmmhesays | i'm still on 7 someghing |
| 13:31.48 | hmmhesays | *something |
| 13:32.00 | benjk | so you can now have your dial macro do some actions in h inside the macro |
| 13:32.10 | hmmhesays | i see |
| 13:32.18 | blitzrage | yah... for some reason the later 8.x FW files didn't take... only got up to 8.0... although I should have just left it on 7.3 where it was working |
| 13:32.24 | hmmhesays | write a patch to write cdr's before extension h is called |
| 13:33.21 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:33.45 | benjk | well, my CDRs are no longer off now |
| 13:33.49 | *** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:34.03 | benjk | it sort of fixed itself as a result of handling the CANCEL properly |
| 13:36.08 | benjk | what was your issue with the CDRs again |
| 13:37.54 | jhamlyn | I set allowguest=yes and still find that there is a Authetication request failing on the connection... |
| 13:39.20 | *** join/#asterisk potsboy (n=chrisg@196.211.16.202) |
| 13:39.25 | inspired | anyone seen this? |
| 13:39.25 | inspired | Aug 10 15:31:59 WARNING[7742]: pbx_spool.c:346 scan_service: Unable to open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting |
| 13:39.25 | inspired | Aug 10 15:31:59 WARNING[7742]: pbx_spool.c:388 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/1.call' |
| 13:39.46 | inspired | 1.call belongs to user asterisk and group asterisk |
| 13:39.51 | inspired | and asterisk is running as asterisk.asterisk |
| 13:41.29 | [TK]D-Fender | inspired: : Possibly a problem with part of the path? |
| 13:41.46 | inspired | what do you mean? |
| 13:41.55 | [TK]D-Fender | jhamlyn: Also make sure their callerID does not match a peer entry.... |
| 13:42.08 | [TK]D-Fender | jhamlyn: (of the incoming caller). |
| 13:42.25 | jhamlyn | ok - follow the logic - |
| 13:42.55 | jhamlyn | I have been calling from one asterisk to another... assuming the sending party to be anonymous |
| 13:43.37 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:43.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:43.42 | Sonderblade | is it possible to make the voicemail phone menu not be password protected? |
| 13:43.56 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:43.57 | jhamlyn | exten => 303,1,Dial(Sip/201@otherasterisk.com) |
| 13:44.21 | jhamlyn | see the request come into the rxing machine and fails authetication.. |
| 13:44.59 | jhamlyn | The rxing asterisk knows nothing about the sending system.. no registrations, peers etc |
| 13:45.16 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:45.45 | jhamlyn | have context to rx any incoming calls... |
| 13:46.36 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 13:46.55 | *** join/#asterisk apardo (n=apardo@87.217.144.161) |
| 13:48.41 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-c9a84271cf8fff19) |
| 13:51.03 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 13:51.13 | *** join/#asterisk monkey13 (n=monkee13@69.7.217.155) |
| 13:55.04 | *** join/#asterisk juice (i=1000@mo-71-50-31-176.dhcp.embarqhsd.net) |
| 13:55.47 | sumasuma | is public calls allowed through fwd ?? |
| 13:55.57 | sumasuma | i mean calls from other domains ? |
| 13:56.28 | [TK]D-Fender | jhamlyn: Wel in "otherasterisk" you made a context, filled it with those extens, added "allowguest" and made sure that the callerID of the personall being sent over does not match a peer there? |
| 13:57.12 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 14:01.01 | *** join/#asterisk visba (i=user@16.sub-70-222-215.myvzw.com) |
| 14:02.43 | *** join/#asterisk hohum (n=dcorbe@12.195.58.236) |
| 14:07.12 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:07.57 | pigpen | Does anyone know if I can have an agi or app that will update a users vm pass to a database when they change their pass? |
| 14:09.30 | [TK]D-Fender | pigpen: Yes, yyou can have a trigger program called on change. |
| 14:09.46 | murf | sumasuma: as far as I know, no, fwd wasn't doing public calls anymore... too much abuse. |
| 14:10.51 | pigpen | [TK]D-Fender, could you give me a hint where to start...I have been google'ing for quite a bit....tks. |
| 14:12.12 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 14:12.25 | [TK]D-Fender | pigpen: Guessing you didn't look to hard... http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf |
| 14:12.37 | [TK]D-Fender | pigpen: "EXTERNPASS" |
| 14:12.51 | pigpen | err...sorry. Thanks though. |
| 14:12.58 | *** join/#asterisk stack_ (n=sgerstac@63.239.190.202) |
| 14:13.34 | stack_ | I have a Grandstream HT-386 that stops working after a while. It doesn't get a dial tone. If I reboot it, it works fine... Anyone here experience this? |
| 14:15.21 | [TK]D-Fender | stack_: Where is located relative to your * server? |
| 14:16.31 | stack_ | [TK]D-Fender: It goes HT -> Switch -> Server |
| 14:17.14 | [TK]D-Fender | stack_: Do you have "qualify=yes" for its entry? |
| 14:17.41 | xnon | hey friends i cant register mi FWD number in my asterisk |
| 14:17.43 | stack_ | [TK]D-Fender: no I don't... what does that do? |
| 14:17.56 | xnon | show me registration refused! |
| 14:18.55 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 14:20.28 | xnon | <PROTECTED> |
| 14:22.03 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 14:23.51 | *** join/#asterisk serg_b (n=sergey@9i.ru) |
| 14:23.54 | xnon | Registration of '791710' rejected: 'Registration Refused' from: '192.246.69.186' |
| 14:24.01 | xnon | :S why i cant |
| 14:24.24 | xnon | Error opening firmware directory '/usr/share/asterisk/firmware/iax': No such file or directory |
| 14:24.31 | *** part/#asterisk serg_b (n=sergey@9i.ru) |
| 14:24.46 | xnon | anybody can help me with my errors in console |
| 14:34.44 | *** join/#asterisk TW (n=anonymou@213.217.143.222) |
| 14:38.28 | *** join/#asterisk hensema (n=erik@scrat.hensema.net) |
| 14:40.27 | hmmhesays | bah, damn math |
| 14:42.41 | sumasuma | anybody does asterisk consulting for pay ? |
| 14:43.27 | *** join/#asterisk TeePOG (n=TeePOG@dsl-145-170-99.telkomadsl.co.za) |
| 14:43.43 | *** part/#asterisk TeePOG (n=TeePOG@dsl-145-170-99.telkomadsl.co.za) |
| 14:44.15 | queuetue | Is there perhaps a debugger for asterisk dialplans? So we can watch (and possibly interact) with call progress? |
| 14:45.04 | [TK]D-Fender | queuetue: Nope. Write it and watch it in CLI. Thats all we've got. |
| 14:45.16 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:45.26 | hmmhesays | hmm can you use math to do something like (1+2)/(3+4)? |
| 14:45.43 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:46.12 | hmmhesays | MATH($[1+2]/$[3+4]); doesn't seem to want to work |
| 14:47.42 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:49.33 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:49.51 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 14:51.33 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 14:52.12 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:52.23 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 14:53.22 | dlynes_laptop | Sounds like Digium will be growing even faster...they just got 13.8M in VC funding... |
| 14:54.01 | dlynes_laptop | http://news.tmcnet.com/news/2006/08/09/1781562.htm |
| 14:54.21 | *** join/#asterisk Trakkasure (n=Sgemtum@adsl-068-153-217-253.sip.bct.bellsouth.net) |
| 14:55.00 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 14:55.04 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
| 14:55.26 | rg1_ | need some help please on an "IF" statement |
| 14:55.31 | rg1_ | Here it is: |
| 14:55.32 | rg1_ | exten => s,n,Set(TQMUR_PROMPT_TO_PLAY=${IF($[${NUM_CONTINUE} = 1?${TQMUR_PROMPT_FIRST_REQUEST}]:${TQMUR_PROMPT_SUBSEQUENT_REQUEST}) |
| 14:56.09 | rg1_ | what is happening is that if the expression is true, the prompt_to_play IS getting set to the FIRST_REQUEST - but if it is not, it is not getting set at all |
| 14:56.17 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 14:56.22 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.232) |
| 14:56.32 | rg1_ | i'm thinking it might have something to do with the bracketing i am doing "[ ]" |
| 14:57.09 | queuetue | I have the following line sin my dialpla: http://pastebin.ca/125266 When I dial 603-878-XXXX, I see "LD: 1 detected." and get connected over my voip, as I would expect to. When I dial 603-235-ZZZZ, I get a fast busy. This behavior is repeatable. There is nothing I can discern in this dialplan that is specific to either number - why the heck does one work and the other not? |
| 14:57.27 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 14:58.02 | xnon | i have some problems autenticating my FWD account in my asterisk server and other things anybody can helpme this is my console http://pastebin.ca/125270 |
| 14:58.10 | file | queuetue: NXXNXXXXXX is what you want |
| 14:58.26 | rg1_ | can someone help me on that IF thing above? TQMUR... |
| 14:58.39 | queuetue | Oh! N must be special? Hitting docs. |
| 14:58.49 | queuetue | file: Thanks, once again. |
| 14:59.22 | file | xnon: did you setup your FWD account for IAX2, and wait? |
| 14:59.27 | hmmhesays | Aug 10 09:57:09 DEBUG[6219]: pbx.c:1589 pbx_substitute_variables_helper_full: Expression result is '0' why does asterisk do that when I have a float number smaller than 1 as a result? |
| 15:00.07 | xnon | file yes! |
| 15:00.23 | sumasuma | can anyone help me in configuring sipura |
| 15:00.27 | sumasuma | i can pay for the same ! |
| 15:00.36 | xnon | would u like see my extensions.conf and iax.conf? |
| 15:00.53 | hmmhesays | ok |
| 15:00.56 | hmmhesays | whats the issue? |
| 15:00.58 | file | xnon: if your FWD number is right, and your password is right... but the registration gets rejected, there's not a lot you can do about that |
| 15:01.17 | *** join/#asterisk fernando (n=fernando@unaffiliated/musb) |
| 15:01.20 | fernando | hi all |
| 15:01.22 | fernando | Aug 10 11:58:01 WARNING[2059]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3playe |
| 15:01.23 | hmmhesays | file: why is my math broken? |
| 15:01.44 | file | hmmhesays: "why is" are dangerous words |
| 15:02.00 | dlynes_laptop | sumasuma: I thought you said you were having a different problem? |
| 15:02.05 | xnon | file i dont know what must i do? |
| 15:02.12 | hmmhesays | why does asterisk return zero when my expression results in a floating point number |
| 15:02.25 | dlynes_laptop | sumasuma: something to do with agent queues, or something? |
| 15:02.35 | xnon | i was do all in the FWD page explain |
| 15:02.45 | xnon | http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 |
| 15:03.15 | hmmhesays | bah, i see now |
| 15:03.25 | file | xnon: there's not a lot you can do, I mean a registration line consists of a username, password, and the hostname - if all those are right, then something is not right on the other side... so talk to FWD |
| 15:03.31 | xnon | but the error go agead |
| 15:03.33 | xnon | ahead |
| 15:03.35 | rg1_ | can anyone help me with a "Set..... using an IF function? |
| 15:03.36 | xnon | Aug 10 09:58:55 NOTICE[10000]: chan_iax2.c:7438 socket_read: Registration of '791710' rejected: 'Registration Refused' from: '192.246.69.186' |
| 15:03.38 | sb_mx | rg1_, this works for me. i think it has to do with the way you set up your { and [ |
| 15:03.42 | sb_mx | rg1_, exten => *99999,1,Set(TEST=${IF($[${CALLERID(num)} = 1001]?1:0)}) |
| 15:04.06 | rg1_ | ok sb_mx - i will try that - thanks! |
| 15:04.28 | dlynes_laptop | xnon: your username or password is incorrect, or your account just plain doesn't exist |
| 15:04.45 | *** join/#asterisk apardo (n=apardo@87.217.144.2) |
| 15:05.10 | hmmhesays | bah, because expressions math operators don't support floating point numbers |
| 15:05.12 | hmmhesays | weeeeeeee |
| 15:05.24 | dlynes_laptop | ~mcc |
| 15:05.26 | jbot | from memory, mcc is the distribtution that started it all |
| 15:05.32 | queuetue | file: You may recall that last weekend, you patched and rebuilt asterisk on my server. Do you recall where you did it, in case I need to rebuilt it again and want to preserve the patch? |
| 15:05.51 | xnon | dlynes_laptop, my account is correct because i can enter with this user in the page |
| 15:05.55 | file | queuetue: #undef ZAPTEL_OPTIMIZATIONS in file.c |
| 15:06.06 | queuetue | file: Ok. |
| 15:06.10 | dlynes_laptop | xnon: but perhaps that account doesn't exist on their sip server yet |
| 15:06.45 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:06.52 | xnon | ummmmmm |
| 15:07.15 | xnon | dlynes_laptop, and so what can i do? |
| 15:07.32 | dlynes_laptop | make sure your sip context, sip username, and sip secret all match |
| 15:07.57 | dlynes_laptop | xnon: and if you're sure all of those match, phone up your ITSP's tech support |
| 15:07.58 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 15:08.10 | xnon | dlynes_laptop, must i type this user in my sip.conf? |
| 15:08.18 | dlynes_laptop | xnon: but whatever you do, don't tell them you're using asterisk |
| 15:08.27 | dlynes_laptop | xnon: or they'll refuse to help |
| 15:08.37 | xnon | ok |
| 15:08.57 | dlynes_laptop | [myusername] username=myusername ; secret=mypassword |
| 15:09.03 | rg1_ | sb_mx - yep, that did the trick - thanks ever so much!!!!!! |
| 15:09.17 | sb_mx | exactly, even carriers over here are like "what kind of pbx are you using.." and we say "an open-source linux based pbx". guess what they answer is next |
| 15:09.22 | sb_mx | rg1_, np |
| 15:09.41 | xnon | dlynes_laptop, in iax.conf? |
| 15:09.56 | hmmhesays | bah this sucks I have to call MATH function like 3 times to get this to work right |
| 15:10.05 | dlynes_laptop | xnon: or sip.conf, depending on what technology you're using to connect to your ITSP |
| 15:10.23 | dlynes_laptop | hmmhesays: write a function to handle it then |
| 15:10.28 | xnon | im using FWD |
| 15:10.29 | dlynes_laptop | hmmhesays: you know C, right? |
| 15:10.30 | xnon | http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 |
| 15:10.42 | dlynes_laptop | xnon: Yeah...I don't want to sign up for fwd |
| 15:10.56 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 15:10.58 | dlynes_laptop | xnon: but if you use it, you should know whether it supports iax or sip |
| 15:11.12 | rpm | bah my tdm400p is dieing. [6784658.785000] Power alarm on module 4, resetting!, i get a couple of those every day now |
| 15:11.12 | coppice | sb_mx: "why, so are we"? :-) |
| 15:11.18 | *** join/#asterisk mcreedjr (n=mcreedjr@oh-65-41-206-20.sta.embarqhsd.net) |
| 15:11.22 | rpm | unless its my power supply |
| 15:11.44 | mcreedjr | Is there any way to target a specific Zap channel with the flash command? |
| 15:11.50 | xnon | what ISP provider is recomended for agree in my asterisk server? |
| 15:11.54 | mcreedjr | err flash application even |
| 15:11.58 | hmmhesays | dlynes_laptop: somewhat, but... I can just call MATH a bunch of times |
| 15:11.59 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:12.29 | sb_mx | coppice, haha i wish.. they have always answered "ahhh.." as if to say its your problem . of course, after a while we slap em silly :P and make em see its their problem |
| 15:12.29 | dlynes_laptop | hmmhesays: yeah...just remember you telling someone yesterday that they should write an app, instead of writing an agi in c :) |
| 15:12.33 | xnon | dlynes_laptop, can you say me what SIP Provider can you recomended¿ |
| 15:12.48 | dlynes_laptop | xnon: there's plenty of people using fwd with asterisk...i'm sure it works |
| 15:12.55 | hmmhesays | dlynes_laptop: I was? |
| 15:13.10 | dlynes_laptop | hmmhesays: i think it was you...maybe my memory's bad though |
| 15:13.14 | dlynes_laptop | hmmhesays: it was in #asterisk-dev |
| 15:13.20 | xnon | dlynes_laptop, but why i cant so? |
| 15:13.22 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 15:13.29 | coppice | sb_mx: if you say avaya or nortel they will still say "aah". If you point out that smoke is billowing from their exchange building they will still say its your fault |
| 15:13.32 | hmmhesays | dlynes_laptop: wasn't me then, I haven't been in dev in awhile |
| 15:13.38 | dlynes_laptop | xnon: rephrase? your sentence does not compute |
| 15:13.46 | sb_mx | coppice, hahaha, so true |
| 15:13.47 | xnon | i dont understand all data is right in the scripts |
| 15:14.08 | dlynes_laptop | xnon: just make sure the part in between |
| 15:14.26 | xnon | the error is: Aug 10 10:09:45 NOTICE[10000]: chan_iax2.c:7438 socket_read: Registration of '791710' rejected: 'Registration Refused' from: '192.246.69.186' |
| 15:14.30 | dlynes_laptop | xnon: just make sure the part in between '[' and ']' is the same as the part after username= and that it matches your username |
| 15:14.43 | dlynes_laptop | xnon: also make sure that the part after secret= is the same as your password |
| 15:15.05 | dlynes_laptop | xnon: erm...actually....i'm half awake |
| 15:15.20 | dlynes_laptop | xnon: it's failing on your register => username:password@hostname line |
| 15:15.21 | xnon | ummm ok |
| 15:15.43 | xnon | register => 791710:123456789@iax2.fwdnet.net |
| 15:15.55 | xnon | this are |
| 15:16.01 | dlynes_laptop | xnon: you just told the whole world what your username and password are for fwd |
| 15:16.03 | xnon | are this or whatever jejee |
| 15:16.04 | dlynes_laptop | xnon: good job |
| 15:16.29 | dlynes_laptop | xnon: now everyone can use your account to make long distance calls |
| 15:17.15 | dlynes_laptop | xnon: but, yeah...that's the line |
| 15:17.22 | xnon | i dont care i have other accounts this account is only probe account |
| 15:17.23 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 15:17.38 | dlynes_laptop | xnon: you need to make sure that the username and password match whatever fwd gave you |
| 15:17.58 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:17.58 | dlynes_laptop | xnon: if they do match, and that account hasn't been cancelled, call up their tech support and complain |
| 15:18.00 | *** join/#asterisk Trakkasure (n=Sgemtum@adsl-068-153-217-253.sip.bct.bellsouth.net) |
| 15:18.52 | *** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net) |
| 15:19.00 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 15:19.25 | fourcheeze | and if it hasn't been cancelled yet, ask for it to be now |
| 15:20.15 | pigpen | hi all, I am trying to get externpass working with voicemail. I have: externpass=/usr/sbin/1.sh in my voicemail.conf file, which grabs $1 $2 $3 and echo's it to a file in /tmp. |
| 15:20.19 | pigpen | however it isn't working... |
| 15:20.37 | pigpen | I figure If I cannot get this to work, no sense in trying to have it update a database... |
| 15:20.38 | pigpen | ideas? |
| 15:20.46 | xnon | dlynes_laptop, the username that i use for enter from the website is pbxwvt |
| 15:20.52 | Idle | we have a ISDN PRI (t1) here in the office, going to a nortel PBX, what would be the easiest way to interface these so we can maintain our existing phones, and still add voip (sip) phones |
| 15:21.18 | dlynes_laptop | xnon: yeah, but that's not necessarily the same name as the one you use to connect, right? |
| 15:21.19 | xnon | in this line y need to type the number or the username? |
| 15:21.23 | TheCompWiz | Idle... when you le me know... share the wealth ;) |
| 15:21.34 | xnon | ummm let me see it |
| 15:21.34 | fourcheeze | pigpen: put some debugging/logging in your script |
| 15:21.41 | TheCompWiz | * when you know... let me know & share the wealth. (brain isn't working yet this morning) |
| 15:21.46 | Idle | ah |
| 15:22.01 | pigpen | fourcheeze, well, with the debugging I have done, it doesn't look like it is even calling it. |
| 15:22.06 | Idle | well, I was thinking get 2 T1 cards for asterisk, and use 1 t1 as a trunk, and the other out to the telco |
| 15:22.07 | TheCompWiz | I need (would like) to do the same thing |
| 15:22.20 | pigpen | debug level at 10 in asterisk...with no mention of the script running. |
| 15:22.27 | fourcheeze | pigpen: is it executable? |
| 15:22.28 | xnon | yes in the site i cant enter with fwd number and username |
| 15:22.33 | TheCompWiz | Idle... I'd be happy if I could find out how to get my pbx to talk to the nortel box. |
| 15:22.37 | xnon | is the same |
| 15:22.47 | TheCompWiz | but I know almost nothing about nortel systems. |
| 15:22.49 | Idle | :S well, a T1 is just a standard line, at both ends, no? |
| 15:22.54 | Idle | nor do I |
| 15:23.00 | pigpen | 777 in /usr/sbin .... which the directory is 755 |
| 15:23.03 | xnon | dlynes_laptop, i think that the error is the domain name! |
| 15:23.03 | Idle | its not even 'ours', its more of a clients |
| 15:23.21 | dlynes_laptop | xnon: maybe it's iax.fwdnet.net, and not iax2.fwdnet.net? |
| 15:23.22 | fourcheeze | pigpen:ok,when you run it from the command line does it behave as you expect? |
| 15:23.29 | pigpen | yes... |
| 15:23.36 | *** join/#asterisk dmz (n=dmz@64.151.98.180) |
| 15:23.38 | dlynes_laptop | xnon: i'd do some checking around on their website to find out what it is |
| 15:23.52 | TheCompWiz | Idle... I know with T1s... you need to configure 1 side to control the timing... and beyond that... no clue. I know how to setup data t1s ... point-to-point... but never really messed with telephone stuff until now. |
| 15:24.00 | [TK]D-Fender | pigpen: Are you running * as root? |
| 15:24.08 | pigpen | fourcheeze, it is a simple shell script.... just an echo -e "$1 $2$3" > /tmp/file.txt |
| 15:24.08 | Idle | they have a few things that they really hate about their telco... like it takes 1 month notice to setup a phone conference call... |
| 15:24.21 | pigpen | [TK]D-Fender, no...asterisk:asterisk |
| 15:24.22 | Idle | TheCompWiz: same |
| 15:24.38 | TheCompWiz | heh... why 1 month for phone conference? |
| 15:24.38 | [TK]D-Fender | pigpen: The how on earth do you expect it to have ANY rights to SBIN? |
| 15:24.39 | Idle | I know one side is the DCE and one is the DTE.. generally the DCE is the telco |
| 15:24.45 | Idle | TheCompWiz: yea, its bad |
| 15:24.51 | [TK]D-Fender | pigpen: .......... |
| 15:25.18 | Idle | TheCompWiz: I guess I need to find a T1 capable of the DCE |
| 15:25.20 | pigpen | [TK]D-Fender, drwxr-xr-x 2 root root 4824 Aug 10 10:00 sbin/ |
| 15:25.33 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 15:25.38 | fourcheeze | [TK]D-Fender: normally users normally have rights on sbin |
| 15:25.41 | [TK]D-Fender | pigpen: I suggest moving the file elsewhere for now |
| 15:25.48 | pigpen | k |
| 15:25.58 | fourcheeze | pigpen: can you su asterisk and still run the script? |
| 15:26.01 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 15:26.11 | gaspiz | hi, in my CDR table some calls have the src 't' did anyone experience this? |
| 15:26.54 | eKo1 | gaspiz: I have. |
| 15:27.06 | gaspiz | could you fix it? |
| 15:27.19 | eKo1 | No. I usually ignore those. |
| 15:27.28 | gaspiz | :) |
| 15:27.41 | gaspiz | nobody knows the answer on how to fix it? |
| 15:28.04 | gaspiz | I want to do some billing in the future and I can't with this bug |
| 15:28.10 | eKo1 | I think that is normal behaviour and hence not fixable. |
| 15:28.24 | eKo1 | Just ignore those records. |
| 15:28.36 | pigpen | fourcheeze, yes...I can su to asterisk and it works fine. |
| 15:28.47 | blitzrage | ok, so if I enable 't' in Dial(), then want to use the *2 in the featuremap of features.conf, after removing the default ; comment from the features.conf file, do I need a full restart to make it active? |
| 15:28.54 | pigpen | the funny thing is that in the asterisk cli, I see no error. |
| 15:29.13 | dlynes_laptop | blitzrage: shouldn't...should just be a simple reload |
| 15:30.54 | blitzrage | yah thats what I figured... but *2 does nada... |
| 15:31.13 | blitzrage | and 't' is to allow the called party to transfer... which I see... so... shoudl be good to go... |
| 15:31.17 | *** join/#asterisk Gregabyte (n=greg@gateway.digium.com) |
| 15:31.25 | blitzrage | can't get any transfer stuff to work at all on this version of trunk for some reason... |
| 15:31.51 | blitzrage | SIP transfers don't work, and Asterisk doesn't seem to respond to *2 for an attended transfer after uncommenting it from the feature map |
| 15:32.39 | dlynes_laptop | blitzrage: is it showing up in show features? |
| 15:33.00 | blitzrage | lets see |
| 15:33.15 | blitzrage | yep |
| 15:33.20 | blitzrage | shows it in the Current column |
| 15:33.23 | dlynes_laptop | blitzrage: then the reload worked |
| 15:33.28 | blitzrage | aye |
| 15:33.35 | blitzrage | let me double check to make sure the 't' flag is really really there :) |
| 15:33.44 | blitzrage | stupid stuff like that is always a pain :) |
| 15:33.45 | phearless | where is the doc to make ouside calls ? |
| 15:34.38 | dlynes_laptop | phearless: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 15:34.50 | dlynes_laptop | phearless: that's for both outside calls and internal calls |
| 15:34.59 | phearless | thank you i will have a look |
| 15:35.09 | dlynes_laptop | phearless: you might also want to read the channel specific documentation for iax2 and sip as well |
| 15:35.19 | dlynes_laptop | phearless: it's available on the same website |
| 15:35.23 | dlynes_laptop | ~docs |
| 15:35.25 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:35.31 | phearless | okay |
| 15:36.08 | *** join/#asterisk paryl (n=chatzill@209.236.78.59) |
| 15:36.13 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 15:37.39 | mcreedjr | Is there any way to flash a specific Zap channel? |
| 15:37.59 | [TK]D-Fender | phearless: Actually, supplemental to all that sounds like you shold read... THE BOOK |
| 15:38.00 | mcreedjr | I'm trying to get call forwarding to work from the phone |
| 15:38.01 | [TK]D-Fender | ~book |
| 15:38.03 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:38.27 | paryl | hey guys! i'm having an issue with an IAX2 connection... periodically one side of the conversation just goes to dead air, and it comes back within 3-10 seconds. i *just* found a real error on it (it's been happening for a long time, but i couldn't find the actual error in the logs)... it is "Aug 10 10:23:43 WARNING[22108] app.c: No audio available on IAX2/[ip censored]:4569-1??" |
| 15:39.03 | dlynes_laptop | mcreedjr: You can only flash on a connected zap channel |
| 15:39.21 | phearless | ok [TK]D-Fender ... damn this is a lot of stuff ... |
| 15:39.23 | dlynes_laptop | mcreedjr: so, yes, you can flash a specific zap channel (the one you're connected to) |
| 15:40.21 | paryl | any idea how on earth i can track the source of that error down? |
| 15:40.31 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:40.46 | [TK]D-Fender | phearless: Well you do seem to be only starting.... the book is the best guide from scratch out there, and the rest (wiki, etc) are more like reference material once you are ready to taget something more specific later |
| 15:41.42 | mcreedjr | dlynes_laptop: ermm.. so when a call comes in on a PSTN line, rings SIP extension and receives a 302 Moved Temporarily response, it forwards the call to the local extension. I want to flash the Zap channel there, but the active channel is Local. |
| 15:41.49 | mcreedjr | dlynes_laptop: Any idea how to do that? |
| 15:42.03 | phearless | [TK]D-Fender: a colleague just told me that he got a hard copy of it |
| 15:42.07 | phearless | thats great :) |
| 15:42.08 | TheCompWiz | anyone in here have a gxp-2000 with the current firmware (not beta) |
| 15:42.09 | mcreedjr | dlynes_laptop: I'm trying to do call forwarding with transfer. |
| 15:42.25 | mcreedjr | dlynes_laptop: err three way calling with transfer |
| 15:42.50 | mcreedjr | dlynes_laptop: since the Local context answers the Zap channel, shouldn't I be able to flash it somehow? |
| 15:43.40 | [TK]D-Fender | phearless: I have nice laser printers @ work and don't care to run copies of things like this for myself at will... |
| 15:44.06 | [TK]D-Fender | phearless: I've cleared entire forests single-handedly IMO.... |
| 15:44.36 | phearless | haha |
| 15:45.57 | [TK]D-Fender | Paperless society MY ASS! At 55ppm I'm PAPER-FULL in no time flat! |
| 15:46.40 | *** join/#asterisk salviadud (n=ralfalfa@201.135.2.253) |
| 15:46.42 | eKo1 | More like treeless society |
| 15:46.48 | pigpen | [TK]D-Fender, you sound like your are from Texas |
| 15:47.04 | salviadud | I reckon he is |
| 15:48.17 | blitzrage | he's not |
| 15:48.20 | blitzrage | he's a frog ;) |
| 15:48.21 | pigpen | well, I cannot get this dam externpass script to run to save my life at the moment... |
| 15:48.35 | pigpen | frog? |
| 15:49.19 | *** join/#asterisk monkey13 (n=monkee13@69.7.217.155) |
| 15:49.34 | blitzrage | pigpen: externpass=/usr/bin/php -q /etc/asterisk/include/externpass.php |
| 15:49.36 | blitzrage | for example |
| 15:50.36 | *** join/#asterisk aydiosmio (n=aydiosmi@65.213.70.43) |
| 15:50.39 | blitzrage | then there are 4 arguments passed to it from Asterisk: the script called, the vm_context, the extension, and the passwd |
| 15:51.11 | blitzrage | frog is a not so nice way of saying Quebecker |
| 15:51.14 | aydiosmio | anyone remember how to request a variable length of input digits terminated by a # key with $AGI->get_data? |
| 15:51.26 | pigpen | well, I have: externpass=/usr/sbin/externpass.sh which just is echo'ing the values to /tmp/test.txt |
| 15:51.52 | TheCompWiz | anyone using a granstream gxp2000? |
| 15:52.02 | blitzrage | I'm sure some people are |
| 15:52.06 | pigpen | all of which , asterisk is not showing that is it trying to run it, nor is the file being created....and all the perm's are fine, as I can su to asterisk and run it fine. |
| 15:52.34 | pigpen | Quebec eh? |
| 15:52.34 | blitzrage | and you're doing this via voicemail? |
| 15:52.37 | TheCompWiz | well.. I keep getting wierd problem... I dial any *XXX number and after the 2nd digit... the phone goes back to no digits being dialed. |
| 15:53.01 | blitzrage | sounds like it's matching on built in *XX numbers |
| 15:53.02 | TheCompWiz | (dosn't reboot... but current state resets or something) |
| 15:53.19 | TheCompWiz | as far as I know... there are no built-in *XX numbers in the phone... |
| 15:53.25 | TheCompWiz | or any built in numbers. |
| 15:53.34 | TheCompWiz | all config is menu-driven or via a web page. |
| 15:53.49 | pigpen | blitzrage, yes., i have it in the general section..,. |
| 15:53.54 | xnon | anybody have a fwd account configured in your asterisk iax.conf? |
| 15:55.43 | pigpen | TheCompWiz, I know the polycom has a digitmap section...maybe the grandstream does too... |
| 15:55.55 | TheCompWiz | nope. |
| 15:56.22 | xnon | anybody have a fwd account configured in your asterisk iax.conf? |
| 15:56.38 | TheCompWiz | define "fwd account"? |
| 15:56.47 | xnon | yes |
| 15:56.56 | TheCompWiz | maybe |
| 15:57.15 | xnon | i was do it! |
| 15:57.19 | xnon | but have errors |
| 15:57.30 | xnon | the error is Aug 10 10:52:51 NOTICE[10275]: chan_iax2.c:7438 socket_read: Registration of '791710' rejected: 'Registration Refused' from: '192.246.69.186' |
| 15:57.50 | TheCompWiz | because your registration information is incorrect. |
| 15:58.01 | xnon | no my information is correct |
| 15:58.07 | xnon | o was comprobed! |
| 15:58.10 | xnon | in the website |
| 15:58.21 | paryl | if i update asterisk to the most recent version, do i /have/ to update zaptel and libpri at the same time? |
| 15:58.31 | TheCompWiz | your registration was rejected... because your information is incorrect. don't argue. |
| 15:58.44 | paryl | i'm at libpri-1.2.2 and zaptel-1.2.5 |
| 15:58.44 | *** join/#asterisk Zalbag (n=hat_and_@adsl-156-148-145.mia.bellsouth.net) |
| 15:59.06 | xnon | i have other error relationed with this error posibily |
| 15:59.07 | xnon | Aug 10 10:37:51 WARNING[10261]: chan_iax2.c:1387 reload_firmware: Error openingfirmware directory '/usr/share/asterisk/firmware/iax': No such file or directory |
| 15:59.14 | xnon | firmware! |
| 15:59.30 | eKo1 | Crap, I just crashed * after an 'extensions reload' |
| 15:59.31 | xnon | :S i dont know why? |
| 15:59.32 | TheCompWiz | I doubt it's related. |
| 15:59.35 | blitzrage | xnon: not an error unless you're trying to reflash an IAXy |
| 15:59.42 | TheCompWiz | it sounds 100% like you have your box setup wrong. |
| 15:59.45 | *** part/#asterisk Zalbag (n=hat_and_@adsl-156-148-145.mia.bellsouth.net) |
| 16:00.38 | xnon | other error is: |
| 16:00.39 | xnon | Aug 10 10:37:51 WARNING[10261]: chan_iax2.c:9599 load_module: Unable to open IAX timing interface: No such file or directory |
| 16:00.52 | xnon | :( |
| 16:00.55 | blitzrage | you have no timing interface -- i.e. hardware or zt_dummy |
| 16:01.06 | blitzrage | thus you can't use trunking with IAX2 |
| 16:01.13 | xnon | i dont hardware only asterisk in my pc |
| 16:01.18 | *** join/#asterisk _deg_ (n=deg@200.181.137.62) |
| 16:01.34 | xnon | what can i do ? |
| 16:01.41 | blitzrage | use zt_dummy, like I said |
| 16:01.50 | blitzrage | or, turn off trunking in iax.conf |
| 16:02.13 | xnon | trunking context? |
| 16:02.19 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 16:02.21 | blitzrage | no.... |
| 16:02.37 | blitzrage | look for "trunking" in the iax.conf.sample file and read the description |
| 16:02.47 | [TK]D-Fender | xnon: Anywhere you use"trunk=yes" should be removed |
| 16:03.06 | blitzrage | guess [TK]D-Fender is more likely to hand-hold than I :) |
| 16:03.28 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 16:03.33 | xnon | i dont have trunk=yes text in my iax.conf |
| 16:03.59 | xnon | [TK]D-Fender, i dont have this text in all my iax.conf |
| 16:05.15 | [TK]D-Fender | xnon: Ok, then just ignore the message |
| 16:05.39 | xnon | but my fwd account cant register! |
| 16:05.45 | fourcheeze | [TK]D-Fender: apropos our discussion about a cloud of asterisks |
| 16:06.03 | fourcheeze | suppose I just did the following: |
| 16:06.16 | fourcheeze | sipusera calls sipuserb on server 1 |
| 16:06.30 | fourcheeze | if sipusera answers then all well and good |
| 16:06.37 | fourcheeze | if not server 1 tries sipuserb on server 2 |
| 16:06.50 | fourcheeze | if still no answer -> voicemail |
| 16:06.51 | eKo1 | How come the country us-old get's parsed as us-o in indications.conf. This must be a bug. |
| 16:06.58 | fourcheeze | [TK]D-Fender: does that sound completely mad? |
| 16:07.02 | eKo1 | err gets |
| 16:07.25 | nounoursfr | do you have test loadbalancing on asterisk ? |
| 16:07.48 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:08.43 | quid246 | noun: Load balancing or just load testing? |
| 16:09.28 | dlynes_laptop | mcreedjr: do you have 'link' capability on your line? |
| 16:12.09 | fourcheeze | [TK]D-Fender: so the general rule is that first the current server is tried, then the others in turn (just 1 other to start) and then to voicemail |
| 16:12.12 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 16:12.17 | fourcheeze | any reason why that wouldn't work? |
| 16:12.42 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 16:15.36 | *** join/#asterisk kevinfcn_ (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 16:21.29 | *** join/#asterisk nobell (n=jdegraff@70.103.228.158) |
| 16:21.41 | nobell | any body using hudlite? |
| 16:21.57 | *** join/#asterisk vlt (n=dm@p54B30197.dip0.t-ipconnect.de) |
| 16:24.16 | vlt | Hello. How can I SET ${EXTEN} to a new value (I need this to transform incoming calls with national code prefix). I tried "exten => _49.,1,SET(EXTEN=0${EXTEN:2})" but it doesn't work ... How is the correct synatx? |
| 16:27.04 | syzygyBSD | VLT: goto |
| 16:27.12 | *** part/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 16:27.35 | syzygyBSD | vlt: goto(${EXTEN:0},1) |
| 16:28.02 | vlt | syzygyBSD: Thank you, I'll try ... |
| 16:28.04 | syzygyBSD | or something.. you get the idea |
| 16:28.55 | *** join/#asterisk saftsack (n=saftsack@p54A7F3CB.dip.t-dialin.net) |
| 16:29.32 | saftsack | would it be smart to combinate asterisk with a patton 4552 bri switch? |
| 16:30.18 | vlt | syzygyBSD: Yes it works, thanks. |
| 16:30.19 | dos000 | anyone can explain the pro cons of using odbc instead of mysql ??? |
| 16:31.04 | vlt | Why is it not possible to define an exten rule with pio 1 and then with prio 3? Why is prio 2 needed? |
| 16:32.09 | fourcheeze | dos000: odbc means you can use any rdbms back end |
| 16:32.29 | aydiosmio | any db with an ODBC driver. |
| 16:32.49 | aydiosmio | and lol @ unixodbc and mssql on linux. |
| 16:33.12 | aydiosmio | that's a fricken trainwreck. |
| 16:33.20 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:33.25 | smackus | I am looking for the ability to make and take calls while logged into a queue. I have found that I can do that with AgentCallbackLogin. The problem is as I read the bugs on this command that it is potentially system crashing. http://bugs.digium.com/view.php?id=6626 Is there another way to do this? I want to be able to log into a queue at the beginning of the day, and log out at the end of the day. not have to log in after every time I cradle the phone lik |
| 16:33.51 | *** join/#asterisk _w^x_ (n=w^x@cpe-66-87-4-181.ut.sprintbbd.net) |
| 16:33.58 | nobell | mysql has come a long way in the last few years - and does most everything that any other major database system does. It's worth learning how to use it. |
| 16:34.00 | dos000 | in terms of stability are both libs the same when it comes to asterisk integration ? |
| 16:34.55 | ionix | stores procedure? |
| 16:34.57 | ionix | stored |
| 16:35.04 | [TK]D-Fender | smackus: User static members and use pause/unpause |
| 16:35.09 | fourcheeze | dos000: I don't think there's much to choose, but for the record I use odbc with mysql backend |
| 16:35.28 | smackus | [TK]D-Fender: ok, thank you. |
| 16:36.03 | smackus | so would static members mean that agents are assigned to the phone. cant just move around? |
| 16:36.11 | [TK]D-Fender | smackus: Yup. |
| 16:36.13 | smackus | ok |
| 16:36.22 | nobell | mysql 5.0 has stored procedures. |
| 16:36.34 | fourcheeze | finally |
| 16:36.38 | nobell | http://dev.mysql.com/doc/refman/5.0/en/stored-procedures.html |
| 16:36.43 | smackus | i really wish they would get this one fixed so I can have the best of both worlds. |
| 16:36.53 | fourcheeze | nobell: does it have foreign keys yet? |
| 16:38.01 | SkramX | Is there a female voice to be used with asterisk? |
| 16:38.04 | nobell | yes http://dev.mysql.com/doc/refman/5.0/en/example-foreign-keys.html |
| 16:38.06 | SkramX | such as festival? |
| 16:39.07 | nobell | it's also important to learn how to create indexes - so that your left joins run smoother. research mysql.org on how to optimize queries. I have been told that an optimized mysql database will run faster than oracle. |
| 16:39.52 | *** join/#asterisk tempest1 (n=asf@c-68-58-187-78.hsd1.sc.comcast.net) |
| 16:40.10 | hmmhesays | ok why is my TIMEOUT(absolute) not working? |
| 16:40.49 | nortex | SkramX, Check Cepstrail |
| 16:41.04 | nortex | I mean Cepstral |
| 16:41.18 | SkramX | nortex: Yeah.. I know, but no; I use them but this is for a friend that needs a FREE female voice |
| 16:41.30 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:43.24 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 16:45.32 | *** join/#asterisk sergee (n=opera@ppp83-237-59-184.pppoe.mtu-net.ru) |
| 16:45.57 | SkramX | so? |
| 16:46.43 | *** part/#asterisk sergee (n=opera@ppp83-237-59-184.pppoe.mtu-net.ru) |
| 16:47.25 | coppice | SkramX: i've never seen one. despite most of the world's TTS being based on Festival, very little ever seems to have been contributed back to it. |
| 16:47.36 | *** join/#asterisk sergee (n=opera@ppp83-237-59-184.pppoe.mtu-net.ru) |
| 16:47.43 | SkramX | coppice: indeed |
| 16:47.50 | SkramX | i use cepstral for my own projects |
| 16:49.34 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 16:49.48 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 16:50.36 | hmmhesays | no is there any way I can play a beep to the calling party while the call is active? |
| 16:50.44 | MRH2 | Hi anyone know what effect srtp has on bandwidth consumption - i am budgeting 90kbps for g711 and wondering what extra srtp might add |
| 16:51.37 | dlynes_laptop | MRH2: you might want to budget 100kbps for g711 before srtp |
| 16:51.45 | dlynes_laptop | MRH2: there's sip and tcp overhead as well |
| 16:52.17 | MRH2 | ok |
| 16:54.58 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 16:56.22 | MRH2 | so how much more is srtp likely to add to this? :) |
| 16:56.48 | sevard | Has anyone ever used an AudioCodes IAD? I have an MP-118FXS sitting here and every time it registers I get -- Got SIP response 481 "Call/Transaction Does Not Exist" back from <ip>, it happens for each registration and since there's a different sip client on each port I get it 8 times. |
| 16:57.19 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 16:57.20 | sevard | I've tried sip debug <peer> to try and isolate why it's throwing those messages but I can't track it down. |
| 17:01.35 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 17:01.46 | *** join/#asterisk apardo (n=apardo@87.217.146.52) |
| 17:01.59 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:04.09 | hmmhesays | anyone use option L in their cmd dial? |
| 17:04.39 | sergee | hmmhesays: i do |
| 17:06.02 | sergee | MRH2: you can plaay beep periodicaly with L() option to dial, to both - caller and calle |
| 17:06.05 | *** join/#asterisk kevinfcn_ (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 17:06.13 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:06.40 | sergee | oops :) wrong man |
| 17:06.41 | hmmhesays | serge if I wanted to limit a call to 60 seconds playing the warning every 5 seconds at 30 seconds left it would be Dial(SIP/foo,,L(60:30:5)) right? |
| 17:07.19 | hmmhesays | oops |
| 17:07.35 | hmmhesays | Dial(SIP/foo,,L(60000:30000:5000)) |
| 17:09.53 | wunderkin | doesnt really make sense to me to have that in ms |
| 17:10.31 | sergee | hmmhesays: exactly |
| 17:11.20 | _deg_ | Anybody here using trunk on a macosx ppc? |
| 17:12.02 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:12.06 | *** join/#asterisk abozanich (n=adamboza@dsl081-246-226.sfo1.dsl.speakeasy.net) |
| 17:12.11 | hmmhesays | so it is in milliseconds |
| 17:12.16 | hmmhesays | mine did not want to play |
| 17:12.17 | hmmhesays | at all |
| 17:12.39 | abozanich | Does anybody know who I should contact about a security problem w/ asterisk? |
| 17:12.39 | hmmhesays | <PROTECTED> |
| 17:12.48 | hmmhesays | what security problem? |
| 17:13.15 | abozanich | can't say at the moment |
| 17:13.21 | abozanich | is there a security contact? |
| 17:13.52 | angler | probably contact kpfleming@digium.com |
| 17:14.04 | abozanich | thanks. |
| 17:14.30 | MRH2 | ...as the irc bots just flag that address for a million spam messages... |
| 17:14.36 | dos000 | is there a way i can see the logs for odbc messages only .. i am starting * with -cvvvv and the logs fly too fast |
| 17:14.46 | file | MRH2: our spam filtering is really good :D |
| 17:15.02 | MRH2 | :) |
| 17:15.02 | angler | yes it is actually |
| 17:15.03 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 17:15.40 | sergee | you can redefine sounds with appropriate variables |
| 17:16.00 | file | angler: !!! |
| 17:17.36 | *** join/#asterisk viperdude (n=jon@195.74.96.117) |
| 17:18.08 | viperdude | hi, anyone had issues with soxmix not being able to find liblame.so.0 ? |
| 17:18.26 | Dr-Linux|work | fuck Israel |
| 17:18.32 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 17:19.09 | sergee | viperdude: find / -name "liblame.so*" |
| 17:20.10 | viperdude | sergee: its in /usr/local/lib and i am passing that through in LDFLAGS on the configure script but it stills complains it can't find it even though the configure script reports support for mp3 |
| 17:20.15 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.207) |
| 17:20.38 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 17:21.26 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 17:21.43 | vader-- | anyone here ever hook up asterisk through tdm2400p card to an elevator phone |
| 17:22.00 | vader-- | im having a problem where the box in the elevator places the call and then rings and hangs up |
| 17:22.06 | vader-- | someitmes it doesn't even ring |
| 17:22.15 | vader-- | asterisk is showing through the console everything is fine |
| 17:22.40 | vader-- | is there something i may need to adjust in my zapata.conf? |
| 17:25.38 | blitzrage | nub question: in svn trunk, where does it get the version from? I just updated and svn info gives me a different version than 'show version' in asterisk... |
| 17:25.39 | quid246 | anybody played around with AstCC? |
| 17:25.39 | sevard | hmmhesays: have you ever dealt with something like that? |
| 17:25.43 | hmmhesays | and my calling card application is complete |
| 17:25.56 | hmmhesays | sevard: two chicks at once? unfortunately not |
| 17:25.56 | Un1x | hmmhesays youre making a calling card agi script? |
| 17:25.59 | blitzrage | hmmhesays: congrats |
| 17:26.33 | sevard | hmmhesays: you need a million dollars for that |
| 17:26.36 | mut | nice |
| 17:26.40 | hmmhesays | Un1x: no agi |
| 17:26.44 | hmmhesays | all in the native dp |
| 17:26.45 | quid246 | hmmhesays: Does it allow simultaneous calling per card? |
| 17:26.45 | mut | post on rentacoder.com |
| 17:26.48 | Un1x | oh |
| 17:26.54 | mut | max $5000 bid for a voip billing prog |
| 17:26.54 | hmmhesays | quid246: yes |
| 17:27.12 | hmmhesays | but the math would be a little funkified |
| 17:27.19 | quid246 | hmmhesays: nice, I'm trying to figure that one out now... not the code, but the concept so I can't ger orbbed blind |
| 17:27.24 | quid246 | robbed |
| 17:27.38 | hmmhesays | why do you need simultaneous calls per card? |
| 17:28.09 | quid246 | because it's a possibility |
| 17:28.27 | quid246 | What if a user wants to three wy? |
| 17:28.35 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 17:28.41 | quid246 | a call I mean ;) |
| 17:28.44 | hmmhesays | buy another freaking card |
| 17:28.58 | hmmhesays | users always complicate things |
| 17:29.24 | quid246 | rentacoder - the outsourcing whorehouse |
| 17:29.25 | quid246 | haha |
| 17:29.45 | hmmhesays | you'd need some seroius coding to do the time calculations accurately on a multi user card |
| 17:29.58 | hmmhesays | *serious even |
| 17:30.38 | MRH2 | whorehouse? they are young students who need the money ;) |
| 17:30.51 | blitzrage | !!! |
| 17:31.02 | blitzrage | I whore myself out on a consulting basis :) |
| 17:31.14 | MRH2 | are you pretty? |
| 17:31.19 | Qwell | MRH2: very |
| 17:31.22 | blitzrage | LOL |
| 17:31.26 | blitzrage | Qwell: shush you :) |
| 17:31.29 | MRH2 | lol |
| 17:31.43 | blitzrage | I meant asterisk consulting, not... that kind of consulting |
| 17:31.50 | file | silly blitzrage |
| 17:32.08 | blitzrage | file: so transfers work on this OTHER box... just not on that one particular rev... |
| 17:32.21 | MRH2 | but it still comes with a a happy ending i am sure |
| 17:32.28 | *** join/#asterisk evilbit (i=hhoffman@gateway/tor/x-f548ff6adeea47d2) |
| 17:32.33 | Qwell | MRH2: that costs extra |
| 17:32.49 | blitzrage | did anything since 38826 change in re: to SIP transfers? |
| 17:32.54 | blitzrage | MRH2: lol |
| 17:33.02 | file | blitzrage: commit list :P |
| 17:33.08 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 17:33.14 | evilbit | hi, what type of wav file is compatible with *? I'm d/l'ing a prompt from digium and am not sure what format to use |
| 17:33.29 | blitzrage | file: yah, I was looking there... but wasn't sure how to filter out all the other branches |
| 17:34.44 | blitzrage | aha... think I got it now... |
| 17:35.21 | *** join/#asterisk kevinfcn_ (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 17:36.05 | [TK]D-Fender | file, blitzrage : ! ! ! |
| 17:36.15 | file | zomg hi |
| 17:36.29 | quid246 | I've seen one provider do the following... they freeze 2 hours of credit (based upon call cost) from your account, then relinquish anything after the call is complete (and so on for the second call)... thing is I don't know how to handle what would happen if one went over the time limit |
| 17:36.40 | blitzrage | [TK]D-Fender: I don't want to meet your mom! |
| 17:37.02 | [TK]D-Fender | blitzrage: : I just wan.... |
| 17:37.09 | blitzrage | ! ! ! |
| 17:37.11 | nortex | Is there a way to play a tone to both ends of a call after connect? I have used the A option in my dial statement to play a sound file after the callee picks up but I want the caller to hear a tone too. |
| 17:37.13 | [TK]D-Fender | (cue file) |
| 17:37.17 | file | bang bang bang! |
| 17:37.21 | [TK]D-Fender | :D |
| 17:37.26 | evilbit | ah, n/m |
| 17:37.51 | blitzrage | lol... "Really destroying SIP dialog..." soooooo many! :) |
| 17:38.02 | *** join/#asterisk EvilDeshi (n=SkunK@oxford-bb-occam9-ws-25.dsl.maqs.net) |
| 17:40.01 | *** join/#asterisk saftsack (n=saftsack@p54A7D9ED.dip.t-dialin.net) |
| 17:41.54 | blitzrage | wow... 8.0 FW for 7960 is useless! |
| 17:42.09 | charles___ | blitzrage: get 8.3 |
| 17:42.20 | blitzrage | gotta get my soekris working with astlinux sometime so I can get a tftp server setup to upgrade it |
| 17:42.32 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 17:42.32 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:42.49 | charles___ | lito |
| 17:44.05 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 17:44.26 | sb_mx | hey guys, is there a way to concatenate commands? ie: exten => s,n,Read(VAR,Playback(audio1&audio2)) ? since Read wont allow audio1&audio2 syntax |
| 17:45.05 | charles___ | sb_mx: no |
| 17:45.22 | charles___ | sb_mx: you can playback while reading already |
| 17:45.27 | charles___ | sb_mx: what do you want to do ? |
| 17:45.42 | blitzrage | sb_mx: no, but you can use sox to create a new file that is a concatonation of those two files |
| 17:46.09 | sb_mx | blitzrage, yah, i've done that in the past. was wondering if there was an "easier" way of doing it |
| 17:46.42 | sb_mx | charles___, basically ask the user for a password by playingback more than 1 audio file since that's how our audio setup is atm |
| 17:47.12 | charles___ | sb_mx: just concatenate the audio cat audio1.gsm audio2.gsm >fullaudio.gsm |
| 17:47.22 | charles___ | sb_mx: cat may not work, I will recomend sox |
| 17:47.49 | smackus | ok, so i have two phones logged in using addqueuemember(queuename) but only one of them ever gets the calls. even if it is on the phone. if the phone is offhook, i dont want another queued call to be delivered to it. what could i be doing wrong? |
| 17:47.51 | sb_mx | charles___, yup, will do just that (or i might just get my hands dirty and enable Read's concat syntax?) |
| 17:47.56 | [TK]D-Fender | ... |
| 17:48.04 | mut | dcc exploit |
| 17:48.08 | *** part/#asterisk nobell (n=jdegraff@70.103.228.158) |
| 17:48.11 | mut | isn't that like.. old.. |
| 17:48.22 | charles___ | sb_mx: yes go ahead, it's going to be very usefull for yourself |
| 17:48.26 | [TK]D-Fender | mut L just got the /msg? |
| 17:48.38 | mut | ya just like everyone else |
| 17:49.02 | sb_mx | charles___, might be usefull for someone else too. anyway thanks for the help |
| 17:49.06 | charles___ | sb_mx: no one uses that |
| 17:49.21 | *** join/#asterisk Ixitxachitl (n=m@209.151.130.10) |
| 17:49.36 | charles___ | sb_mx: no need to open two or more fd while sox do the job |
| 17:49.56 | blitzrage | suppose someone doesn't already have 8.3 FW for a Cisco on a TFTP server they could let me use for a few minutes to upgrade this phone? |
| 17:50.34 | sb_mx | charles___, yeah i understand that. i was just wondering why playback and background have that option and read doesnt. of course, playback/background and read have different functionality |
| 17:50.36 | charles___ | blitzrage: that person can get arrested, their hands put in a dark wall and they will get shot in the back |
| 17:50.50 | quid246 | anybody still play around with AstCC? Everything works but the Generate Card option... Apache gives me a header error in the logfile |
| 17:51.14 | blitzrage | charles___: I think Cisco gives away the firmware now |
| 17:51.28 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 17:52.09 | quid246 | haha, I doubt that... Cisco would charge you for the light that a 7960 gives off, if they could figure a way |
| 17:52.24 | quid246 | blitz: If they give it away, go get it from them |
| 17:52.29 | Idle | haha |
| 17:52.37 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 17:52.38 | Idle | cisco doesn't give anything away |
| 17:53.11 | quid246 | Idle: That's not true... I had a pretty good free lunch at a free Cisco seminar once. |
| 17:53.22 | nortex | blitzrage, The version 8.2 is avaliable for download, but not 8.3 |
| 17:53.26 | nortex | http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 |
| 17:53.34 | blitzrage | nortex: ahhhh, ok, I knew one of them was |
| 17:53.50 | [TK]D-Fender | quid246: Let us know if they gie you your Immortal Soul back ;) |
| 17:53.53 | Idle | quid246: how much did you pay for the seminar... I guarentee it was paid from that |
| 17:54.03 | blitzrage | quid246: because I don't have a tftp server setup, and don't really have the time right now, so figured if someone already had it up and running I'd try and save some time -- and if you don't already know me, feel free to ignore me |
| 17:54.06 | quid246 | Idle: Read my comment again. :) |
| 17:54.07 | Idle | well, not you, but how much did your seat cost whomever |
| 17:54.15 | quid246 | oh no, it was *totally* free |
| 17:54.18 | Idle | :P |
| 17:54.22 | Idle | craz |
| 17:54.24 | Idle | y |
| 17:54.28 | quid246 | I don't work for a company... I just singed up |
| 17:54.29 | quid246 | signed |
| 17:54.59 | charles___ | blitzrage: lito, tftp server setup ? tftp dgram udp wait nobody /usr/sbin/tcpd in.tftpd |
| 17:58.00 | watchy | yay at&t is giving me a /21 |
| 17:58.10 | Idle | crazy |
| 17:58.17 | Idle | how many servers do you have |
| 17:58.20 | watchy | 4 |
| 17:58.24 | Idle | ... |
| 17:58.24 | watchy | i run a wireless isp |
| 17:58.29 | Idle | a /21 for 4? |
| 17:58.33 | Idle | oooh |
| 17:58.35 | Idle | :) |
| 17:58.36 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 17:59.17 | sevard | watchy: wherein? |
| 18:00.20 | watchy | south arkansas |
| 18:02.22 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 18:03.11 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:03.19 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:03.26 | Shaun2222 | watchy: why dont you get ip space directly from arin? |
| 18:03.43 | watchy | Shaun2222: no idea, i dont really handle that should i? |
| 18:04.01 | Shaun2222 | why get it from a upstream when you can get it from the source :) |
| 18:04.31 | Shaun2222 | watchy: from a abuse and management standpoint it's better... but it's more difficult and lengthy to get ip space from arin. |
| 18:04.41 | *** join/#asterisk [hC] (n=hardcore@190.10.9.191) |
| 18:05.55 | dos000 | darn ... how do you normally tell asterisk where zaptel,odbc include headers are installed ? mine are located in a different location |
| 18:05.59 | Shaun2222 | course your running a wireless isp, so not like you cant just renumber easily... with hosting it's more of a pain in the ass to renumber all customers :) |
| 18:07.02 | dos000 | mind you autoconf is only nice .. to the user ! |
| 18:09.07 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 18:09.10 | watchy | Shaun2222: you know anything about running an isp? |
| 18:10.41 | *** join/#asterisk hohum (n=dcorbe@12.195.58.194) |
| 18:14.31 | xnon | i have some warnings in my console asterisk anybody can help me? |
| 18:14.40 | xnon | anybody have a time for help me? |
| 18:15.08 | docelmo | Aparently not much.. |
| 18:15.20 | xnon | http://pastebin.ca/125507 |
| 18:15.26 | xnon | this is my asterisk console |
| 18:16.58 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 18:18.39 | clyrrad | anyone know of a command like show queues that will show the FULL name of all queues? |
| 18:18.58 | watchy | u know what i hate i got a client that complains about clipping over a wireless connection |
| 18:19.09 | watchy | but theyont find out if the people local to it experience clipping |
| 18:19.11 | *** join/#asterisk kevinfcn_ (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 18:19.20 | watchy | so i dunno where to fucking start lookin |
| 18:20.00 | watchy | anyone here ever use QOS for voip? |
| 18:21.23 | *** join/#asterisk _MDC_ (n=marcus@c-6efde255.06-72-6c6b7013.cust.bredbandsbolaget.se) |
| 18:21.42 | javar | xnox, what do you need? |
| 18:23.01 | _MDC_ | I've got trouble hearing background noice in one end, using alaw codec, is there any more option I could change? This happens both in SIP and ZAP |
| 18:26.08 | *** join/#asterisk c4t3l (n=c4t3l@72.16.250.149) |
| 18:27.22 | c4t3l | has anyone ever had problems using ADT alarm system => asterisk => ADT call center? |
| 18:27.44 | c4t3l | the system dials out but ADT can't receive data |
| 18:27.55 | watchy | data dont send worth shit over voip man |
| 18:28.06 | watchy | you cant tivo over it either |
| 18:28.15 | watchy | just hook it straight up to analog |
| 18:28.32 | c4t3l | going out over PRI |
| 18:28.38 | *** join/#asterisk _eMAC_ (n=YMironek@220-19-207-82.pool.ukrtel.net) |
| 18:28.56 | watchy | i dont think its gonna work |
| 18:28.56 | c4t3l | the previous alarm ran through an adtran IAD |
| 18:29.05 | c4t3l | i agree with you watchy |
| 18:29.10 | watchy | i never could get my tivo to work it sucks |
| 18:30.05 | c4t3l | should I just give up and get an analog line? |
| 18:30.20 | watchy | well u might wanna ask a little more in here but fax dont really work worth shit |
| 18:30.37 | watchy | and from what i understand its next to impossible to send computer data over voip |
| 18:30.56 | c4t3l | my provider seems ok generally. I use Cbeyond |
| 18:31.11 | watchy | well ive gone directly from ata to pstn |
| 18:31.15 | watchy | it still didnt work for me |
| 18:31.44 | c4t3l | watchy, which car did you use |
| 18:31.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:32.09 | watchy | 2 tdm400p analog cards, using sipura ATA |
| 18:32.21 | c4t3l | yep, same here |
| 18:33.29 | c4t3l | i've been going crazy trying to get this to work |
| 18:34.04 | *** join/#asterisk Mike (n=mike@201.112.50.158) |
| 18:34.36 | watchy | yea me to i needed my tivo to sync |
| 18:34.37 | watchy | heh |
| 18:34.47 | *** part/#asterisk e-MAC (n=YMironek@220-19-207-82.pool.ukrtel.net) |
| 18:36.16 | charles___ | usa Alaw |
| 18:36.27 | charles___ | watchy: it should work over g.711 |
| 18:37.10 | c4t3l | so allowing alaw should do it?? |
| 18:37.38 | watchy | charles: at like 28,8k +? |
| 18:37.38 | hmmhesays | and my calling card dialplan is finished |
| 18:37.53 | hmmhesays | NO AGI, woot |
| 18:38.59 | clyrrad | is there a command that will return the FULL name of all Queues? |
| 18:39.10 | clyrrad | or an application? |
| 18:42.27 | fernando | anyone have x-lite audio (/dev/dsp) problem? I'm using ubuntu 6.06. |
| 18:42.46 | [TK]D-Fender | clyrrad: Check out the AMI optiosn. Maybe in there |
| 18:44.43 | clyrrad | hey TKD :) |
| 18:45.00 | clyrrad | I found a Manager Command QueueStatus |
| 18:45.03 | clyrrad | just not sure how to use it |
| 18:45.18 | clyrrad | its shown here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+QueueStatus |
| 18:45.19 | c4t3l | now I have to wait for this guy to test his alarm |
| 18:45.19 | *** part/#asterisk unlord (n=nathan@va-65-41-104-14.dyn.embarqhsd.net) |
| 18:45.21 | c4t3l | :( |
| 18:46.13 | Ixitxachitl | hmm, once zaptel is compiled, im left with UNCONFIG status in zttool/CLI (zap show status) for my TE205P card...how do i 'configure' it? |
| 18:46.44 | c4t3l | very strange... Asterisk shows the call being dialed,the hand-off and the connection status |
| 18:46.55 | c4t3l | the data is just not getting through |
| 18:47.11 | Un1x | anyone know some good places |
| 18:47.14 | Un1x | to get toll free dids? |
| 18:47.48 | c4t3l | set up some different codecs in sip.conf. now I'm just wating for this guy to finish up before we can test |
| 18:48.34 | c4t3l | I hate wating >:| |
| 18:48.43 | *** join/#asterisk imdabest (n=imdabest@202.147.186.58) |
| 18:48.52 | imdabest | hello roo |
| 18:48.54 | imdabest | room |
| 18:48.55 | imdabest | :) |
| 18:49.07 | Un1x | anyone know some good palces to get 8** dids ONLY! |
| 18:49.08 | Un1x | ? |
| 18:49.54 | _MDC_ | OK, seems like this background noise is a feature - is there a way to turn this off? |
| 18:51.16 | imdabest | hi i am having trouble with inbound calls is there any one can help me out |
| 18:51.55 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 18:52.17 | GerbilWrk | Anyone know of a cheap place online to order ATA deviecs, like the linksys pap2? |
| 18:54.37 | hmmhesays | Sevard you're a little biatch |
| 18:55.14 | [TK]D-Fender | GerbilWrk: Depends where you are |
| 18:55.16 | *** join/#asterisk momelod (n=momelod@Toronto-HSE-ppp3893206.sympatico.ca) |
| 18:55.20 | momelod | hey people |
| 18:55.22 | sevard | hmmhesays: lick my balls |
| 18:55.28 | momelod | is there a gsm player for linux |
| 18:55.33 | momelod | or like a codec? |
| 18:55.44 | *** join/#asterisk zedkatuf (n=audela@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 18:55.51 | justinu|laptop | sevard: making friends? :P |
| 18:55.54 | GerbilWrk | [TK]D-Fender, US |
| 18:56.11 | sevard | justinu|laptop: no idea what provoked that, i think he's been doing drugs |
| 18:56.11 | *** join/#asterisk kevinfcn_ (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 18:56.12 | hmmhesays | hahaha |
| 18:56.15 | [TK]D-Fender | GerbilWrk: www.voipsupply.com www.atacomm.com www.voxilla.com |
| 18:56.21 | hmmhesays | on teh drugz |
| 18:56.34 | GerbilWrk | [TK]D-Fender, awesome, thanks |
| 18:56.37 | justinu|laptop | heh |
| 18:56.38 | hmmhesays | sevard you need to test out my calling card app now |
| 18:56.43 | sevard | NOW |
| 18:56.47 | sevard | like a fucking 5 year old |
| 18:56.49 | sevard | where is it |
| 18:56.57 | hmmhesays | you want the dialplan? |
| 18:57.03 | hmmhesays | bah, nevermind |
| 18:57.09 | sevard | I'd have to turn on the non-production servers |
| 18:57.14 | hmmhesays | i'm going to post it on the lost packet with the database schema |
| 18:57.22 | sevard | meeeeeeeh, they're all the way across the room, meeeehhhhh |
| 18:58.00 | sevard | hmmhesays: send me something to eat |
| 18:58.10 | sevard | all i got is this god damned awful ramen |
| 18:58.47 | justinu|laptop | hmmhesays: there's a website that will fedex overnight ppl dogshit for you |
| 18:58.51 | justinu|laptop | :) |
| 18:59.10 | hmmhesays | the woman made me breakfast the other day, then she gave it up |
| 18:59.12 | Supaplex | or if your neighbor complies, you'll get it free |
| 18:59.13 | hmmhesays | it was a good day |
| 18:59.20 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 18:59.20 | eBody | hey guys how can a tdm400 come w/ more than 1 fxo mod if there are only 4 input jacks? |
| 18:59.23 | hmmhesays | I'm wondering what she wants me to buy her |
| 18:59.31 | *** join/#asterisk jeebusmobile (n=jeebusmo@12.180.154.130) |
| 18:59.58 | *** join/#asterisk dapatrick (n=dapatric@dsl253-031-098.phl1.dsl.speakeasy.net) |
| 19:00.07 | sevard | justinu|laptop: sweet, where |
| 19:00.25 | dapatrick | I have a quick question - how would I configure a zap channel to do *nothing* on incoming calls. |
| 19:00.30 | dapatrick | That is, just ignore them. |
| 19:00.39 | sevard | don't configure it. |
| 19:00.48 | file | eBody: huh? |
| 19:01.12 | eBody | file, u know the tdm400 digium card.... |
| 19:01.24 | dapatrick | I need to be able to use it for outgoing. |
| 19:01.32 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 19:01.33 | file | eBody: yes, you can put 4 modules into it - and each module corresponds to a jack |
| 19:01.54 | sevard | dapatrick: send it into a context that just gives instant congestion and hangup |
| 19:02.11 | eBody | so it's not like a tdm2400 fxo mod that supports 4 lines per mod?? |
| 19:02.12 | dapatrick | Yes, I'm sort of doing that now. |
| 19:02.26 | file | eBody: the TDM400 modules are 1 per mod |
| 19:02.49 | dapatrick | The intricacy there is that there is another device that answers on that channel/line (a Windows server running GFI fax). |
| 19:03.12 | eBody | oh! so they are different than the fxo mods i have in my tdm2400?? |
| 19:03.17 | clyrrad | Anyone know how to use the Manager Command QueueStatus? I cant find it documented anywhere |
| 19:03.25 | eBody | because thos fxo mods support 4 lines each |
| 19:03.29 | file | the ones for the TDM2400 fit more circuitry onto each single module |
| 19:03.31 | sevard | eBody: he's said that over and over. |
| 19:03.38 | justinu|laptop | sevard: can't find it anymore... the man must have shut them down |
| 19:03.41 | eBody | thank you. |
| 19:03.42 | dapatrick | Right now, I'm sending the calls on that channel to a context that does Wait(120), then Hangup. |
| 19:04.01 | sevard | justinu|laptop: proably for heath hazord reasons and other such nonsense |
| 19:04.07 | sevard | hazard |
| 19:04.11 | justinu|laptop | nonsense indeed |
| 19:04.25 | sevard | dapatrick: why are you waiting so long? |
| 19:04.34 | sevard | dapatrick: just CONGESTION, Hangup |
| 19:04.40 | sevard | bam bye call |
| 19:05.31 | dapatrick | Okay, but let's say I CONGESTION hangup, that will hang up the call, perhaps before the the Fax server can answer it. |
| 19:05.56 | sevard | ..... i thought you wanted no incoming calls |
| 19:06.12 | dapatrick | I don't want to get rid of the call entirely, I just don't want Asterisk to answer the calls on this particular analog line. |
| 19:06.36 | c4t3l | dapatrick, then don't route the line through asterisk |
| 19:07.09 | _MDC_ | OK, small question; is silence_suppression on or off by default? |
| 19:07.20 | dapatrick | Yes, that's what we were doing previously, but we need to be able to use the line for outgoing calls when it is not in use for faxing. |
| 19:07.39 | Lyfe | would anyone have a link to a page describing the best recommended way to implement queues & agents? I've been looking around at stuff, and am rather confused by that there seems to be a lot of outdated stuff all over, incluing on the voip-info wiki. |
| 19:07.56 | Lyfe | rather, use queues & agents. |
| 19:08.42 | sevard | dapatrick: so set it up like this, line from telco -> fax machine -> fax machine line out -> fxo card, and set your fax to answer on the first ring |
| 19:09.08 | dapatrick | sevard: Genius. |
| 19:09.11 | sevard | you'll still be able to make outbound calls via * but yourfax will pick up calls before your * does |
| 19:09.29 | dapatrick | Excellent solution. Thank you very much! |
| 19:09.33 | dapatrick | So simple and elegant. |
| 19:09.40 | sevard | see fuckers? i'm smrt 2. |
| 19:09.40 | dos000 | anyone tried compiling asterisk with unixodbc and zaptel headers located in a non standard place ? |
| 19:10.04 | c4t3l | have any of you ever tried to route ADT alarm through * ? |
| 19:10.18 | c4t3l | alarm system that is... |
| 19:10.28 | dos000 | why ??? |
| 19:10.32 | Idle | I have 2 fxo cards, and 2 fxs cards... one line was used for faxing before, but now I have them both auto-bridge with my second fxs module... its pretty neat stuff dapatrick |
| 19:10.41 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 19:10.54 | sevard | Idle: explain this auto bridging |
| 19:11.14 | dapatrick | How does that work Idle? |
| 19:11.14 | Idle | its on the wildcards |
| 19:11.23 | c4t3l | customer doesn't want to pay for another analog line... been fighting with this for a few hours |
| 19:11.46 | sevard | c4t3l: can you hook it up to an ATA? |
| 19:11.53 | sevard | however stupid that may be |
| 19:11.53 | c4t3l | it is now |
| 19:11.58 | Idle | exten => fax,1,Dial(Zap/3) |
| 19:12.00 | [TK]D-Fender | dapatrick: Not a great idea... set a context for that Zap channel that has an s,1,Hangup on it. |
| 19:12.04 | c4t3l | i knowm i know |
| 19:12.48 | [TK]D-Fender | dapatrick: Don't take chances with the machine answering first.... thats an accident waiting to happen. |
| 19:12.48 | sevard | c4t3l: why don't you put the alarm system on the analog line and run all their voice on the ata? |
| 19:13.11 | dapatrick | Hmm. |
| 19:13.13 | c4t3l | its a cellular back-up line :( |
| 19:13.14 | *** join/#asterisk Manipura (n=chatzill@S01060011954c9c46.cg.shawcable.net) |
| 19:13.24 | Manipura | hello |
| 19:13.27 | Idle | faxdetect=both in your zapata.conf then itll dial the 'fax' extension on the context |
| 19:13.34 | dapatrick | Okay, let me weight the options a bit. |
| 19:13.36 | [TK]D-Fender | sevard: SMRT ..... you forgot ass++ <- |
| 19:13.46 | [TK]D-Fender | :D |
| 19:14.04 | c4t3l | i feel the flames a' comin' |
| 19:14.04 | Idle | dapatrick: mine accepts fax'es on both of my lines, as well as making proper phone calls |
| 19:14.12 | sevard | [TK]D-Fender: <homer> I am smart, s,m,r,t |
| 19:15.05 | c4t3l | i've set allowable codecs to alaw only in the ATA's peer listing |
| 19:15.13 | Idle | http://www.voip-info.org/wiki-Asterisk+fax#Zapfaxdetection |
| 19:15.37 | dapatrick | Idle, but you're not using a normal fax machine, are you? |
| 19:15.51 | dapatrick | Or are you, and it's sitting at the end of an extension? |
| 19:15.52 | c4t3l | now i'm waiting (ever so impatiently) for the owner to test the damn alarm again |
| 19:15.54 | Idle | yes I am |
| 19:15.57 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:16.06 | sevard | he's using a normal fax machine but also using an extra fxs port |
| 19:16.07 | Ixitxachitl | hmm, is there somewhere i can check the reasons for getting a yellow alarm after trying to set up zap span |
| 19:16.07 | Idle | its some mamoth HP all in one device |
| 19:16.16 | Idle | yea, thats the downside |
| 19:16.19 | sevard | where my solution frees up a port |
| 19:16.36 | sevard | i didn't know about faxdetec tthough |
| 19:16.40 | Idle | if you do that, itll take 2 rings to even start ringing on asterisk |
| 19:16.42 | c4t3l | zttest?? |
| 19:16.43 | sevard | i have a good use for that |
| 19:16.53 | Idle | depending on the fax machine |
| 19:17.10 | c4t3l | why don't you switch to iaxmodem??? :D |
| 19:17.18 | Idle | gross |
| 19:17.22 | Idle | fax on voip still sucks |
| 19:17.30 | puppet | idle: no it all depends |
| 19:17.34 | Idle | not that its an especially good protocol to begin with |
| 19:17.38 | Idle | puppet: true |
| 19:17.41 | c4t3l | iaxmodem has a buffer!! |
| 19:18.02 | Idle | but its still a dumb idea IMHO... I would rather have a fax machine or fax modem recieve it, then email it off |
| 19:18.03 | sevard | Idle: I have about 5 sipura 2002s at different locations all running fax machines, sip to my * box then out of my tdm |
| 19:18.17 | c4t3l | fax is based on old ass telegrapghy!! I can't beleive its still around |
| 19:18.17 | Idle | sevard: decent |
| 19:18.19 | sevard | Idle: I agree whoeheartedly, fax is retarded, why pay when you can email. |
| 19:18.27 | Idle | yep |
| 19:18.29 | sevard | c4t3l: exactly |
| 19:18.35 | Idle | not to mention its unreliable |
| 19:18.40 | sevard | most fax machines nowadays scan as well as fax anyway |
| 19:18.48 | Idle | I get fax spam... makes me cry |
| 19:18.57 | sevard | no shit, unreliable even on good analog phone lines |
| 19:19.02 | sevard | i thought fax spam was illegal |
| 19:19.06 | Idle | its funny, sometimes fax detection puts telemarketers to my fax machine |
| 19:19.10 | c4t3l | me too |
| 19:19.11 | sevard | haha |
| 19:19.18 | Idle | sevard: canada. :( |
| 19:19.21 | sevard | ah |
| 19:19.25 | sevard | sad |
| 19:19.28 | Idle | yea |
| 19:19.41 | puppet | haha idle |
| 19:19.43 | sevard | there are services out there that will take your emails and fax them and take your faxs and email them |
| 19:20.03 | c4t3l | oh lord |
| 19:20.04 | Idle | my wildcard has some shitty hangup detection... and when using sip -> zap, I get major echo for the first 10 seconds |
| 19:20.05 | sevard | hell with * you could start one |
| 19:20.16 | Idle | yep |
| 19:20.30 | Idle | we had it at work |
| 19:20.35 | Idle | linked with our FirstClass server |
| 19:20.42 | sevard | yarg at FC |
| 19:20.54 | Idle | FC++ |
| 19:21.02 | sevard | i'd hate to maintain an fc server |
| 19:21.13 | Idle | its actually stupid easy |
| 19:21.13 | c4t3l | i use gentoo |
| 19:21.26 | Idle | I love my FC |
| 19:21.30 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 19:21.31 | sevard | Idle: really now, it looks quite complicated and bloated from the outside |
| 19:21.45 | Idle | its complicated, but damned easy to admin :D |
| 19:21.50 | sevard | haha |
| 19:21.55 | Idle | we cant break the thing... we've tried |
| 19:21.56 | sevard | it's expensive ,no doubt |
| 19:21.56 | AlexCTI | hi, there is someone familiar with queues? |
| 19:22.01 | Idle | oooh yea |
| 19:22.03 | c4t3l | FC has very good init scripting as well |
| 19:22.09 | sevard | AlexCTI: fireaway |
| 19:22.09 | Idle | but we run 1300+ users, unclustered |
| 19:22.29 | sevard | Idle: I know the school districts here run them and they have well over 14,000 employees |
| 19:22.32 | Idle | c4t3l: yea, the batch admin scripting is pretty decent |
| 19:22.54 | Idle | sevard: yep, we're on the advisory committee... half the places that use it are school |
| 19:22.55 | c4t3l | better than Gentoo's runscript stuff |
| 19:23.07 | Idle | c4t3l: I think we're talking about different fc's |
| 19:23.11 | sevard | ahaha |
| 19:23.17 | sevard | he's talking about fedoracore |
| 19:23.18 | Manipura | Anyone know anything about dynamic load balancing asterisk? |
| 19:23.21 | sevard | which, imho, blows |
| 19:23.32 | sevard | c4t3l: we're talking about FirstClass |
| 19:23.34 | justinu|laptop | FC sucks... half the shit is broken on initial install |
| 19:23.40 | sevard | no shit |
| 19:23.44 | sevard | slackware 4 leif. |
| 19:23.46 | Idle | FC is amazing, you can run sooo many users before clustering... and there is so much clustered services in it... I love it |
| 19:23.52 | c4t3l | oops |
| 19:23.53 | Idle | yea, Fedora Core licks some nuts |
| 19:23.57 | sevard | hahaha |
| 19:24.10 | sevard | for the same reasons I don't use Fedora Core I don't use CentOS |
| 19:24.12 | c4t3l | Manipura, check out SER |
| 19:24.19 | Idle | yep |
| 19:24.51 | sevard | I'm goin to try out these fancy new Ubuntu and Gentoo one of these days, I haven't used anything but slackware on a daily basis for a long time |
| 19:25.09 | c4t3l | i like gentoo only for the portage app |
| 19:25.18 | c4t3l | emerge kicks ass |
| 19:25.20 | justinu|laptop | ubuntu works nicely out of the box |
| 19:25.21 | Manipura | c4t3l, SER with asterisk? Or SER without Asterisk |
| 19:25.34 | c4t3l | with * :D |
| 19:25.37 | sevard | I heard portage is neat, but tooks like those scare me, I don't like anything that's installing and compiling shit on my box automatically |
| 19:26.01 | Idle | I like breaking package managers and building everything from source |
| 19:26.03 | sevard | I'd rather know every library every module that I'm putting into my system, i'm an administrator, I don't trust or need other administrators to administrate my boxen. |
| 19:26.20 | Idle | sevard: hear hear! |
| 19:26.35 | Idle | lol :D |
| 19:26.43 | puppet | http://www.aish.com/movies/PhotoFraud.asp |
| 19:26.44 | sevard | <PROTECTED> |
| 19:27.20 | *** join/#asterisk LoneShadow (n=duh@59.92.139.23) |
| 19:28.00 | c4t3l | ever use *BSD?? |
| 19:28.11 | puppet | isnt it possile to connect a regular modem and that way get ONE zap channel? |
| 19:28.47 | *** join/#asterisk e-MAC (n=YMironek@251-20-207-82.pool.ukrtel.net) |
| 19:29.27 | *** join/#asterisk Rawplayer (n=kevin@braadharing.oom-killer.org) |
| 19:29.30 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 19:29.41 | sevard | yes, i've used BSD, my personal opinion is bleh, i don't like it that much -- i do enjoy using linux -- on the flipside os x is pretty -neat- |
| 19:29.48 | Rawplayer | hi, i have a question about voip/astrisk but its not about configuration |
| 19:29.56 | Rawplayer | its about traffic |
| 19:30.13 | Rawplayer | when you have 1,2mbit upload how many clients can make usage of that? |
| 19:30.21 | SplasPood | Rawplayer: depends on the codec used |
| 19:30.28 | Rawplayer | or is it just depended on the latency |
| 19:30.31 | Rawplayer | hmm |
| 19:30.35 | SplasPood | Rawplayer: if you search for codec bandwidth on voip-info.org there's a nice chart |
| 19:30.45 | Rawplayer | okay thanks |
| 19:31.31 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 19:31.34 | Rawplayer | http://www.voip-info.org/wiki-Bandwidth+consumption |
| 19:32.43 | nortex | I'm trying to implement a paging setup where * accepts a call to the chan_oss and then plays a tone to both the caller and soundcard before the caller says anything. What I have done is use the A option in Dial to play an announcement tone, but it is only for the soundcard in this case. How can I do this? |
| 19:32.58 | sevard | puppet: those are horribly doctored photographs |
| 19:33.07 | sevard | one could almost do better in paint |
| 19:33.22 | puppet | lol sevard |
| 19:33.46 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 19:33.54 | c4t3l | ALMOST ;) |
| 19:34.05 | Un1x | sevard: youre more of a windows guy you love guis :p |
| 19:34.38 | smackus | if i am using the addqueuemember command. is there anywhere that shows logged in status of the device? similar to show agents? |
| 19:34.44 | sevard | Un1x: ....what..? |
| 19:34.56 | smackus | found it... show queues. |
| 19:34.56 | nortex | puppet, Mind reposting the link? |
| 19:35.06 | puppet | http://www.aish.com/movies/PhotoFraud.asp |
| 19:35.36 | smackus | now.. can i make addqueuemember add the device to multiple queues? |
| 19:35.39 | sevard | Un1x: At the moment i'm in console, talking to you on a TTY, in epic5 connected to shell, I IM from centericq and do most of my browsing in lynx |
| 19:35.56 | watchy | hmm |
| 19:35.57 | sevard | how am I a "GUI guy"? |
| 19:36.15 | watchy | i have phones over a 3mbit wireless link should i change the codec from ulaw to something else? |
| 19:36.18 | smackus | sevard: show off... |
| 19:36.22 | watchy | they say they are gettting clipping |
| 19:36.22 | sevard | heh |
| 19:36.24 | eKo1 | watchy: nay |
| 19:36.34 | smackus | i am using a screen keyboard and clicking each letter with my mouse :-D |
| 19:36.43 | watchy | eko1: hrm, i gotta fix the clipping some how |
| 19:36.53 | eKo1 | eKo1: change the codec and see if it helps |
| 19:37.09 | watchy | talking to me? |
| 19:37.12 | sevard | i'm using a handspring visor writing asm with my stylus with my teeth talking to you over a telnet connection on irc |
| 19:37.17 | eKo1 | watchy: oops, yeah |
| 19:37.20 | *** part/#asterisk Ixitxachitl (n=m@209.151.130.10) |
| 19:37.26 | watchy | eko1: what should i change it to? |
| 19:37.27 | c4t3l | oh yeah... well i'm connected through my Pcom and typing this out to you through the System() command in asterisk!!! |
| 19:37.34 | eKo1 | watchy: gsm |
| 19:37.50 | sevard | eKo1: 'sudeen' confuses me. |
| 19:37.50 | watchy | so in sip.cfg put codecallowed=gsm or whatever? |
| 19:37.58 | watchy | i cant remember the exact setting name |
| 19:38.05 | c4t3l | allow=gsm |
| 19:38.13 | watchy | yea |
| 19:38.23 | eKo1 | sip.cfg? |
| 19:38.29 | watchy | i mean sip.conf |
| 19:38.50 | eKo1 | Looks like everyone is confused today :P |
| 19:39.03 | watchy | allow=ulaw |
| 19:39.07 | watchy | thats what its set to now |
| 19:39.20 | watchy | if i set it to gsm can i just reload without issues? |
| 19:39.26 | eKo1 | Yes. |
| 19:39.45 | *** join/#asterisk easel (n=erik@interlink-gw1.ilsw.com) |
| 19:39.49 | eKo1 | Make sure the phones/ATAs support it though. |
| 19:39.55 | watchy | whats the command to see if any calls are in place |
| 19:40.07 | watchy | eko: the phones are polycom 501s601s im sure they support it |
| 19:40.31 | justinu|laptop | they do not support gsm |
| 19:40.39 | watchy | oh |
| 19:40.43 | watchy | that sucks |
| 19:40.51 | justinu|laptop | g711 and g729 only, iirc |
| 19:41.00 | eKo1 | use g729 then |
| 19:41.16 | watchy | dont you gotta buy g729 |
| 19:41.32 | eKo1 | You don't have a license? |
| 19:41.32 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 19:41.59 | *** join/#asterisk pingwin (n=pingwin@216.249.143.62) |
| 19:42.00 | sevard | licenses are 10 dollars from digium.com |
| 19:42.08 | easel | anyone here have some sort of 'send call to voicemail' button working with cisco 7960's with sip firmware? |
| 19:42.12 | sevard | Pay with a CC and get it in about 4 minutes |
| 19:42.38 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 19:43.00 | watchy | eko1: hrm i need like 30 of them |
| 19:43.00 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 19:43.12 | rene1 | how do i pull a file from asterisk svn |
| 19:43.15 | rene1 | a specific file |
| 19:43.31 | watchy | are g729 licenses worth it? |
| 19:43.51 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 19:44.01 | sevard | watchy: no idea, I'm about to test g729 on my network next week |
| 19:44.35 | watchy | sevard: why do you need it instead of say ulaw? |
| 19:44.45 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.148.139.telnor.net) |
| 19:44.57 | watchy | i know i think i need something that will conserver bw though since im using like 6 lines across a wireless link |
| 19:45.18 | sevard | watchy: ulaw == 64Kbps (plus overhead) @ toll quality, g729 is supposedly toll quality but takes 8Kbps plus overhead |
| 19:45.23 | Shaun2222 | doesnt gsm have the ability to use low bw? |
| 19:45.25 | justinu|laptop | g729 sounds pretty good |
| 19:45.49 | justinu|laptop | for voice... however it makes call progress tones and music sound like shit |
| 19:45.52 | angom_w | hello, someone that can recommend a CDMA PCI card o gateway that works with asterisk ? |
| 19:45.54 | watchy | sevard |
| 19:45.58 | watchy | sevard: wow |
| 19:46.00 | *** join/#asterisk sp0n9e_ (n=sp0n9e@phpurge.com) |
| 19:46.00 | sevard | justinu|laptop: it's highly compressed, no? |
| 19:46.10 | sevard | takes lots more CPU |
| 19:46.12 | justinu|laptop | yeah, 10kbps streams |
| 19:46.33 | justinu|laptop | modern cpus are pretty powerful, so you can do a fair bit of transcoding |
| 19:46.54 | watchy | sevard: so your saying ulaw is 64kbps as in isdn would be 128kbps |
| 19:47.31 | justinu|laptop | no, he's saying it ends up beig about 80kbps |
| 19:47.34 | sevard | iirc isdn is 64Kbps, but I don't recall that nonsense |
| 19:47.34 | DrkShdw | justinu|laptop: at the expense of a single dropped packet causing a lot more of the voice data to be lost |
| 19:47.38 | justinu|laptop | because of IP/UDP/RTP overhead |
| 19:47.42 | Shaun2222 | does the cisco7960 and polycom phones support g729? |
| 19:47.46 | [TK]D-Fender | GSm is nearly interchangeable with G729 for normal use. I'd rather use GSM where possible for license reasons. |
| 19:47.48 | watchy | jesus christ |
| 19:48.12 | watchy | ulaw eats a shitlaod of bw |
| 19:48.23 | sevard | I'm an audiophile and I can tell a noticeable difference between ulaw and GSM, gsm sounds like -shit- |
| 19:48.39 | justinu|laptop | makes you realize how poor cell phone calls really are |
| 19:48.41 | pfn | gsm is effectively voice-only |
| 19:48.54 | hmmhesays | [TK]D-Fender: i'm thinking about getting the gf a keyboard, she had to give up piano a few years back for lack of piano |
| 19:48.56 | pfn | gsm will compress any non-voice data into worthlessness |
| 19:49.03 | sp0n9e | i'm having issues with wanpipe on x86_64 |
| 19:49.06 | pfn | audiofile or not, gsm is worthwhile to compress to if you're low on bandwidth |
| 19:49.08 | Shaun2222 | how great of quaility are we trying to get here.,.. it's a phone.. |
| 19:49.10 | [TK]D-Fender | hmmhesays: Get her one like I just got :) |
| 19:49.12 | justinu|laptop | i have a yamaha s80 |
| 19:49.16 | justinu|laptop | not a bad setup |
| 19:49.24 | hmmhesays | i need some suggestions for under a grand |
| 19:49.25 | [TK]D-Fender | s80? High end... sweet |
| 19:49.28 | watchy | how many kbps is say g711a? |
| 19:49.36 | justinu|laptop | same |
| 19:49.37 | hmmhesays | about 70 |
| 19:49.41 | sevard | Shaun2222: once you use ulaw for a year then switch to Speex, then you can talk |
| 19:49.44 | *** join/#asterisk rephorm (n=rephorm@cpe-24-27-8-18.austin.res.rr.com) |
| 19:50.01 | [TK]D-Fender | justinu|laptop: : Mine : http://www.m-audio.ca/products/en_ca/KeystationPro88-main.html |
| 19:50.01 | pfn | ulaw is about 80kbps |
| 19:50.04 | sp0n9e | s/sanoma/sangoma/ |
| 19:50.26 | justinu|laptop | fender: how's the action on that? |
| 19:50.47 | [TK]D-Fender | hmmhesays: If she has a computerthis is a great unit. Cost me $375 |
| 19:50.53 | Shaun2222 | so does the cisco7960 and polycom phones support g729 then? |
| 19:50.58 | watchy | so could i say get 6 licences on g729 and leave the rest on ulaw? |
| 19:51.00 | [TK]D-Fender | justinu|laptop: Love it. More than my Roland HP-137 even. |
| 19:51.16 | watchy | it looks like the polys do |
| 19:51.17 | justinu|laptop | the hammer action on the s80 is pretty good, but the piano sounds kinda suck |
| 19:51.31 | [TK]D-Fender | justinu|laptop: Although I really do understand the hammer difference from one piano to another of any kind. |
| 19:51.45 | watchy | jesus i may just buy 6 copies of g729 to test across my wireless network |
| 19:51.51 | watchy | and see if that fixes clipping |
| 19:51.56 | Shaun2222 | i cant seam to get the poly's to work on a nat connecting to a remote asterisk server.. |
| 19:52.00 | watchy | would you guys recommend that? |
| 19:52.00 | [TK]D-Fender | watchy: Use G7.29 only from * to the outside and leave all internal on ULAW |
| 19:52.12 | Shaun2222 | the phone connects and it rings and stuff |
| 19:52.17 | justinu|laptop | watchy: you need to know if your problem is caused by packet loss |
| 19:52.17 | Shaun2222 | but i cant talk through it |
| 19:52.27 | justinu|laptop | watchy: if you have packet loss, g729 will suck too |
| 19:52.33 | watchy | tk: i have 6 phones across a 3mbit wireless network |
| 19:52.41 | justinu|laptop | wireless is tough |
| 19:52.47 | watchy | they are complaining about clipping |
| 19:52.55 | justinu|laptop | any interference from a microwave, or something, and you can kiss your voice calls goodbye |
| 19:53.00 | eKo1 | What is the sample size for g729 in *? |
| 19:53.26 | sevard | Run your wireless voice data on a 5.7 wireless network |
| 19:53.28 | watchy | this is high end 5.8ghz wireless equipment just this aint 802.11b/g |
| 19:53.30 | [TK]D-Fender | watchy: Packet loss is packet loss... I seriously doubt you hve BW concerns at all. I'd lay bets its JITTER. |
| 19:53.35 | watchy | sevard: its 5.8ghz |
| 19:53.44 | sevard | watchy: there you go, which product? |
| 19:53.50 | watchy | Alvarion |
| 19:54.06 | watchy | but were getting jitter/clipping from what the client says |
| 19:54.08 | sevard | with that high of a frequency you need *direct* line of sight |
| 19:54.14 | watchy | yea it is |
| 19:54.21 | sevard | how does your SnR ratio look? |
| 19:54.36 | justinu|laptop | clean up the thruput/jitter, and g711 will be fine |
| 19:55.09 | hmmhesays | can you change your global variables with cmd SET? |
| 19:55.33 | watchy | not sure. lemme call my wireless guys and get the ips for the radios and ill let you know |
| 19:55.34 | sevard | SnR, DSCP (if your radios will do it) and make sure your rtp packet sizing is at 0.020 (20 ms) |
| 19:55.45 | sevard | give me the IP aswell |
| 19:55.45 | sevard | :) |
| 19:55.48 | justinu|laptop | heh |
| 19:55.52 | watchy | where do i check the rtp packet size? in asterisk? |
| 19:55.58 | justinu|laptop | on the phones |
| 19:56.01 | justinu|laptop | polycom defaults to 20ms |
| 19:56.05 | sevard | in your sip client |
| 19:56.11 | justinu|laptop | sipura defaults to 30ms |
| 19:56.15 | sevard | lots of the sipura atas love to set themselves to 30 ms, which is a no no |
| 19:56.22 | watchy | sevard: its polycom phones |
| 19:56.28 | sevard | especially on a netwoek like yours |
| 19:56.33 | sevard | network* |
| 19:56.37 | justinu|laptop | bitch to your wireless ppl about your QoS |
| 19:56.39 | sevard | are you doing DSCP? TOS? QOS? |
| 19:56.43 | justinu|laptop | it's obviously not good enough for voIP |
| 19:56.51 | justinu|laptop | DSCP can really help |
| 19:56.57 | justinu|laptop | if the link is saturated |
| 19:56.58 | AlexCTI | Sevard: I didn't found the way to put a msg into a queue that say a soon the caller arrive the systm tell him the agents are busy, and either a msg that say please press 0 for leave a msg. |
| 19:57.14 | watchy | sevard: none i dont think. |
| 19:57.14 | sevard | no, a 3mbit 5.8 wireless backhaul will do voip fine, but if your wireless people fucked up dscp they deserve to be slapped |
| 19:57.23 | sevard | watchy: dscp man :) go go |
| 19:57.29 | watchy | they have 2 linksys vpn routers that connect the 2 places |
| 19:57.44 | watchy | and the phones run over them. they are RV082 model linksys routers |
| 19:57.51 | justinu|laptop | wow, spare no expense |
| 19:58.07 | watchy | ok check this sevard |
| 19:58.10 | sevard | there's another variable, ssl encapsulation takes extra bandwidth |
| 19:58.16 | watchy | should i replace them with cisco 501s? |
| 19:58.26 | sevard | I don't know much about the 501s |
| 19:58.27 | [TK]D-Fender | AlexCTI: You can't have it play immediately, only on interval. |
| 19:58.34 | justinu|laptop | well, cisco IOS has endless QoS options |
| 19:58.47 | watchy | so i really need QoS probably? |
| 19:59.01 | justinu|laptop | i would say yes |
| 19:59.09 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 19:59.11 | [TK]D-Fender | justinu|laptop: So now he make sure his VoIP packets are the first ones lost? ;) |
| 19:59.12 | Wazb^ | hi to all |
| 19:59.17 | watchy | i wonder if my radios have QoS built in |
| 19:59.20 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 19:59.40 | justinu|laptop | well, the idea is that if the radio channel is saturated, you want to Queue non-voip packets for transmittal |
| 19:59.44 | AlexCTI | TKD-Fender, OK, and how can i make to play a press 0 to Voicemail, i used announce but it didn't take it. |
| 19:59.47 | sevard | AlexCTI: for your 'press 0 to leave a message' use context=queue-out |
| 19:59.47 | justinu|laptop | and forward VoIP packets without delay |
| 19:59.52 | watchy | yea |
| 20:00.04 | watchy | so should i do it in the wireless radios or in the routers? |
| 20:00.08 | [TK]D-Fender | AlexCTI: PB your queue definition. |
| 20:00.22 | clyrrad | Does anyone have any documentation on the parameter for QueueStatus? |
| 20:00.22 | Wazb^ | is it possible to use Macro which is in Extensions.conf file in AGI ? |
| 20:00.25 | sevard | !pb |
| 20:00.27 | sevard | ~pb |
| 20:00.28 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 20:00.52 | justinu|laptop | it depends on whether your wireless radios understand layer 3 or not |
| 20:00.59 | AlexCTI | sevard: I did and if i press 0 it works, the only thing is that i cann't make that the msg be played. during the queue hold |
| 20:01.13 | justinu|laptop | i dunno much about wireless, other than my linksys wrt54g running dd-wrt |
| 20:01.29 | rpm | whats the best way of doing authentication via sip? should i be doing md5 such as auth = username#md5secret@context ? or sticking with the username=user and secret=plaintext secret? |
| 20:01.45 | clyrrad | ... TKD - Do you know about this function QueueStatus? |
| 20:01.45 | sevard | AlexCTI: announce-holdtime = yes |
| 20:01.47 | rpm | do the phones have to support the md5 hashing or does asterisk do the hash? |
| 20:02.06 | sevard | also, announce-frequency = 30 |
| 20:02.11 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:02.24 | *** join/#asterisk taker (n=relas@u5-78.dsl.vianetworks.de) |
| 20:02.29 | [TK]D-Fender | clyrrad: the AMI one? Nope |
| 20:02.34 | AlexCTI | severd: that command is enable, but it says the hold time |
| 20:02.40 | Wazb^ | is it possible to use Macro which is in Extensions.conf file in AGI ? |
| 20:02.47 | taker | Hello! I'm using bristuff-0.2.0-RC8s. Which chan_capi should I install? |
| 20:02.47 | [TK]D-Fender | sevard: Double-wrong. |
| 20:02.57 | clyrrad | yea the AMI one - it has ZERO documentation |
| 20:03.01 | clyrrad | cant find it anywhere |
| 20:03.04 | [TK]D-Fender | AlexCTI: "periodic-announce" |
| 20:03.06 | AlexCTI | and the announce-frecuency too |
| 20:03.11 | sevard | [TK]D-Fender: that works fine fo rme |
| 20:03.46 | [TK]D-Fender | AlexCTI: read the big print in http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 20:04.04 | [TK]D-Fender | sevard: that does position announcements, but not where you should be putting general stuff. |
| 20:04.06 | sevard | [TK]D-Fender: unless that is what he was asking for those two options i posted work great for me, there's eriodic-announce and periodic-announce-frequency i guess |
| 20:04.13 | [TK]D-Fender | AlexCTI: "New feature (Jul 31, 2005 CVS HEAD)" |
| 20:04.30 | AlexCTI | TK D Fender: periodic-announce = <filename> without extension? path or something else? |
| 20:04.30 | [TK]D-Fender | sevard: Thats sort of expressly what its for. |
| 20:04.47 | [TK]D-Fender | AlexCTI: It works the same way as every other sound file in * |
| 20:05.04 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 20:05.07 | rene1 | QueueStatus takes no parameter as far i can tell |
| 20:05.08 | AlexCTI | TK D Fender, Thnks, ill read the big print.. |
| 20:05.14 | sevard | AlexCTI: sound files are pulled from /var/lib/asterisk/sounds, you define it without extension in the config file. |
| 20:05.18 | *** join/#asterisk Assid (i=assid@203.115.83.213) |
| 20:05.25 | rene1 | AMI QueueStatus that is |
| 20:06.07 | rene1 | it will just tell you about members, callers and general queue stats |
| 20:06.18 | AlexCTI | I'll take care about that too, sevard, thanks |
| 20:06.45 | watchy | i need to find out if the radio supports QOS i guess |
| 20:06.50 | watchy | or ToS or someething |
| 20:08.07 | watchy | hmm in the radio conf it says something about ToS |
| 20:09.36 | sb_mx | rene1, are you using AGIs to communicate with AMI? |
| 20:09.58 | rene1 | sb_mx: no i am using the astmanproxy |
| 20:10.23 | rene1 | astmanproxy can be queried asterisk way or via http |
| 20:10.27 | rene1 | it is quite cool |
| 20:10.40 | rene1 | you can use even JavaScript to query it (ajax) |
| 20:10.47 | smackus | does anyone know if it is possible to auto log off an agent who is not in the agents.conf, one that is added using AddQueueMemeber(<queuename>)? |
| 20:11.08 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 20:11.26 | wunderkin | smackus, remove queue member |
| 20:11.29 | sb_mx | rene1, yup. i've used it before. im trying to use an agi atm but all i'm getting is a msg saying "Queue status will follow" :S |
| 20:11.31 | xnon | anybody here agree your accound fwd in your asterisk server? |
| 20:11.37 | rene1 | ahhh |
| 20:11.39 | rene1 | well |
| 20:11.44 | rene1 | weird |
| 20:12.00 | rene1 | it sometimes happens to me, but i reissue the command and it shows |
| 20:12.20 | rene1 | the action QueueStatus triggers QueueParams and QueueMember events |
| 20:12.26 | sb_mx | rene1, does the manager need any specific read/write permits? |
| 20:12.33 | rene1 | you need to capture those, maybe you are filtering them out |
| 20:12.37 | smackus | sorry... I did not provide the full question. What I was looking for is if an agent does not answer a call delivered to them by the queue, can they be auto logged off. I am looking for the same function as provided by the agents.conf "autologoff="15" |
| 20:12.53 | sb_mx | rene1, ahhh of course. that must be it. stupid me. thanks man. im gonna kick this agi's butt |
| 20:13.04 | rene1 | sure |
| 20:13.10 | [TK]D-Fender | sb_mx: it doesn't respond immediately to the packet, it sends out info packets later so you have to wait to poll for them. |
| 20:13.22 | wunderkin | smackus, use a local channel, and after the dial check the dialstatus for noanswer |
| 20:13.50 | xnon | Aug 10 15:08:08 NOTICE[20614]: chan_iax2.c:7500 socket_read: Registration of '791710' rejected: 'Registration Refused' from: '192.246.69.186' |
| 20:13.58 | sb_mx | [TK]D-Fender, k, ty |
| 20:14.14 | smackus | wunderkin: ok... sounds over my head. can you elaborate and maybe give some guidance? |
| 20:17.20 | wunderkin | smackus, check the wiki for local channels and dialstatus |
| 20:17.29 | smackus | awesome, thanks |
| 20:17.34 | [TK]D-Fender | smackus: Option : make the agent ring time in your Local channel less than the queue timeout and use the dialplan to kick them. |
| 20:17.47 | *** join/#asterisk lin00bies (n=lin00bie@210.213.198.60) |
| 20:18.11 | smackus | [TK]D-Fender: hmmm. interesting. |
| 20:18.33 | *** join/#asterisk oadaeh (n=jason@216.241.54.132) |
| 20:18.59 | smackus | so something to the effect of if it rings more than 15 seconds then next priority would be removequeumember? something like that? |
| 20:19.34 | [TK]D-Fender | smackus: You learn quickly my young Jedi ;) |
| 20:20.10 | smackus | better than being the guy everyone flames cuz there a dumbass :-D |
| 20:21.19 | [TK]D-Fender | smackus: Extremely few have caught anything worth calling "flames" from me. Only the persistantly stupid. |
| 20:24.20 | [TK]D-Fender | /mewaves back |
| 20:24.34 | justinu|laptop | some of us even tip fender for helping the noobs |
| 20:24.49 | [TK]D-Fender | justinu|laptop: ;) Even I have my limits |
| 20:25.18 | sevard | the only flames that come from [TK]D-Fender is out his ass |
| 20:25.21 | sevard | bada tish. |
| 20:25.59 | rene1 | ahaha that was retarded but funny |
| 20:26.14 | [TK]D-Fender | For idiots of (g)astronimical proportions ;) |
| 20:26.30 | [TK]D-Fender | sevard: oneup++ :D |
| 20:26.43 | sevard | you always have to oneup somebody's bad joke with one of your own |
| 20:26.46 | sevard | ah, you beat me to it |
| 20:27.40 | [TK]D-Fender | heh |
| 20:27.59 | Lyfe | interesting, so you can do a queue dynamically like that, to add agents and delete agents, and the way to delete is to have the fallthrough go to 'removequeuemember' and then return to the queue? |
| 20:28.22 | Lyfe | (with a goto, or whatnot) |
| 20:29.35 | Manipura | whats the point of a blade server? Is it just smaller or something? |
| 20:29.48 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:30.02 | Manipura | My friend says blades are the best.. but I'm trying to find out why.... |
| 20:30.32 | eKo1 | Yes, blade servers are much more compact. |
| 20:30.41 | eKo1 | I wish I had one. |
| 20:30.59 | eKo1 | Instead, I have three racks with towers and 4U servers in there. |
| 20:31.08 | eKo1 | there being my server room. |
| 20:32.37 | Lyfe | blade: high investment price, good space requirements. |
| 20:33.03 | Lyfe | myself, i'm more interested in 1U & 2U servers, and virtualization. |
| 20:33.16 | [TK]D-Fender | Lyfe: no need for goto. |
| 20:33.44 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:33.46 | *** part/#asterisk [TK]D-Fender (n=Administ@toronto-HSE-ppp4122655.sympatico.ca) |
| 20:34.00 | Lyfe | i think i'm confused on how the queues work then.. need more info :\ |
| 20:34.17 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 20:34.18 | *** join/#asterisk cybertrickle (n=cybertri@wsip-70-167-111-3.ph.ph.cox.net) |
| 20:34.32 | rene1 | well a blade server has mainly two purposes, comitting suicide with it when it crashes or threaten to cut those porn downloaders heads or worse with it, |
| 20:34.53 | rene1 | really man, once you tell them you have a blade server they will look up to you, i mean |
| 20:35.00 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 20:35.06 | cybertrickle | I have a low load of about 2, 70% idle, 90% free memory. But I have static on all of the lines. Anything I can try ? |
| 20:35.28 | *** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 20:35.43 | eKo1 | cybertrickle: Are you using an FXO/FXS card? |
| 20:35.51 | pingwin | aren't blades the half cabinet that you can stack 1U's into? |
| 20:35.58 | cybertrickle | eKo1, Yes |
| 20:36.16 | Lyfe | i have click's on my t1 (e&m wink, not a pri) |
| 20:36.38 | cybertrickle | eKo1, These servers have 3 T1 zap cards. They just handle that zap channel and a connection to a sip server. |
| 20:36.55 | cybertrickle | eKo1, The problem happens when they use more then 2 of the zap cards. |
| 20:37.00 | x86 | cybertrickle: why not just a single quad port? |
| 20:37.30 | cybertrickle | x86, opps. Sorry its a single card, quad port card. |
| 20:37.36 | Lyfe | pingwin: typically blades are smaller than 1U servers, and they're not exactly a generic cabinet. they tend to have very very specific form factors, and often they're mounted vertically, instead of horizontally like a typical 1U server (or whatnot) |
| 20:37.46 | *** join/#asterisk adorah (n=Administ@87.68.173.125.cable.012.net.il) |
| 20:37.47 | eKo1 | cybertrickle: maybe it is grounding problem |
| 20:38.02 | x86 | cybertrickle: did you look at the LBO for each circuit? |
| 20:38.05 | pingwin | Lyfe: yeah that's what I thought. are they clustered? |
| 20:38.09 | Lyfe | and, once you go with a particular blade server vendor, you have to stick with their parts, ('cept generics, like harddrives). |
| 20:38.13 | pingwin | otherwise,whats so good about them? |
| 20:38.14 | x86 | cybertrickle: and have you tried frogging the circuits? |
| 20:38.19 | Lyfe | pingwin: depends how you set them up, far as i know. |
| 20:38.44 | Lyfe | dunno, that's part of the unknown world to me, is in-depth blade server knowledge. |
| 20:38.49 | cybertrickle | x86, How do I check LBO? |
| 20:39.02 | cybertrickle | x86, How do I frog curcuits ? |
| 20:39.06 | Lyfe | one of those "eh, for now, 1U & 2U servers make more sense, since we buy one every 4-5 months, and that's it." |
| 20:39.23 | cybertrickle | eKo1, How would I be a grounding problem ?> |
| 20:40.08 | x86 | cybertrickle: not sure with a zaptel card... |
| 20:40.15 | x86 | cybertrickle: (how to check LBO) |
| 20:40.33 | Lyfe | pingwin: i believe the big thing about blades is that performance per U is higher. |
| 20:40.43 | x86 | cybertrickle: but you can easily frog circuits just by swapping which port they are connected to |
| 20:41.03 | pingwin | ahhh |
| 20:41.33 | Lyfe | cause, you can jam 10 or so servers into a <10U space. |
| 20:41.37 | cybertrickle | Why would you do that again ?? |
| 20:41.41 | pingwin | i do have a quick asterisk question tho, and I'm new to asterisk, PBX's, VOIP, Pri and well everything asterisk hehe |
| 20:41.57 | x86 | cybertrickle: to see if it was the 3rd circuit that has issues? |
| 20:42.03 | De_mon | I setup the extension 'exten => _51NXXNXXXXXX' how can I forward another extension to it? |
| 20:42.25 | x86 | cybertrickle: if you take the 3rd circuit and put it in the 1st position and still have issues with it, you know it's the circuit |
| 20:42.39 | pingwin | but if I don't have the Pri card yet, can I still configure asterisk to use the phones over the PoE switch? |
| 20:42.43 | Lyfe | De_mon: I think you might be interested in the "goto" command. (there might be another more appropriate way) |
| 20:42.46 | x86 | cybertrickle: if you still have issues with the 3rd port after the swap, you know it's a problem with the card or some settings |
| 20:43.18 | Lyfe | pingwin: i don't see any reason you couldn't. |
| 20:43.54 | cybertrickle | All ports have static issues, when you move a t1 off of that machine. the issues completely go away. |
| 20:44.08 | pingwin | Lyfe: cool, thanks. |
| 20:44.19 | Lyfe | pingwin: many walkthroughs on setting up asterisk walk you through configuring phones before configuring any way to get out of the system (eg, through pots, or fwd, or whatnot) |
| 20:44.21 | x86 | cybertrickle: RMA your card then |
| 20:44.42 | pingwin | compiling it all now, the pri card should be in next week, but we're anxious to know if this equipment will work at all |
| 20:44.59 | cybertrickle | RMA ? |
| 20:45.01 | Lyfe | pingwin: ahh, i see. |
| 20:45.33 | x86 | cybertrickle: return to sender ;) |
| 20:45.43 | x86 | cybertrickle: Return Merchandise Authorization |
| 20:45.57 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 20:45.59 | a1fa | hey |
| 20:46.10 | a1fa | i am trying to add an extension event s-* for "Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 20:46.11 | Lyfe | sounds like he needs to RMA the system instead of the cards |
| 20:46.21 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 20:46.48 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 20:47.11 | *** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw) |
| 20:47.25 | x86 | Lyfe: why would you suggest his system was to blame, as opposed to the card? |
| 20:47.35 | Lyfe | cause he said all the cards give static, i believe. |
| 20:47.43 | x86 | no |
| 20:47.53 | x86 | he said all the CIRCUITS give static when plugged into that card |
| 20:48.10 | x86 | but he plugs them up to another device and they work fine |
| 20:48.29 | Lyfe | ahh.. got confused by this one: cybertrickle> All ports have static issues, when you move a t1 off of that machine. the issues completely go away. |
| 20:48.52 | justinu|laptop | maybe he has interrupt sharing problems |
| 20:49.01 | x86 | justinu|laptop: single card, single interrupt? |
| 20:49.19 | De_mon | Lyfe: -- Executing Goto("SIP/jon-3958", "518638774177") in new stack -- Goto (local,18638774177,2147483647) |
| 20:49.25 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 20:49.42 | De_mon | it didn't goto local,51863... and I duno where that 214etc came from |
| 20:50.21 | *** join/#asterisk aydiosmio (n=aydiosmi@65.213.70.43) |
| 20:50.31 | justinu|laptop | x86: sorry, i haven't exactly been paying attention |
| 20:50.35 | watchy | sevard: you there |
| 20:50.41 | Lyfe | hmm.. goto takes 3 parameters? |
| 20:50.45 | De_mon | I'm trying to move a call from _1NXXNXXXXXX to _51NXXNXXXXXX |
| 20:50.46 | aydiosmio | can asterisk speak currency? or am I gonna have to bust out festival? |
| 20:51.23 | Juggie | do you think festival is really required to say 'dollars' & 'cents'? |
| 20:51.24 | De_mon | with: exten => _1NXXNXXXXXX,1,Goto(5${EXTEN}) |
| 20:51.38 | aydiosmio | Juggie: yes |
| 20:51.48 | De_mon | Juggie he'd have to write a complex dialplan otherwise! |
| 20:51.49 | Lyfe | it'd be exten => _1NXXNXXXXXX,1,Goto(5${EXTEN},1) (i think) |
| 20:52.06 | aydiosmio | or else you get five one six seven dollars and six one cents |
| 20:52.18 | De_mon | Lyfe heh.. You may have a point |
| 20:52.19 | Juggie | aydiosmio, * does not support currency. |
| 20:52.40 | Lyfe | goto threw me off on that priority part too. |
| 20:52.43 | a1fa | . |
| 20:52.44 | aydiosmio | I just want to have the number spoken instead of spelled out. |
| 20:52.46 | a1fa | i am trying to add an extension event s-* for "Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 20:52.56 | Juggie | aydiosmio, asterisk does the number properly, use SayNumber instead of SayDigits |
| 20:53.02 | aydiosmio | oh |
| 20:53.04 | aydiosmio | duh |
| 20:53.06 | aydiosmio | thank you |
| 20:53.23 | a1fa | where can i find extensions events for s-BUSY,s-...? |
| 20:53.23 | Juggie | then break your number into the part before the decimal and the part after |
| 20:53.27 | De_mon | perfect. woot |
| 20:53.54 | Lyfe | De_mon: glad it works. |
| 20:53.55 | Juggie | so SayNumber(100) Background(Dollars) Saynumber(45) Background(cents) would say, 100Dollars 45cents |
| 20:53.57 | *** join/#asterisk jarrod (n=jarrod@juniperyour.net) |
| 20:53.59 | Lyfe | ./ |
| 20:54.04 | Lyfe | woops. |
| 20:54.15 | jarrod | any particular reason why asterisk seems to ignore my featuremap entries from my polycom? |
| 20:54.17 | Juggie | you would obviously have to record 'dollars' & 'cents' or find the voice files somewhere else. |
| 20:54.39 | De_mon | a1fa huh? exten => s-BUSY is the extension for s-BUSY |
| 20:54.45 | aydiosmio | right |
| 20:54.46 | aydiosmio | thanks |
| 20:54.49 | Juggie | np, gl. |
| 20:55.08 | Juggie | aydiosmio, if your going to use it alot, i suggest looking @ writing a macro for it. |
| 20:55.13 | rephorm | jarrod: thigns like *8? |
| 20:55.17 | jarrod | rep: yea |
| 20:55.17 | a1fa | De_mon : i need for no route to destination |
| 20:55.23 | jarrod | rep: during a call |
| 20:55.54 | De_mon | a1fa you mean... http://www.voip-info.org/wiki/view/Asterisk+addon+rate-engine |
| 20:56.30 | De_mon | hrm, thats not specifically for no route, but it does mention it |
| 20:56.36 | rephorm | jarrod: i don't have features set up yet (just got some phones in yesterday) but there is a weird pause between hitting * and a digit. like the polycoms 'dialplan' doesn't like it |
| 20:57.04 | GerbilWrk | is there a way to change the voicemail prompting? |
| 20:57.55 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 20:58.14 | De_mon | a1fa i can't find it, but +101 is a failure even, so if dial failes (no route and other reasons) it jumps to n+101 |
| 20:58.50 | aydiosmio | cents.gsm, dollars.gsm -- lovely |
| 20:59.41 | jarrod | rpe: i'll delete my dialplan on the polycom so it sends what i dial |
| 20:59.42 | jarrod | thanks |
| 20:59.54 | aydiosmio | and.gsm |
| 21:00.04 | rephorm | GerbilWrk: doesn't it use the vm-*.gs files? |
| 21:00.12 | GerbilWrk | not sure |
| 21:00.16 | rephorm | .gsm |
| 21:00.26 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 21:00.27 | rephorm | vm-instructions.gsm iirc |
| 21:00.49 | [TK]D-Fender | jarrod : x.T|#x.T|*x.T |
| 21:01.06 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-103-07.telkomadsl.co.za) |
| 21:01.08 | [TK]D-Fender | jarrod : and ImpossibleMatchHandling ="2". |
| 21:01.27 | [TK]D-Fender | jarrod : that'll tell Polycom's to take whatever the heck you feel like feeding it. |
| 21:02.06 | *** join/#asterisk Trakkasure (n=Sgemtum@adsl-068-153-217-253.sip.bct.bellsouth.net) |
| 21:02.10 | Lyfe | anyone know if there's a way to execute an AGI when a call in a queue is answered by an agent? |
| 21:02.34 | Lyfe | (or execute *anything*) |
| 21:02.37 | rephorm | when you guys provision your polycoms, do you just modify the xml configs or do you keep the originals and have an override file listed earlier? |
| 21:03.03 | rephorm | (i found a white paper recommending the later, which seems to work for most things but doesn't want to override th MESSAGE_WAITING warble) |
| 21:04.06 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:05.28 | blitzrage | everyone's calls terminating today? :) |
| 21:05.54 | [TK]D-Fender | rephorm :sip.cf primary, phone config next. I always build my from the base firmware pack and rebuild between revisions (1.5, 1.6, 2.0) |
| 21:06.04 | jarrod | but once in a call is what you push matched against a dialplan on your phone? |
| 21:06.39 | [TK]D-Fender | rephorm : <mac>-phone.cfg should never be used. Local settings = bad |
| 21:06.41 | blitzrage | dialplan execution stops during the Dial() app |
| 21:07.10 | syzygyBSD | Unless you cheat |
| 21:07.14 | [TK]D-Fender | jarrod : with my sample the phone won't care what you give it, it'll jsut pass it on the * to be refused or accepted |
| 21:07.18 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 21:07.49 | jarrod | well it seems like asterisk isnt reading my digits in order to access the feature |
| 21:08.06 | [TK]D-Fender | jarrod :During a call the phone doesn't process the DTMF for anything. Its in-channel. Dialplan is for when you are trying to dial. |
| 21:08.23 | [TK]D-Fender | jarrod : what feature? |
| 21:08.49 | jarrod | blindxfer - #1, or automon - *1, or my testfeature #9 |
| 21:08.58 | jarrod | the stream is going thru asterisk |
| 21:09.32 | [TK]D-Fender | jarrod : Sounds like you aren't using the dial app properly or your DTMF mode is wrong. |
| 21:09.32 | [TK]D-Fender | anthm : you around? |
| 21:09.50 | anthm | yes? |
| 21:09.50 | jarrod | i have to specify an option in the Dial statement to allow for features? |
| 21:09.53 | [TK]D-Fender | anthm : Just another big thanks for app_valetparking :D |
| 21:09.58 | [TK]D-Fender | jarrod : Yes. |
| 21:10.01 | anthm | welcome |
| 21:10.06 | jarrod | oh well dangit |
| 21:10.44 | [TK]D-Fender | anthm : Its so ludicrously simple and versative its kinda ridiculous :) Makes me wonder why we bother with * Parking at all.... |
| 21:10.53 | jarrod | i suck |
| 21:11.03 | anthm | shh you gonna start a flame war |
| 21:11.15 | [TK]D-Fender | :D |
| 21:11.19 | [TK]D-Fender | *I*?!?!?! |
| 21:12.04 | AlexCTI | TK-D Fender: Thanks.. it worked fine.. |
| 21:12.22 | [TK]D-Fender | AlexCTI : Quite welcome |
| 21:14.08 | [TK]D-Fender | anthm : Lt me guess... didn't want to disclaim it? |
| 21:14.22 | anthm | actually i believe it was over the name |
| 21:14.37 | anthm | they wanted me to change the name to not confuse it with the other parking |
| 21:14.42 | anthm | and i was not in the mood |
| 21:15.07 | justinu|laptop | it's always something |
| 21:15.15 | rephorm | [TK]D-Fender: on polycom's site they have a whitepaper (by the link to the admin guide) that recommends doing something like: CONFIG_FILES="${mac_addr}-registration.cfg, phone1.cfg, sip-local.cfg, sip.cfg" where phone1.cfg and sip.cfg are the unaltered ones that came with the firmware |
| 21:15.35 | rephorm | [TK]D-Fender: the other two just include the tags you need |
| 21:15.41 | anthm | i think in openpbx they copied a version of it and hooked it up to the sip phones park button or someting iirc |
| 21:15.49 | rephorm | [TK]D-Fender: i.e. ones that differ from the default |
| 21:16.03 | rephorm | [TK]D-Fender: so, when i new firmware comes out you don't have to rebuild, just replace the default ones |
| 21:16.21 | [TK]D-Fender | rephorm : Thats jsut psycho.... |
| 21:16.47 | [TK]D-Fender | rephorm : only run into issues in major releases and its so quick to build this stuff anyways I harly see the point... |
| 21:17.37 | [TK]D-Fender | anthm : Walks like a duck, quacks like a duck.... pick people sheesh... I'd like to toy around with the parking feature on my Polycom's next. Ever tried this yourself? |
| 21:17.44 | jarrod | is there anything that needs to be specified in Dial to use BlindXfer? |
| 21:18.03 | anthm | no |
| 21:18.08 | [TK]D-Fender | jarrod : You have a SIP phone.. you should NOT be using * DTMF features for transferring calls.... |
| 21:18.16 | aydiosmio | what genius didn't include a decimal round in perl? |
| 21:18.18 | rephorm | [TK]D-Fender: i kinda like how it keeps it cleanly separated. (esp. since those configs are about the worst example of xml i've ever seen) |
| 21:18.19 | jarrod | #1 is blind xfer |
| 21:18.24 | [TK]D-Fender | jarrod : And more than that... a aPOLYCOM. |
| 21:19.02 | [TK]D-Fender | aydiosmio : There are only 2.0 kinds of people.... those that understand decimals, and those that don't ;) |
| 21:19.17 | [TK]D-Fender | jarrod : #1 = waste of time. use your phone like it is intended..... |
| 21:19.33 | jarrod | i dont have a blind feature |
| 21:19.36 | Lyfe | sure it's not 2.5? there are people that can read it, but have no idea what it means. :) |
| 21:19.45 | [TK]D-Fender | rephorm : I only needed 2 levels. Global and Phone. |
| 21:19.52 | [TK]D-Fender | jarrod : yes you certainly do. |
| 21:20.05 | [TK]D-Fender | jarrod : Transer, then the blind soft-key. |
| 21:20.38 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 21:21.00 | [TK]D-Fender | jarrod : [Transfer] [Blind] *then* the number.... |
| 21:22.13 | *** join/#asterisk clyrrad1 (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 21:25.00 | *** join/#asterisk mtaht4 (n=m@dsl-63-249-108-30.cruzio.com) |
| 21:25.58 | [TK]D-Fender | jarrod : Found it? |
| 21:26.19 | hmmhesays | damnit I just farked my site up |
| 21:29.52 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
| 21:36.25 | a1fa | De_mon |
| 21:36.28 | a1fa | i figured it out |
| 21:36.47 | a1fa | exten => s-CHANUNAVAIL |
| 21:36.54 | a1fa | :) |
| 21:36.57 | a1fa | works like a charm |
| 21:41.09 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:43.26 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 21:48.16 | dos000 | anyone tried compiling asterisk with unixodbc and zaptel headers located in a non standard place ? |
| 21:48.33 | dos000 | this is driving me nuts |
| 21:48.57 | rene1 | an asterisk engineer, who could mm hack something like chan_whisper or chan_net2phone how much could earn in the marketplace? i would guess they can command a lot more money than other integrators as they are in the top of the food chain.. but how much in $ per year would such a guy can make? |
| 21:49.16 | rene1 | would suck a guy make |
| 21:49.18 | rene1 | such |
| 21:49.19 | rene1 | sorry |
| 21:50.11 | twisted[asteria] | uhh |
| 21:50.15 | twisted[asteria] | freudian slip? |
| 21:50.15 | eKo1 | rene1: That question is not specific enough so the answer is: it depends. |
| 21:50.43 | eKo1 | dos000: just mod. the Makefile so that it finds the headers. |
| 21:52.08 | brookshire | rene1: 1 billion dollars |
| 21:52.39 | eKo1 | brookshire: yeah right dr. evil |
| 21:53.55 | rene1 | twisted: no |
| 21:54.23 | rene1 | poor written english skills |
| 21:55.09 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:55.18 | *** join/#asterisk AJaymn (i=AJaymn@70.59.126.198) |
| 21:55.51 | dos000 | eKo1, prob is i cant see if this freakin thing detected the headers or no |
| 21:56.39 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:57.52 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-205-26.telkomadsl.co.za) |
| 21:58.45 | eKo1 | eKo1: Sure you can. If you have chan_zap.so and app_odbc.so, then you're golden. |
| 21:58.54 | eKo1 | and res_odbc* |
| 21:59.13 | rene1 | eko1: a guy who programs asterisk at the chan_, or app_ level, probably would make more money that someone who just hacks at AMI or AGI level, and much more than someone whose experience is limited to @home systems, right? on the other hand the demand for such specialized asterisk people might not be too high, so money-wise is there a ball-park for the income such knowledge generates when being an employee for somebody else? |
| 21:59.40 | rene1 | a so called market -rate for such knowledge? |
| 22:00.33 | rene1 | i know not every kernel-hackers is worth millions but on the average they probably are making more than say web developers |
| 22:00.45 | eKo1 | rene1: I have programmed several app_*'s. Do I get anything more for it? NO |
| 22:01.13 | eKo1 | Quite frankly, I'm debating whether I should move to AGI because it is much more flexible. |
| 22:01.44 | rene1 | really? weird well C programming isnt that |
| 22:01.49 | eKo1 | app_*'s are so inflexible when you need to upgrade them. |
| 22:02.07 | eKo1 | C programming is a PITA |
| 22:02.12 | rene1 | attractive to me |
| 22:03.11 | rene1 | there are some things that can only be done on C, and it is just plain hard, and yet no more money for it? :( |
| 22:04.37 | eKo1 | rene1: You mean C as a whole or the Asterisk's C API? |
| 22:05.07 | rene1 | asterisk C |
| 22:05.22 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 22:06.17 | rene1 | i have seen monster jobs for SIP engineers that pay 100K-120K / year |
| 22:06.23 | eKo1 | I can't think of any examples that I could do in a C app_* that I couldn't do with an AGI. Can you? |
| 22:06.30 | rene1 | sure |
| 22:06.32 | rene1 | channels |
| 22:06.40 | rene1 | chan_whisper_mode |
| 22:06.48 | eKo1 | I was talking about app_*'s. |
| 22:06.48 | rene1 | chan_skype net2phone |
| 22:06.50 | rene1 | ok |
| 22:07.01 | rene1 | well apps can be made in AGIs |
| 22:07.11 | rene1 | or AMI scripts |
| 22:07.25 | eKo1 | Channel drivers have to be made in C because the API is C. |
| 22:07.35 | rene1 | i mean you could implement asterisk app_queue in an agi |
| 22:07.38 | rene1 | but it could be slow |
| 22:07.46 | eKo1 | I think... |
| 22:08.44 | eKo1 | Hmm...I think you could code a channel driver in any other language. As long as you produce a .so module for it, then it doesn't matter what language it is in. |
| 22:09.02 | *** join/#asterisk saftsack (n=saftsack@p54A7D9ED.dip.t-dialin.net) |
| 22:09.23 | rene1 | well but lets talk about the depth of knowledge involved, that should matter shouldnt it? |
| 22:09.39 | eKo1 | Then again, I know nothing about programming dynamic modules in Linux in non-C languages. |
| 22:10.13 | *** join/#asterisk johnny2211 (n=matija@193.19.222.15) |
| 22:10.41 | rene1 | or is something one should learn for the love of it and hopefully find a way to profit from? |
| 22:12.12 | eKo1 | What is your point rene1? |
| 22:12.32 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 22:13.48 | rene1 | my point is learning asterisk and C programming to the level of creating chan_whatever is a worthy goal (money-wise_ |
| 22:13.51 | rene1 | ? |
| 22:17.12 | eKo1 | Worthy to who? |
| 22:22.11 | rene1 | ok rephrasing: i can code ami scripts for asterisk, my C is rusty and my knowledge of Asterisk C API is non existant. you said it didnt make a difference in terms of $$ for you to learn how to code app_*s do you think it could make a difference for others? you said that what you were doing for asterisk in C could be done by means of AGI scripting. what about chan_*s stuff, do you think such a developer would do mucho money? |
| 22:23.55 | eKo1 | The more expierence you have, the more marketable you are (generally). |
| 22:24.35 | eKo1 | expierence meaning time-wise and software-wise |
| 22:31.39 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 22:32.57 | dos000 | eKo1, is there a way i can tell i have odbc support in asterisk ? |
| 22:33.31 | eKo1 | You should have app_odbc and res_odbc stuff in your modules directory. |
| 22:35.27 | dos000 | eKo1, ok now .. i changed the make file in apps to add ODBC support however odbc is still not compiled. and i dont get errors ! |
| 22:36.34 | eKo1 | dos000: Post the relevant Makefile somewhere so I can look at. |
| 22:37.50 | dos000 | eKo1, i only removed the couple of comments ... |
| 22:38.17 | dos000 | http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage |
| 22:38.29 | eKo1 | err, then how is it supposed to find the odbc headers if you don't give it the path? |
| 22:38.54 | eKo1 | you need to hardcode it in there |
| 22:38.57 | dos000 | eKo1, ok .. i'll post it .. one sec |
| 22:39.43 | blitzrage | eKo1: there is an odbc.ini and odbcinst.ini files in /etc/ which creates the interface to the drivers |
| 22:40.40 | blitzrage | the inst file declares the drivers, then you define the interface (DSN) which Asterisk references and connects with |
| 22:40.56 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 22:41.09 | eKo1 | blitzrage: yes, I know. |
| 22:44.11 | dos000 | blitzrage, i have all that figured .. i just cant get asterisk to support odbc when i compile it |
| 22:44.47 | sb_mx | dos000, what we had to do was compile it and then install it with yum. we can't find why tho |
| 22:45.36 | dos000 | eKo1, http://pastebin.ca/125807 this is the first apps/Makefile |
| 22:46.54 | *** join/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 22:47.02 | blitzrage | yum install odbc-devel ? |
| 22:47.15 | blitzrage | without the dev packages asterisk won't create the modules |
| 22:49.57 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 22:50.19 | dos000 | eKo1, i posted the diffs to the main makefile ... http://pastebin.ca/125815 |
| 22:50.21 | rowter | anyone had problems with overruns on sangoma cards? |
| 22:50.35 | dlynes_laptop | rowter: overrun? no |
| 22:50.41 | dos000 | blitzrage, i am trying to install from source |
| 22:50.46 | dlynes_laptop | rowter: what kind of overrun problem are yhou getting? |
| 22:51.34 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 22:51.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:51.53 | dos000 | eKo1, it does not even complain about missing libs ... because it just never gets compiled |
| 22:52.01 | eKo1 | dos000: Looks like the Makefile expects the ODBC libs to be in /usr/lib. You need to make it point to your custom install. |
| 22:52.19 | eKo1 | It never complains because it doesn't find them. |
| 22:52.46 | rowter | dlynes_laptop, am noticing, overruns increases with big load of calls, and suddenly the E1 slot goes down, and I need to restart sever.. |
| 22:53.10 | dos000 | eKo1, which line is that ? |
| 22:53.13 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:53.18 | eKo1 | dos000: that is good question |
| 22:53.23 | dlynes_laptop | rowter: as in you're getting more calls than you can handle? |
| 22:53.33 | dlynes_laptop | i.e. more than 29 simultaneous calls? |
| 22:53.35 | dos000 | eKo1, this is 1.2.10 i am working with ! |
| 22:53.39 | eKo1 | The interesting think I'm noticing is that DUSE_ODBC_STORAGE never gets used anywhere but in that Makefile. |
| 22:53.46 | eKo1 | s/think/thing |
| 22:54.18 | eKo1 | Maybe that CFLAG is obsolete or something. |
| 22:54.23 | dos000 | eKo1, tow ! |
| 22:54.45 | rowter | dlynes_laptop, well yeah, its a 4E1 card, so am filling it up.. and overruns its going up.. but it handles the calls without problem, the problem is on the E1 that controls a rhino, that one is droping |
| 22:55.05 | eKo1 | wait, nevermind |
| 22:55.07 | eKo1 | I found them |
| 22:55.20 | dos000 | pray tell |
| 22:55.32 | *** join/#asterisk Givemelove (n=non@208.57.229.162) |
| 22:56.09 | *** join/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 22:56.20 | eKo1 | Yes, they're being used by app_voicemail. |
| 22:56.43 | Z_God | is there a command in asterisk to so it eat one or more digits? |
| 22:56.50 | dos000 | i just greped in my source to no avail ! |
| 22:56.56 | eKo1 | If I were you, I would just install the ODBC lib and headers in /usr/include and /usr/lib and be done with this. |
| 22:57.02 | Z_God | I'm using goto commands, but I want to drop the digit that's used to determine the goto |
| 22:57.05 | dos000 | eKo1, which version are you looking at ? |
| 22:57.15 | eKo1 | dos000: you need to grep for USE_ODBC_STORAGE. |
| 22:57.46 | dos000 | eKo1, i cant touch the installed odbc on the target system ! |
| 22:58.47 | *** join/#asterisk MatsK (n=mats@83.233.97.229) |
| 22:58.58 | *** join/#asterisk niter3 (n=niter3@d57-102-239.home.cgocable.net) |
| 22:59.16 | dos000 | eKo1, just explain how it determines to use odbc or not |
| 22:59.16 | niter3 | hey guys, just wondering how I can make an extension do dial the default context so I can see what it's like when somebody dials into the pbx |
| 22:59.43 | dos000 | eKo1, i have been staring at this makefile with no success |
| 23:00.50 | MatsK | ~book |
| 23:00.52 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:01.56 | MatsK | ~dialplan |
| 23:01.57 | jbot | rumour has it, dialplan is the thing configured in extensions.conf |
| 23:03.38 | MatsK | niter3: you set the context in sip.conf or what ever channel you are comming in over and match it with what you have the same context as "somebodys" has |
| 23:04.12 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 23:06.14 | niter3 | MatsK: huh |
| 23:06.22 | niter3 | i have a sepearte context for sip users |
| 23:06.29 | niter3 | and the dial in context is something different |
| 23:06.37 | niter3 | is there a way I can set an extension up to call the other context? |
| 23:07.10 | eKo1 | dos000: I didn't write the Makefiles so I can't give you specifics. |
| 23:07.52 | MatsK | niter3: huh, thats well explained in the ~book |
| 23:07.58 | dlynes_laptop | rowter: yeah...i'm lost there...I know nothing about channel banks |
| 23:08.15 | *** join/#asterisk mike800 (n=Michael@wsip-70-183-59-19.oc.oc.cox.net) |
| 23:08.20 | dlynes_laptop | rowter: you might try asking [TK]D-Fender though...I know he's worked with Sangomas and channel banks |
| 23:08.29 | rowter | ooh thanks |
| 23:08.38 | dlynes_laptop | rowter: you could also try contacting sangoma...you have a 3 yr warranty on the hardware |
| 23:09.03 | niter3 | got it |
| 23:09.09 | niter3 | MatsK: yah I found it |
| 23:09.13 | rowter | dlynes_laptop, yeah am trying to.. thanks.. |
| 23:09.15 | niter3 | Goto(context,s,1) |
| 23:09.21 | niter3 | ok that's what I wanted. |
| 23:09.25 | dlynes_laptop | rowter: yeah...they're pretty good about answering their phones |
| 23:09.41 | dlynes_laptop | rowter: if you get voicemail, just leave a voicemail; they'll call you back |
| 23:10.01 | rowter | dlynes_laptop, let me call them thanks. |
| 23:10.02 | mike800 | Anyone wanna talk about Asterisk and its faxing capabilities? :-) |
| 23:10.58 | *** part/#asterisk sp0n9e (n=sp0n9e@phpurge.com) |
| 23:11.09 | MatsK | biter3: nice the book is a deep well of knowledge |
| 23:11.17 | MatsK | ;-) |
| 23:11.54 | dlynes_laptop | mike800: depending on how you're attempting to do it, you might have good success, little success, or no success |
| 23:12.16 | mike800 | well, no success when trying to go T.30 over ulaw :-) |
| 23:12.17 | dlynes_laptop | mike800: there's even the lucky few that have no problems |
| 23:12.38 | dlynes_laptop | mike800: yeah...forget ulaw, unless it's in a lan environment |
| 23:13.04 | dlynes_laptop | and even then, it's dicey |
| 23:13.18 | mike800 | Well...thats my question...cause my sister has Voicepulse (their normal res service), and she's able to fax flawlessly |
| 23:13.30 | mike800 | Voicepulse uses Astrisk |
| 23:13.37 | dlynes_laptop | yep |
| 23:13.43 | dlynes_laptop | i think they do, anyways |
| 23:13.56 | mike800 | so, how are they so successful at it? |
| 23:13.59 | dlynes_laptop | but they probably also have expensive head end equipment to deal with faxes |
| 23:14.09 | mike800 | :-\ |
| 23:14.18 | dlynes_laptop | using t.38 |
| 23:14.22 | dlynes_laptop | not t.30 |
| 23:14.23 | mike800 | how is the t.38 implementation coming along in astrisk? |
| 23:14.44 | dlynes_laptop | She's using a sipura 2000 unit or something similar right? |
| 23:14.48 | dlynes_laptop | With two ATA ports? |
| 23:14.49 | mike800 | ya |
| 23:15.05 | dlynes_laptop | And she can only fax from the one ATA port, not the other one, right? |
| 23:15.06 | mike800 | and thats exactly what im using in my test environment |
| 23:15.27 | mike800 | Well, the unit is capable of handling 2 lines, and she only has VoIP service on line 1 |
| 23:15.38 | mike800 | so the fax and phone both connect into Line1 |
| 23:15.43 | dlynes_laptop | Oh |
| 23:16.07 | dlynes_laptop | Anyways...probably autodetecting fax and switch appropriately into different head end equipment |
| 23:16.11 | dlynes_laptop | That would be my guess |
| 23:16.22 | dlynes_laptop | Or, voicepulse has invested into some custom programming for asterisk |
| 23:16.36 | mike800 | so my best bet is having astrisk send faxes out over Zap channels? |
| 23:16.45 | dos000 | loader.c:325 __load_resource: /home/sswitch/dist/usr/lib/asterisk/modules/app_hasnewvoicemail.so: undefined symbol: odbc_smart_execute |
| 23:16.54 | dos000 | #$%@$^@$5 |
| 23:17.04 | dlynes_laptop | mike800: pri channels |
| 23:17.17 | rene1 | mike800: yes, fax over zap is the only thing reliable. even lan faxing over atas isnt |
| 23:17.44 | dlynes_laptop | dos000: load res_odbc.so |
| 23:17.57 | dlynes_laptop | dos000: then load app_hasnewvoicemail.so |
| 23:18.05 | mike800 | So, its fully possible to have a SIP ata (or iaxy?) conenct a fax machine to asterisk, and send a fax over a pots line conencted to a TDM400? |
| 23:18.45 | dlynes_laptop | mike800: yes, but as rene1 stated, it's not reliable |
| 23:19.14 | dos000 | dlynes_laptop, ok .. i see why ... res_odbc.so is not getting built because of the stupid makefile issue i have |
| 23:19.17 | mike800 | ohh...i get it |
| 23:19.56 | dlynes_laptop | mike800: as soon as a network becomes part of the equation, your reliability rate goes out the window |
| 23:20.08 | mike800 | So if I was to have the fax machine connect to the TDM400 using a FXS port, and send it out over FXO, then we're all good |
| 23:20.16 | niter3 | how can I play an mp3 in the background |
| 23:20.25 | niter3 | Backgroun(file.mp3) does not work |
| 23:20.26 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 23:20.29 | hads | mike800: Yeah, that usually works |
| 23:20.29 | dlynes_laptop | mike800: from what I understand, fxs ports are not terribly reliable either |
| 23:20.31 | niter3 | background that is |
| 23:20.34 | dos000 | if anyone understand the wisdom on how res_odbc gets built i would apreciate |
| 23:20.45 | dlynes_laptop | mike800: and that's why so many people end up using channel banks instead |
| 23:20.48 | niter3 | i can do MP3Player, but I need to play in the background |
| 23:20.50 | hads | dlynes_laptop: FXO -> FXS native bridge always works for me |
| 23:21.11 | sb_mx | dos000, this is what we do: |
| 23:21.13 | sb_mx | ./configure --disable-gui --sysconfdir=/etc |
| 23:21.13 | sb_mx | make |
| 23:21.13 | sb_mx | make install |
| 23:21.17 | dlynes_laptop | hads: ok, so why is it so many people on here have problems with it? or are they just too good for analog? |
| 23:21.18 | mike800 | hads, what device are you using? a TDM? |
| 23:21.52 | sb_mx | dos000, oh wait, that's not for res_odbc. that is for unixODBC |
| 23:21.58 | mike800 | apparently, digium states that their hardware doesnt support fax :-) |
| 23:21.59 | hads | dlynes_laptop: Probably going from PRI -> FXS over the PCI bus probably isn't as reliable. |
| 23:22.07 | dos000 | sb_mx, in unixodbc ?? i already have that compiled . i just cant get asterisk to figure where the include and libs of the odbc are located. |
| 23:22.15 | dlynes_laptop | hads: are you using a tdm400p, or an a200? |
| 23:22.16 | hads | But on the same card I've never had trouble yet. |
| 23:22.22 | hads | TDM400's |
| 23:22.24 | dos000 | sb_mx, i mean i mean this is freakin black magic to me so far |
| 23:22.29 | dlynes_laptop | ah |
| 23:22.45 | hads | Sorry, were you talking a200's? |
| 23:22.56 | dlynes_laptop | hads: no...he's talking about a tdm400p |
| 23:23.07 | hads | OK, I just butted in like normal ;) |
| 23:23.26 | dos000 | sb_mx, where is the logic that decides whether to build odbc or not ! .. shuld be in the make file right ? .. guess again ! |
| 23:23.28 | dlynes_laptop | hads: I've heard sangomas don't have an issue, but the people that are complaining about having to use channel banks were all using digium hardware |
| 23:23.50 | dlynes_laptop | hads: but yeah...maybe they were using pris and trying to use tdm400p's with te110p's or something |
| 23:24.11 | hads | Yeah, that's what I've heard too. But staying on the same card, i.e. all analog seems to work fine from my experience. |
| 23:24.21 | niter3 | yah this is gay |
| 23:24.34 | mike800 | thanks for the help |
| 23:24.45 | niter3 | i want it to play a mp3 and when I hit 1 it will automatically go to an extension, but instead i hit 1 then it starts wating for a extension to be dialed |
| 23:24.54 | sb_mx | dos000, that's why we compile it and then run "yum install MyODBC" . afterwards we compile libiodbc and then we run "yum install unixODBC" . unfortunately i cant remember exactly why we do it like this |
| 23:25.02 | dlynes_laptop | mike800: but yeah...faxing over voip is kinda hit and miss |
| 23:25.34 | mike800 | dlynes_laptop, ya...with most ITSPs it doesnt work, but I was surprised when it did with Voicepulse |
| 23:25.34 | hads | Actually, I do have one office with two TDM400's and faxes go over the PCI bus from one card to the other and that works too. That might be just luck though :) |
| 23:25.35 | dlynes_laptop | mike800: you get between 0% and 99% success ratios, depending on hardware, location, internet service provider, ... |
| 23:25.48 | mike800 | hahaha |
| 23:26.01 | dlynes_laptop | mike800: i bet if you try using voicepulse from someone else's house, it won't work so well |
| 23:26.13 | dlynes_laptop | mike800: your sister probably just got lucky |
| 23:26.25 | hads | Yeah, even putting fax over a LAN can be dubious. |
| 23:26.50 | rene1 | and those fucking g3 faxes make things much worse |
| 23:27.05 | dlynes_laptop | hads: yeah, but at least on a LAN, if you do all the cabling yourself, and you don't use dubious hardware, you can pretty much guarantee it to work |
| 23:27.19 | mike800 | rene1, what are g3 faxes? |
| 23:27.28 | hads | dlynes_laptop: Agreed. |
| 23:27.35 | dlynes_laptop | hads: just don't use crappy terminators like leviton and the like |
| 23:27.42 | dos000 | sb_mx, the issue is not about odbc .. the problem is i cant figure how asterisk decides to pull in odbc support ! i can get the freakin module built. .. |
| 23:27.44 | hads | Down with fax! |
| 23:28.34 | mike800 | dlynes_laptop, what do you recommend? (my house is wired with ALL Leviton |
| 23:28.40 | mike800 | (came that way) |
| 23:28.45 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 23:29.43 | rene1 | mike800: g3 faxes are newer faxes that can do 33.6 speeds and stuff like color and such |
| 23:29.59 | mike800 | ohh...is that t.34? |
| 23:30.10 | rene1 | mmm maybe i am not sure |
| 23:30.25 | rene1 | those are harder to use with * |
| 23:30.40 | dlynes_laptop | hahaha |
| 23:30.48 | dlynes_laptop | anyways |
| 23:31.08 | dlynes_laptop | No-name Taiwanese stuff is usually pretty good, as is Nortel and AMP |
| 23:31.29 | dlynes_laptop | Leviton is absolutely horrible though |
| 23:31.37 | dlynes_laptop | They don't keep a very good connection |
| 23:31.48 | dlynes_laptop | The cable used can also make a difference |
| 23:32.03 | dlynes_laptop | Mohawk and Taihan cable are both pretty good |
| 23:32.03 | mike800 | gotcha |
| 23:32.20 | mike800 | i use the no-name taiwanese rj-45 heads :-) |
| 23:32.31 | dlynes_laptop | If you're using anything less than CAT5E, you're just asking for trouble |
| 23:32.58 | dlynes_laptop | CAT5 isn't really up to the task for voip |
| 23:33.04 | *** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com) |
| 23:33.10 | watchy2 | does sjphone support multiple lines? |
| 23:33.10 | mike800 | whats the diff between Cat5 and Cat5e? |
| 23:33.20 | dlynes_laptop | mike800: frequency rating |
| 23:33.33 | mike800 | hmm...ok |
| 23:33.45 | dlynes_laptop | mike800: CAT5E has about three times the frequency rating of CAT5 |
| 23:33.47 | mike800 | but isnt VoIP just data passing over a line anyway? |
| 23:33.49 | MatsK | I use CET3 and it work nicely |
| 23:34.01 | MatsK | CET = CAT |
| 23:34.04 | dlynes_laptop | MatsK: do you try to pass fax over voip on CAT3? |
| 23:34.38 | dlynes_laptop | MatsK: and btw, CAT3 won't pass building code for data cabling, nor will CAT5 |
| 23:35.07 | dlynes_laptop | MatsK: CAT5E is minimum for passing code nowadays, and CAT6 is now being recommended |
| 23:35.23 | watchy2 | wow i didnt know sjphone charged |
| 23:35.25 | MatsK | Whats the difference if its IP with fax or voce 100Mbit is 100Mbit |
| 23:35.30 | dlynes_laptop | at least in North America (Canada and the USA) |
| 23:35.42 | dlynes_laptop | MatsK: signal loss makes a huge difference |
| 23:36.07 | MatsK | Well it passes IP nicely |
| 23:36.21 | MatsK | at 100MBit rate |
| 23:36.33 | dlynes_laptop | MatsK: normal ip transmissions can handle signal loss by compensating (tcp), fax going over udp loses a packet or two, and it's screwed (udp doesn't retransmit) |
| 23:36.51 | mike800 | doesnt asterisk only support udp at the moment? |
| 23:36.57 | dlynes_laptop | mike800: exactly |
| 23:37.15 | dlynes_laptop | mike800: so a lost packet makes a huge difference |
| 23:37.32 | dlynes_laptop | mike800: especially when you're talking packet critical stuff like faxing |
| 23:37.41 | mike800 | ya |
| 23:38.03 | dlynes_laptop | even one lost packet will usually screw up the fax transmission |
| 23:38.19 | mike800 | unless your fax supports error correction...right? |
| 23:38.36 | *** join/#asterisk vlt (n=daniel@dslb-088-073-236-118.pools.arcor-ip.net) |
| 23:38.40 | dlynes_laptop | mike800: even then...it's designed to handle line noise |
| 23:38.45 | dlynes_laptop | mike800: not data corruption |
| 23:38.59 | mike800 | hmm...true |
| 23:39.20 | mike800 | whats wrong with t.38? why hasnt there been more of a push to have it implemented in asterisk? |
| 23:39.21 | dlynes_laptop | mike800: so depending on where the packet got lost, it may or may not be able to recover |
| 23:39.38 | dlynes_laptop | mike800: probably because not many service providers support it |
| 23:40.03 | dlynes_laptop | mike800: until it's more widely implemented, I would imagine it'll take a back burner on asterisk's priority list |
| 23:40.25 | mike800 | o |
| 23:40.26 | mike800 | k |
| 23:40.35 | dlynes_laptop | mike800: but, when 1.4 comes out, it's supposed to have t.38 passthrough support |
| 23:41.12 | mike800 | cool |
| 23:41.15 | dlynes_laptop | mike800: it was supposed to come out last month, originally...no idea when it's coming out now |
| 23:41.29 | mike800 | well, i think they're shooting for reliability |
| 23:41.35 | dlynes_laptop | exactly |
| 23:41.37 | vlt | Hello all. I have a big problem: Sometimes when I initiate a dial command from asterisk's CLI the server hangs completely (even doesn't respond to pings) until the called phone answers. A few minutes ago I dialled a currently not available number ... Any idea how to get the server back to life again? |
| 23:41.50 | Rawplayer | when you are using pots its only possible to use one telephone right? |
| 23:41.55 | Rawplayer | also with call parking |
| 23:41.56 | dlynes_laptop | i'd rather see it late and reliable, than on time and unreliable, myself |
| 23:41.56 | Rawplayer | right? |
| 23:42.12 | mike800 | dlynes_laptop, definitely |
| 23:42.13 | dlynes_laptop | Rawplayer: nah...there's call waiting |
| 23:42.35 | dlynes_laptop | Rawplayer: if you're using an fxo port on a tdm card that is, and not a sipura unit |
| 23:42.42 | Rawplayer | i'am thinking about buying this one for that TDM01B: TDM400P + 1-port FXO bundle |
| 23:42.50 | dlynes_laptop | Rawplayer: i haven't found a way to handle call waiting on a sipura unit yet |
| 23:43.03 | dlynes_laptop | Rawplayer: but i'm not the only person using sipura units, either |
| 23:43.33 | *** join/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 23:43.47 | dlynes_laptop | vlt: stop doing that? use a normal phone? |
| 23:45.26 | hads | Anyone using the pickupexen feature (*8) with SIP phones? I'm getting some weird voice distortion if I do. |
| 23:45.31 | vlt | dlynes_laptop: I'm afraid it's too late now for stopping ... What is this function for when I can't use it? |
| 23:46.19 | hads | vlt: kill :) |
| 23:46.28 | dlynes_laptop | vlt: probably brief testing, but that's a guess on my part |
| 23:47.24 | vlt | hads: The server doesn't answer me anymore ... how to kill asterisk then? |
| 23:47.29 | dlynes_laptop | well, and it was probably a feature put in to get everyone stop whining about why there wasn't a dial command from the cli :) |
| 23:48.11 | vlt | dlynes_laptop: Yes, brief testing was exactly what I did. Mmh, but I didn't want to test that server but the called number ... :-( |
| 23:48.11 | hads | kill `cat /var/run/asterisk/asterisk.pid` |
| 23:48.41 | hads | vlt: Or you might try stopping it nicely first. |
| 23:49.08 | hads | like stop now from the CLI or /etc/init.d/asterisk stop |
| 23:49.14 | vlt | hads: If you have any idea HOW ... |
| 23:49.14 | dlynes_laptop | hads: heh...when asterisk gets locked up nicely like that, stopping it nicely doesn't usually work, and stop now often doesn't work, either |
| 23:50.00 | mike800 | Alright, I'm out... |
| 23:50.02 | hads | dlynes_laptop: Yeah, just covering myself incase someone pipes up and says 'Don't just kill it' :) |
| 23:50.23 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 23:50.51 | vlt | hads: I'd kill it ... twice ... with great pleasure ... if I only could ... |
| 23:51.20 | hads | vlt: Are you at the console or remote? |
| 23:51.58 | vlt | hads: remote (150 km) |
| 23:52.01 | dlynes_laptop | vlt: well, if you just want to kill it, and don't mind a memory leak, try the following: |
| 23:52.13 | dlynes_laptop | vlt: killall -9 safe_asterisk ; killall -9 asterisk |
| 23:52.34 | hads | Oh, I just realised you said it doesn't respond to pings. I guess that means you can't ssh in :) |
| 23:52.37 | dlynes_laptop | vlt: and then reboot it afterwards, so you can reclaim your memory leaks |
| 23:52.47 | dlynes_laptop | hads: oh...didn't know that |
| 23:52.58 | hads | Neither. |
| 23:53.06 | dlynes_laptop | niiice |
| 23:53.13 | vlt | indeed |
| 23:53.20 | dlynes_laptop | vlt: have a nice drive |
| 23:53.25 | vlt | ;-) |
| 23:53.28 | dlynes_laptop | vlt: grab a coffee on the way |
| 23:53.29 | hads | Hah, nasty |
| 23:53.43 | vlt | It's 01:53 am here |
| 23:53.52 | hads | Get two coffees then |
| 23:55.25 | hads | Remote access to the UPS? |
| 23:55.27 | logicwrath | Why doesnt this work "exten => s,n,Background(custom/Thankyou1)" if I have the .gsm file in the /var/lib/asterisk/sounds/custom directory |
| 23:55.41 | logicwrath | I have to remove the custom/ and then copy to the sounds directory |
| 23:56.13 | vlt | hads: No (could ask a neighbor to go to the basement and turn off fuse ...) |
| 23:56.31 | *** join/#asterisk tempest1 (n=asf@c-68-58-187-78.hsd1.sc.comcast.net) |
| 23:58.15 | vlt | I expierienced that behavior before when I was sitting local at the server. In that state when it sometimes(!) hangs waiting for the other phone to answer I can't even do a single key stroke on keyboard ... WTF could cause such a big show stopper? |
| 23:59.22 | hads | No idea, I've never used originate |