00:00.02 | pattieja | it crawled |
00:00.13 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
00:00.15 | Corydon76-work | Load is defined as the average number of processes in a short wait state. IO is considered to be a long wait state. |
00:00.25 | pattieja | and after MailScanner killed itself, it got much better |
00:01.03 | cypromis | imap servers |
00:01.03 | cypromis | about 40.000 mailboxes |
00:01.03 | cypromis | got sometimes far over 100.0 |
00:01.03 | cypromis | and still where useable |
00:01.20 | cypromis | since than don't really trust the LOAD value on linux and slowlaris |
00:01.42 | doughecka | haha |
00:01.50 | kapejod | just make sure your * runs as an RTAI softrealtime task ahead of the linux scheduler ;-) |
00:02.43 | kapejod | then you can run make bzImage -j without jittering your calls |
00:03.02 | doughecka | hah |
00:04.58 | kapejod | dont let the penguin touch your *! ;-) |
00:05.02 | [Barney] | am I missing something, or is there a bug in cdr_pgsql ? |
00:05.19 | [Barney] | it doesn't seem to be logging the accountcode if specified in config files |
00:05.50 | [Barney] | and I can't figure out how to turn on debug logging ... adding debug in logger.conf doesn't show and log entries from cdr_pgsql.c :( |
00:07.16 | Corydon76-work | It's not a bug in cdr_pgsql |
00:07.19 | *** join/#asterisk Noodleman (~tuckerm@adsl-66-72-30-42.dsl.kntpin.ameritech.net) |
00:07.35 | Corydon76-work | That occurs no matter which cdr log you use |
00:07.48 | [Barney] | cdr_csv seems to track the account code though |
00:08.17 | Corydon76-work | Oh, well, file a bug in the bugtracker then |
00:09.15 | [Barney] | yup ... going to ... just wanted to check I wasn't missing something before doing so |
00:09.32 | [Barney] | will see if I can post a patch at the same time |
00:15.07 | bkw_ | blah |
00:15.08 | bkw_ | blah |
00:15.20 | Marlow | bkw_ : interesting conversation :) |
00:15.28 | bkw_ | byte me! :) |
00:15.29 | bkw_ | haha |
00:15.42 | Marlow | bkw_ : that's what the dog is for .. |
00:16.03 | Marlow | :o) |
00:17.01 | bkw_ | dogs can't byte |
00:18.41 | denon | cypromis: man, openvpn is pretty cool .. I havent touched it yet .. |
00:18.44 | denon | but I think I may |
00:19.26 | nocnoc | damn why does my music on hold sounds like jabba the hut snorring? |
00:19.36 | cypromis | hmm I want to test it tonight on a really strange setup |
00:19.36 | cypromis | ;; |
00:19.55 | cypromis | linux -> linux/nat -> isp/nat -> public server |
00:20.03 | nocnoc | i've setup the -r parameter for mpg123 to 8000 and 44100 with the same results |
00:20.17 | denon | cypromis: you know, openvpn seems lightweight enough for an iax2 payload encryption |
00:20.29 | denon | wonder what latency is like |
00:20.40 | cypromis | denon: I used cipe for tht a lot |
00:20.51 | cypromis | worked nicely |
00:21.00 | Connor | nocnoc: You need to use mpg123 not mpg321, check to see if mpg123 is a sym link to mpg321, if so, download mpg123 make and compile it. |
00:23.35 | nocnoc | connor - im sure its mpg123 |
00:23.48 | Connor | double, nah.. TRIPPLE check it. |
00:23.57 | stonefly | Can the voltage on a PSTN line help you adjust tx gain and rxgain? I have three PSTN and two of them are 49.6v, and one is 43.7v. .. |
00:24.08 | Connor | I had the same problem... do a which mpg123 |
00:24.20 | Connor | then check it. |
00:24.57 | ScaredyCat | does it say it's mpg123 when you run it? |
00:25.17 | daork | file `which mpg123` |
00:25.25 | daork | and if its a binary |
00:25.39 | pattieja | Anybody know if there will be any merit to running multiple instances of Asterisk on the same box (if this is even possible) and IAX2 trunk them together to find out whether echo cancellation will get better? My reasoning is that if the software phones do not do echo cancellation (i.e., X-Lite (not sure of), gnophone, DIAX, etc.), then having Asterisk echo cancel twice might just work. |
00:25.49 | ScaredyCat | [root@ASTERISK mike]# mpg123 -v |
00:25.49 | ScaredyCat | High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. |
00:25.50 | ScaredyCat | Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. |
00:25.53 | daork | md5sum `which mpg123` and md5sum `which mgp321` and compare |
00:27.02 | nocnoc | High Performance MPEG 1.0 Audio Player for Layer 1,2 and 3. .. |
00:27.03 | nocnoc | Version 0.58 (97/04/10). Written and copyrights by Michael Hipp. |
00:27.21 | ScaredyCat | get 0.59r |
00:27.27 | *** join/#asterisk forkenb (~forkenb@dailup-219-90-63-226.appscorp.net) |
00:27.30 | nocnoc | will do |
00:28.04 | ScaredyCat | http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz |
00:28.24 | kapejod | pattieja: why do you think that iax2 trunking is related to EC? |
00:28.25 | *** join/#asterisk One (~DR@D5776FC9.kabel.telenet.be) |
00:28.45 | kapejod | even TDMoE wont help you, because it has no EC support |
00:29.00 | forkenb | I'm trying to handle timeout for a queue by putting a 't' extension in the queues context, I also have single digit extensions which work fine when pressed while in a queue, how come a t is not thrown when the queue times out, am I misunderstanding the functionality of timeout parameter in queues.conf |
00:30.35 | cypromis | u are |
00:30.38 | nocnoc | scaredycat - are you using the default parameters within the res_musiconhold.c file? |
00:30.41 | tclark | forkenb: you need a patch for that sse bug.digium.com |
00:30.45 | cypromis | but there is a patch for that somewhere in bugs.digium.com |
00:30.50 | ScaredyCat | timeout just means how long b4 it retries |
00:30.52 | atacomm | grrr, guess i should bother reading before I try doing........ where can I pull CPU usage statistics out (per cpu)? i've tried /proc/stat .... but after finding my bar graph staying steady found out thats number of seconds, whereas i'm looking for a realtime, per second type measurement.... |
00:30.59 | ScaredyCat | nocnoc: yes |
00:31.00 | cypromis | or even talk to bkw cause I think it was his patch :)) |
00:31.25 | forkenb | i c, I'll take a look at it....thanks |
00:32.18 | ScaredyCat | ^---- not THE One then, obviously |
00:35.13 | nocnoc | scaredycat - little bit better now. sounds like jurassic park's velociraptors |
00:35.20 | nocnoc | im getting closer ;) |
00:36.09 | ScaredyCat | bbl |
00:36.33 | ScaredyCat | ; Music on hold class definitions |
00:36.33 | ScaredyCat | ; |
00:36.33 | ScaredyCat | [classes] |
00:36.33 | ScaredyCat | default => mp3:/home/mp3,-z |
00:36.33 | ScaredyCat | loud => mp3:/var/lib/asterisk/mohmp3 |
00:36.34 | ScaredyCat | random => quietmp3:/var/lib/asterisk/mohmp3/ok,-z |
00:37.20 | ScaredyCat | nite all |
00:37.27 | *** join/#asterisk Celtic (Dax@rrcs-central-24-106-64-175.biz.rr.com) |
00:37.31 | heller | w |
00:37.33 | heller | ergh. |
00:38.04 | znoG | hey all, does Asterisk support MSN? (multiple subscriber numbers) |
00:38.35 | kapejod | znoG: it does when combined with capi |
00:39.06 | znoG | ah, good.. so i can do context routing based on the MSN |
00:39.23 | kapejod | sure |
00:39.32 | znoG | excellent |
00:39.39 | *** join/#asterisk killerbee (~Killer@ool-44c1013f.dyn.optonline.net) |
00:39.49 | killerbee | bkw_ |
00:39.50 | znoG | when you say combined with capi, you mean with the www.junghanns.net? |
00:39.54 | kapejod | znoG: even chan_modem_i4l will do this |
00:40.19 | kapejod | with the |
00:40.21 | znoG | i'm going to be using a card that has windows capi drivers, but not linux ones .. i'm hoping www.junghanns.net will work with it |
00:40.28 | kapejod | !info kapejod |
00:40.28 | kapejod | hmm..where is that bot tonight?? |
00:40.30 | znoG | kapejod: i mean with the channel capi driver from that site |
00:40.33 | kapejod | no, it wont. |
00:41.10 | znoG | i think i'm a little confused about how it all fits together |
00:41.38 | kapejod | your winblows capi card will not work |
00:41.55 | znoG | so i need actual linux capi drivers for the card... |
00:41.56 | bkw_ | killerbee yes |
00:42.11 | bkw_ | killerbee I gave up on cracking that ata via war dialing it.. i'm too impatient |
00:42.22 | killerbee | oh :< |
00:42.24 | killerbee | too bad |
00:42.31 | killerbee | just let it run... |
00:43.00 | kapejod | znoG: unless you find a way to run windoze drivers on linux |
00:43.37 | *** join/#asterisk xantus (~david@208.49.241.98) |
00:43.57 | znoG | kapejod: eek, not good. without those capi drivers its gonna have serious echo problems and you mentioned MSN won't work either.. |
00:44.07 | *** join/#asterisk NoCarrier (~NoCarrier@copper.voicepulse.com) |
00:44.08 | znoG | kapejod: i thought that junghanns site had a generic capi driver |
00:44.33 | kapejod | no, they dont. |
00:44.45 | bkw_ | HAHHAHA |
00:44.45 | bkw_ | From: Steve Totaro <stotaro@seepu.com> |
00:44.46 | bkw_ | Reply-To: asterisk-users@lists.digium.com |
00:44.46 | bkw_ | token: asterisk-users@lists.digium.com |
00:44.46 | bkw_ | Subject: Re: [Asterisk-Users] Radius on * |
00:44.46 | bkw_ | looks like critchy is especially bitchy.... |
00:45.03 | znoG | i had a reply setup but i don't really know this Steve guy so i passed on it |
00:45.19 | bkw_ | With all his whinging, if i didn't know any better, I'd suspect he was using a 2400 baud modem... |
00:45.19 | bkw_ | Now I'm off to reply a message and change the subject line.... |
00:45.19 | bkw_ | Andy |
00:45.28 | bkw_ | ScaredyCat you are funny |
00:45.48 | kapejod | i deleted all those radius and unsubscribe mails right away |
00:46.13 | JerJer | bkw_: read the next one... Andrew thinks i run VOnage ?!!?! |
00:46.35 | doughecka | LOL |
00:46.38 | xantus | doh |
00:47.15 | kapejod | JerJer: you dont run vonage? ;-) |
00:47.36 | JerJer | i don't even wish to run vonage.... i wouldn't have gotten in bed with Cisco |
00:48.11 | doughecka | farfone dev is shut down for the day! |
00:48.12 | JerJer | I can tell andrew hasn't acutally tried to deploy a carrier-class VoIP system... |
00:48.37 | doughecka | carrier pidgion? |
00:49.18 | denon | bkw_: you gave up cracking the ATA? |
00:49.20 | forkenb | anybody know of any patches that break into musiconhold to say a message and then return to the music on hold, for example playing promotions randomly while playing a song when a promotion isn't being played? |
00:49.28 | denon | bkw_: Why? you could just let it run all week or somethin right? |
00:50.34 | *** part/#asterisk stonefly (~trillian@toby.stoneflytech.com) |
00:52.48 | bkw_ | denon it caused drama on my box.. haha |
00:52.53 | bkw_ | I migth try it again.. who knows |
00:54.54 | znoG | kapejod: i guess the only card that has full capi support in Linux is Digium's ones right? |
00:56.21 | pino | znoG: i'm quite happy with AVM's cards |
00:56.31 | mishehu | bah. |
00:56.38 | mishehu | I hope my 7960 arrives soon... |
00:56.39 | kapejod | znoG: no digium card has capi support |
00:56.52 | znoG | kapejod: do you know any that do? |
00:57.11 | mishehu | the only thing you'd really need capi for is faxing. |
00:57.23 | mishehu | I can't think of any other reason for capi |
00:57.23 | kapejod | znoG: all AVM card, all active Eicon cards, all primuxisdn cards |
00:57.37 | pino | znoG: if you're australian, you may want to check the mailing list archives... |
00:57.38 | kapejod | mishehu: how about BRI? |
00:57.44 | znoG | kapejod mentioned other features such as MSN (multiple subscriber numbers) |
00:57.57 | znoG | pino: thanks |
00:58.13 | pino | znoG: there are not so many cards with the infamous a-tick certification. |
00:58.22 | znoG | except for AVM right? |
00:58.27 | znoG | AVM!Fritz or whatever its called |
00:58.41 | znoG | to be honest i couldn't care much for the A-tick.. :) its only going to be used in a SoHo environment |
00:58.44 | znoG | as long as it works basically.. |
00:59.11 | pino | i have a fritz and i'm very happy with it. (and with kapejod's chan_capi, of course.) |
01:00.16 | Celtic | Anybody know if either IAX or Zap channels understand the concept of *called number* as different from *calling number* ? |
01:00.53 | cypromis | they do |
01:01.55 | Celtic | cypromis: If I forward a number to an 800 number that then comes to * over IAX is there any way to pick that info up in extensions.conf ? |
01:02.24 | Celtic | cypromis: i.e. detect that the call was originally placed to the first number |
01:02.58 | znoG | pino: hmm what sort of faxing support does chan_capi add? |
01:03.15 | pino | none. capi drivers may do that, though. |
01:03.18 | mishehu | kapejod: I didn't remember if digium had any bri cards or not. |
01:03.35 | pino | mishehu -- they don't :) |
01:03.41 | tclark | Celtic: see ${RDNIS} Redirected Dial Number ID Service |
01:03.52 | kapejod | somebody should make some then. |
01:04.32 | *** join/#asterisk One (~DR@D5776FC9.kabel.telenet.be) |
01:04.36 | pino | znoG: for example, with the fritz (and, i guess, most other CAPI 2 aware cards) you can install capi4hylafax and there you go. |
01:04.49 | znoG | pino: sorry for all the questions.. just trying to figure out what exactly chan_capi is. Sounds like there are boards with CAPI support, and chan_capi is just a linux driver to use those features |
01:04.53 | Celtic | tclark:Do you know if Nufone pass such info onto IAX ? |
01:05.08 | znoG | pino: ahh so on incoming faxes it just palms it off to hylafax? is capi4hylafax part of the chan_capi package? |
01:05.28 | JerJer | what's the diff between DNIS and RDNIS? |
01:05.29 | pino | no, capi is a standard interface, and capi drivers support the capi interface. |
01:05.30 | Celtic | tclark: or do you happen to know if the SIP driver puts anything in there ? |
01:05.49 | pino | both chan_capi and capi4hylafax use the capi interface to talk to ISDN boards. |
01:06.04 | kapejod | capi == Common isdn API |
01:06.41 | znoG | i seee |
01:06.50 | znoG | so without capi, how do you normally interface with the ISDN board? |
01:06.56 | pino | if i can make it very simple, think of alsa. some sound boards have alsa drivers, and then all alsa-aware software can talk to them. |
01:07.30 | pino | many non CAPI-aware boards can talk to * through the isdn4linux channel. |
01:07.33 | kapejod | znoG: some people use isdn4linux |
01:07.35 | pino | i was not happy at all with it, though. |
01:07.45 | *** join/#asterisk bobman (~bobman@me-sebago-cmts1b-15.agstme.adelphia.net) |
01:07.59 | znoG | so capi drivers essentially have full control of the card and depending on the features of the capi driver, you can use them. |
01:08.31 | bevins | Is this a permissions issue? |
01:08.31 | pino | znoG: yes! (or at least I think so) |
01:08.31 | bevins | x=0, open writing: /var/spool/asterisk/voicemail/local/31/INBOX/msg0010 format: wav49, 0x80dbf48 |
01:08.57 | znoG | pino: so capi4hylafax talks to chan_capi which talks to the ISDN board... right? :) |
01:09.14 | bevins | I get this when voicemail tries to record a message |
01:09.14 | kapejod | no |
01:09.28 | Celtic | JerJer: DNIS is the number called, RDNIS is the number that passed the call on to you - I think... |
01:09.30 | kapejod | capi4hylafax talks to capi4linux |
01:09.37 | kapejod | chan_capi also talks to capi4linux |
01:10.08 | znoG | i see.. and the difference between capi4linux and chan_capi is? :) |
01:10.20 | pino | (your board) <-> (board-specific drivers) <-> (capi4linux driver) <-> (any software) |
01:10.28 | UnixDawg | ok who here is on fbsd |
01:10.32 | pino | where any software may be chan_capi or capi4hylafax, in this case. |
01:10.36 | UnixDawg | other then me |
01:11.18 | znoG | pino: so board-specific drivers have to support capi as well? |
01:11.34 | bevins | has anyone seen this message? x=0, open writing: /var/spool/asterisk/voicemail/local/31/INBOX/msg0010 format: wav49, 0x80dbf48 |
01:11.49 | UnixDawg | I need the 2 lines that go in sysct.conf |
01:11.58 | pino | znoG: if your own board supports CAPI under linux, then you can use CAPI, otherwise you're stuck with either isdn4linux or nothing :) |
01:12.28 | kapejod | and you can still run isdn4linux on top of capi4linux (just to complete your confusion) |
01:12.41 | znoG | pino: i see. there is this board, NetJet ISDN Card, which has Windows CAPI drivers but not Linux ones... i thought there may be some generic capi driver for capi-enabled cards |
01:12.44 | znoG | kapejod: yep, it sure did.. :) |
01:12.55 | kapejod | so every capi user can enjoy the latency and echo of chan_modem_i4l :) |
01:13.11 | znoG | i've never had ISDN so this is all new to me.. :) |
01:13.16 | *** join/#asterisk cypromis (~michael@217.11.142.161) |
01:13.23 | kapejod | wb cypromis |
01:13.33 | cypromis | aloha |
01:13.37 | cypromis | :) |
01:13.54 | pino | znoG: active boards usually support capi in hardware; some nice passive boards do that in software, like the fritz. |
01:14.14 | kapejod | pino: i disagree |
01:14.26 | kapejod | that is not a nice passive board, after all it's binary only!!! |
01:14.56 | pino | kapejod: it is :( could we say it's a nice passive board with a tied-down binary capi driver? |
01:15.26 | kapejod | well...there are much nicer passive boards which unfortunately lack a capi driver... |
01:15.33 | pino | basically, it's not nicer than many other boards, but the folks at AVM - for obvious reasons - developed a software CAPI implementation that runs on, well, just their own passive boards. |
01:16.08 | pino | kapejod: someone should make them... ;)) |
01:16.26 | kapejod | they could look like this: http://62.8.140.152/bri/ |
01:16.44 | jsharp | Buh |
01:17.27 | pino | pfff, kapejod, that's science fiction. ;) |
01:17.44 | kapejod | yeah, the gimp can do wonders ;) |
01:17.45 | *** join/#asterisk MagicMan (~alm971@APointe-a-Pitre-101-1-6-172.w81-249.abo.wanadoo.fr) |
01:18.48 | pino | and in such a fictional board, would the dip-switches for NT/TE or something else? |
01:19.03 | pino | *be |
01:19.34 | kapejod | the dip switches near the fictional rj45 plugs would be to enable 100 ohm termination |
01:19.41 | kapejod | the other dip switches would be just for fun |
01:19.55 | denon | cue the fanfare |
01:19.57 | denon | :) |
01:20.23 | pino | i'd only seen those near the plugs... :) |
01:20.41 | kapejod | everybody has his own imagination ;) |
01:21.40 | denon | kram: thanks for fixing that stuff in cvs |
01:24.00 | pino | good night everyone close to this time zone :) |
01:24.11 | kapejod | n8 pino |
01:24.28 | denon | hmm .. first cvsup in weeks .. |
01:26.29 | Celtic | Hohum - 3 IAX feeds and no DNIS or RDNIS apparent on any of them - anybody getting DNIS or RDNIS info over an IAX feed ? |
01:31.48 | Celtic | <PROTECTED> |
01:32.20 | Celtic | or maybe people have beer and don't care :-) - time for my beer methinks |
01:32.41 | *** join/#asterisk Porta (~sockpuppe@69.1.86.32) |
01:34.20 | *** join/#asterisk vefrra (~vefrra@hoochie.digium.com) |
01:39.59 | *** join/#asterisk espen2k (~espen2k@hoochie.digium.com) |
01:40.03 | *** join/#asterisk your_nick (~Ashmed@193.10.185.3) |
01:46.34 | *** join/#asterisk elusive1 (~konversat@19.chicago-12rh15-16rt.il.dial-access.att.net) |
01:49.09 | *** part/#asterisk elusive1 (~konversat@19.chicago-12rh15-16rt.il.dial-access.att.net) |
01:51.47 | denon | the new IAX phone chipset: http://www.soekris.com/ :) |
01:54.39 | ReG-Hexer | huh |
01:54.42 | ReG-Hexer | new phone? |
01:54.48 | denon | was kidding |
01:55.02 | denon | pretty hefty for a phone |
01:55.18 | bkw_ | no its nufone |
01:55.21 | kapejod | a bit slow for a phone ;) |
01:55.23 | bkw_ | NUFONE.. learn it |
01:55.39 | ReG-Hexer | lol |
01:55.47 | denon | bkw_: why didnt you leave your ata186 crackin? |
01:56.08 | bkw_ | ADD |
01:56.13 | bkw_ | lost intrest in the project |
01:56.17 | denon | but you coulda just left it under your desk or somethin |
01:56.19 | bkw_ | was taking too long |
01:56.22 | denon | it was already running |
01:56.33 | bkw_ | no the * server started to flip smooth out after about 8 hours of wardialing |
01:56.47 | bkw_ | was scrolling the dtmfsend's faster than it was accually dialing them |
01:56.51 | *** join/#asterisk loko_moko (loko-moko@pool-151-201-225-244.pitt.east.verizon.net) |
01:56.53 | bkw_ | not sure if it was skiping over some or what |
01:56.54 | knight- | bkw, lost interest in what project? |
01:57.08 | denon | bkw: huh .. dialing too fast? |
01:57.12 | denon | saw your changed the delay |
01:57.15 | denon | your/you |
01:57.26 | bkw_ | denon yes.. I might pick it back up next week |
01:57.46 | denon | sent myself a test message .. havent gotten the email yet :\ |
01:58.15 | denon | it uses the same confs and stuff right? |
01:58.27 | denon | dont need to change anything if I was using voicemail1 before do I? |
01:58.35 | bkw_ | shouldn't |
01:58.54 | knight- | bkw, what project? |
01:59.25 | bkw_ | wardialing a vonage locked ata |
01:59.40 | bkw_ | would take about 20 days to crack it |
01:59.41 | denon | arrrrrrg |
01:59.42 | doughecka | wardialing? |
01:59.49 | bkw_ | go look up wardialing |
01:59.52 | denon | mark said he added those patches |
01:59.55 | doughecka | yea, but how? |
01:59.57 | denon | wtf |
02:00.05 | bkw_ | doughecka an agi |
02:00.07 | doughecka | you trying to guess the password? |
02:00.09 | bkw_ | and a sample.callfile |
02:00.12 | bkw_ | yes |
02:00.18 | doughecka | on the locked vonage thing |
02:00.25 | bkw_ | yes |
02:00.26 | bkw_ | works too |
02:00.30 | bkw_ | tested it |
02:00.31 | doughecka | hmm |
02:00.41 | knight- | locked ata? |
02:00.42 | doughecka | so how do you access it? |
02:01.19 | doughecka | I have the resources at work that I can dedicate to it |
02:01.41 | denon | bkw_: hrm .. http://bugs.digium.com/bug_view_page.php?bug_id=0000542 |
02:01.54 | denon | bkw_: and mark shows that its fixed in the changelogs |
02:02.05 | doughecka | only catche is that I get to keep the ATA once I crack it |
02:02.09 | denon | bkw_: and yet: Content-Type: audio/x-wav; name="msg0034.gsm" |
02:02.13 | doughecka | catch |
02:02.28 | blll | knight-: wow, everyone really does show up here eventually |
02:02.39 | knight- | blll :) indeed |
02:02.56 | knight- | whats a locked ata? |
02:03.20 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
02:03.36 | doughecka | a cisco ATA that vonage "gives" away for thier service |
02:03.44 | doughecka | they locked it, and this is the only way to crack it |
02:03.56 | bkw_ | they dont give it away |
02:04.00 | bkw_ | you pay for it |
02:04.04 | doughecka | yea yea |
02:04.05 | doughecka | :P) |
02:04.09 | bkw_ | YOU PAY |
02:04.10 | bkw_ | trust me |
02:04.22 | knight- | ahhhhhh |
02:04.24 | doughecka | so how are you trying to brute force it? |
02:04.41 | doughecka | does it have a web interface or anything |
02:04.51 | doughecka | telnet? :) |
02:04.55 | denon | dtmf |
02:05.18 | denon | it has an fxs port |
02:05.20 | doughecka | so the ATA listens for a special command? |
02:05.28 | denon | when you put it into command mode, yes |
02:05.31 | doughecka | a special number |
02:05.45 | denon | more like a big red button |
02:05.46 | blll | how about calling them up and asking them? |
02:05.46 | denon | literally. |
02:05.51 | blll | anyone try that? :) |
02:06.04 | doughecka | so I plug a phone into it and dial a special number, and it turns on a command mode? |
02:06.31 | doughecka | blll: hmm, give me the number for an internal tech.. I need to learn social engineering anyway |
02:07.39 | denon | doughecka: you push a big red button and it asks for a command |
02:08.02 | doughecka | on the ata its self |
02:08.12 | denon | yes |
02:08.20 | denon | but you can loop through lots of times with only one button push |
02:08.22 | ReG-Hexer | anyone has any documentation on how asterisk transport layer works? |
02:08.26 | denon | (to answer your next Q) |
02:08.31 | doughecka | hah |
02:08.42 | doughecka | is there a limit on how many times before it resets? |
02:08.54 | denon | not that bkw has seen I dont think |
02:08.59 | doughecka | interesting |
02:09.09 | doughecka | I could let my server do it, since its not doing anything right now |
02:09.25 | doughecka | how is bkw knowing that it let him in? |
02:09.44 | doughecka | the server will keep trying numbers wont it? |
02:10.00 | doughecka | hah |
02:10.48 | *** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net) |
02:11.32 | *** join/#asterisk dguisinger (~dan@206.230.56.197) |
02:11.51 | doughecka | bkw_: well, if you get tired of wardialing the thing, you can send the ATA to me :) |
02:12.28 | doughecka | I can get the thing have you tried, as the password, the number that it was assigned to? |
02:12.53 | doughecka | have you tried using the number that it was assigned to? |
02:18.08 | *** join/#asterisk marrandy (~marrandy@209.216.76.1) |
02:20.20 | marrandy | Hello. I am now trying to move my grandstream sip phone from FWD to the asterisk box. Despite a number of attempts and changes, I am seeing registration failed errors |
02:20.50 | dguisinger | any idea why * isn't listening on port 5060? my config file from my eyes looks right..... |
02:21.27 | marrandy | It is on a static IP 192.168.1.70. The asterisk is on 192.168.1.1. First, does anyone have a good setup for the sip.conf file |
02:28.24 | UnixDawg | anyone here have the libpri patch for fbsd |
02:33.05 | tholo | libpri is completely useless until you have zaptel. |
02:33.47 | *** part/#asterisk jmb-ct (~trillian@solo.staff.chagres.net) |
02:34.05 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
02:42.12 | *** join/#asterisk nassy (~nassy@24-193-228-121.nyc.rr.com) |
02:44.20 | UnixDawg | ok I need the noinvite lines for a grandstream |
02:45.20 | marrandy | <PROTECTED> |
02:45.20 | marrandy | <PROTECTED> |
02:45.21 | marrandy | <PROTECTED> |
02:45.21 | marrandy | NOTICE[81926]: File chan_sip.c, Line 5215 (handle_request): Registration from '<sip:206@192.168.1.1>' failed for '192.168.1.70' |
02:45.43 | *** join/#asterisk Celtic (~dax@user-0cdv656.cable.mindspring.com) |
02:46.16 | marrandy | What is this registration failure all about. I can call and talk between extensions |
02:47.55 | UnixDawg | what are the disallow lines for sip grandstream phones |
02:48.08 | UnixDawg | I know there are 2 set to no but I cant find them |
02:50.41 | *** join/#asterisk Adam_ (~Adam@pseudo.teragen.com.au) |
02:50.45 | atacomm | ummm.....anyone know why my phone company would be giving me a call cannot be completed as dialed message when sending a 10 digit number out a PRI? |
02:51.24 | Stealth_Man | atacomm: try to put "1" |
02:51.32 | atacomm | tried that too |
02:51.44 | Stealth_Man | atacomm: you need to ask them how are they expecting to receive your digits |
02:51.51 | Adam_ | what's a cool name for soft phone software |
02:51.52 | izo | atacomm ; check dialplan |
02:51.59 | atacomm | izo? |
02:52.11 | izo | you can set in pri message if its national or internatioal format |
02:52.19 | atacomm | Stealth: well they cant expect them in too many different formats |
02:52.28 | UnixDawg | ok who here has a grandstream working |
02:52.34 | atacomm | it should be national, i've got the PRI set to National ISDN 2.... |
02:52.36 | Connor | make sure you have the correct switch type. |
02:52.36 | Stealth_Man | izo: is also rigth about Numbering type ... |
02:52.40 | UnixDawg | I need 2 lines to get mine back working |
02:52.47 | atacomm | i can receive incoming just fine |
02:52.54 | izo | <PROTECTED> |
02:52.57 | izo | check your zapata.conf |
02:53.10 | izo | pridialplan |
02:53.20 | atacomm | switchtype = national |
02:53.20 | atacomm | signalling = pri_cpe |
02:53.24 | UnixDawg | I need the disallow invite lines |
02:53.25 | Stealth_Man | Can someone tell UnixDawg 2 lines ???? |
02:53.27 | izo | pridialplan= ?? |
02:53.29 | atacomm | i dont have pridialplan |
02:53.34 | izo | so you need it |
02:53.44 | izo | try pridialplan=unknown |
02:53.52 | izo | and if it doesnt work |
02:53.57 | atacomm | lol, what are the pridialplan options? |
02:53.58 | izo | try national |
02:53.59 | Stealth_Man | izo: do you know settings for GS phones ? 2 lines for allow.disallow ... |
02:54.11 | izo | Stealth_Man : I have no idea :-( |
02:54.13 | Connor | You MUST make sure you have same switch type as telco.. I had a problem, they had NI2 and I had NI.. Didn't work.. We changed to DMS100 and it just worked. |
02:55.56 | Stealth_Man | ANyone has GS phones settings ??? |
02:56.01 | Stealth_Man | izo: thanks :) |
02:56.19 | UnixDawg | stealth hol on I almost have it |
02:56.53 | Stealth_Man | heh .. unixdawg: I am trying my best :)) |
02:57.22 | atacomm | hmm, cant seem to get either of those to work |
02:57.55 | Connor | what's the telco using ? |
02:58.02 | izo | atacom : do you ahve pri debug span 1 set ? |
02:58.09 | UnixDawg | i got it |
02:58.09 | UnixDawg | thnks |
02:58.17 | atacomm | i did, its not now after restarting |
02:58.42 | marrandy | Stealth_Man: you mean on the web setup ? |
02:58.48 | izo | well I could almost bet thats the issue with dialing plan and number format |
03:00.41 | *** join/#asterisk UnixDawg (~UnixDawg@ny-lasalle6c-203.buf.adelphia.net) |
03:01.04 | UnixDawg | exit |
03:02.32 | marrandy | Stealth_Man: you still there ? |
03:04.12 | Stealth_Man | yes |
03:04.36 | marrandy | O.K then - anyone else using SIP phones with static IP |
03:05.01 | Adam_ | yea... |
03:05.07 | marrandy | Stealth_Man: so are you talking about the actual GS config ofr the sip config |
03:05.24 | Stealth_Man | Adam: can i ask you about config for sip.conf for GS phone ? |
03:05.34 | Adam_ | yea |
03:05.45 | UnixDawg | ok it should be registring |
03:05.50 | UnixDawg | hmmm |
03:06.01 | Stealth_Man | unix: not really ... i odn;t know hwy |
03:06.03 | Adam_ | it's pretty simple |
03:06.27 | Adam_ | actually, very simple :) |
03:06.30 | Stealth_Man | [7000] |
03:06.30 | Stealth_Man | type=friend |
03:06.30 | Stealth_Man | host=dynamic |
03:06.30 | Stealth_Man | username=7000 |
03:06.30 | Stealth_Man | secret=testsecret |
03:06.32 | Stealth_Man | dtmfmode=rfc2833 |
03:06.33 | Stealth_Man | mailbox=7000 |
03:06.35 | Stealth_Man | context=sip |
03:06.37 | Stealth_Man | callerid="ext. 7000" |
03:06.39 | Stealth_Man | nat=yes |
03:06.41 | Stealth_Man | here it is ... |
03:06.43 | Stealth_Man | what's wrong here ? |
03:06.48 | Adam_ | what happens? |
03:07.07 | UnixDawg | context=admin |
03:07.07 | UnixDawg | canreinvite=no |
03:07.07 | UnixDawg | reinvite=no |
03:07.10 | UnixDawg | add those 3 lines |
03:07.10 | Stealth_Man | actually i have static ip ... maybe i need to remove host=dynamic ... |
03:07.16 | UnixDawg | no |
03:07.21 | UnixDawg | leave it |
03:07.27 | Stealth_Man | unix: context is differnet :) |
03:07.28 | Adam_ | what's the problem? |
03:07.40 | Stealth_Man | i have GS with this setting and ATA |
03:07.42 | Stealth_Man | on same LAN |
03:07.44 | UnixDawg | yes but you set a context for each phone |
03:08.02 | Stealth_Man | shell I set context for each phone or not ?! |
03:08.15 | Stealth_Man | i can;t dial anywhere :) even from 7000 to 7001 |
03:08.30 | Stealth_Man | 7000 is GS ; 7001 is ATA186 v2.16.1 |
03:08.32 | Adam_ | so it registers correctly? |
03:09.08 | Stealth_Man | *CLI> sip show peers |
03:09.08 | Stealth_Man | Name/username Host Mask Port Status |
03:09.08 | Stealth_Man | 7001/7001 192.168.1.10 (D) 255.255.255.255 5060 Unmonitored |
03:09.08 | Stealth_Man | 7000/7000 192.168.1.103 (D) 255.255.255.255 5060 Unmonitored |
03:09.08 | Stealth_Man | fwd/75574 192.246.69.223 255.255.255.255 5060 Unmonitored |
03:09.30 | Adam_ | looks good (?) |
03:09.38 | Stealth_Man | what is (D) |
03:09.45 | Adam_ | dynamic |
03:09.51 | Adam_ | i think ? |
03:09.57 | Stealth_Man | shit ... maybe i need to change it ... |
03:10.01 | Adam_ | nah nah |
03:10.02 | Stealth_Man | let me try to remove |
03:10.03 | Adam_ | just leave it |
03:10.08 | Stealth_Man | you think so ? |
03:10.11 | Adam_ | just leave it |
03:10.17 | Stealth_Man | i have static ip ... |
03:10.23 | Stealth_Man | but it shouldn;t be really matter ... |
03:10.31 | Adam_ | it's a restriction on who can use it |
03:10.58 | Stealth_Man | [sip] |
03:10.58 | Stealth_Man | ;exten => 7000,1,Dial(SIP/7000,20,tr) |
03:11.06 | Stealth_Man | include => sip |
03:11.11 | Adam_ | i see the prob |
03:11.15 | Adam_ | oh no i don't |
03:11.24 | UnixDawg | mine works |
03:11.24 | Stealth_Man | no no .. |
03:11.27 | Stealth_Man | it is not commented |
03:11.44 | Adam_ | so exactly what's the problem |
03:11.49 | Adam_ | (i assume you're doing sip debug) |
03:11.54 | Stealth_Man | yes |
03:12.03 | Stealth_Man | and i can;t figure out what is the prob :) |
03:12.09 | Stealth_Man | just can;t dial ... |
03:12.36 | Stealth_Man | it is like ringing ,, but i nreality i i have no ringing on secon phone .. |
03:12.48 | Stealth_Man | i am trying 7000 (GS) to call 7001 (ATA) |
03:13.07 | Stealth_Man | once I had success from 7001 to 7000 |
03:13.12 | Stealth_Man | but not vice versa.. |
03:13.13 | Adam_ | did u try unixdawg params? |
03:13.31 | Stealth_Man | yeh... same |
03:13.45 | UnixDawg | stealth you need to go back into the phones |
03:13.52 | UnixDawg | and reset them up |
03:14.08 | UnixDawg | I have a feeling its in the phone where you did not get it all setup |
03:14.25 | Stealth_Man | hmmmm |
03:14.28 | Stealth_Man | maybe ... |
03:14.31 | Stealth_Man | but what ?! |
03:14.47 | Stealth_Man | you mean reset config ? |
03:14.57 | Stealth_Man | andti start from the scratch ? |
03:15.01 | UnixDawg | give me a min I have mine alsmost done |
03:15.12 | Stealth_Man | ok ok |
03:15.36 | znoG | anyone using a grandmaster budgetone? |
03:15.55 | UnixDawg | thats what we are setting up now |
03:16.00 | daork | you mean a grandstream? |
03:16.08 | znoG | oops, yep, grandstream |
03:16.13 | daork | im about to buy some.. are they good? |
03:16.18 | znoG | they look good at $75/$85 |
03:16.23 | znoG | i'd like to know if they're any good :) |
03:16.35 | UnixDawg | they are ok for starter phones |
03:16.35 | daork | znoG: more like 65ish |
03:16.49 | UnixDawg | but whenyou need more power you will have to get a better phone |
03:16.56 | znoG | i got an email from @grandstream.com that they are $75/$85 |
03:17.04 | daork | znoG: chagres.net |
03:17.08 | daork | and many other resellers |
03:17.10 | znoG | UnixDawg: what sort of things can't it do? it seems to do call conference, caller id, call transfer, etc |
03:18.02 | UnixDawg | not all the buttons work it seems the mute does not work send does not work the confrance button I have not got working |
03:18.09 | Stealth_Man | znog: i can sell you GS |
03:18.12 | UnixDawg | I got the msg button working |
03:18.14 | Stealth_Man | together with voice services /// |
03:18.15 | Stealth_Man | :) |
03:18.35 | znoG | Stealth_Man: $$ ? |
03:18.41 | znoG | daork: thanks for that, saved $10 already.. :) |
03:18.54 | znoG | UnixDawg: eek, thats not so good |
03:18.55 | UnixDawg | they are a good basic phone |
03:18.55 | UnixDawg | for learning with |
03:19.27 | znoG | well as long as caller id, call transfer, call hold, and call conference and hmmm speakerphone work.. im happy |
03:19.54 | UnixDawg | speaker phone works I have not tried confrance yet |
03:20.33 | *** join/#asterisk Gary (Gary@218.19.158.239) |
03:20.53 | Stealth_Man | speaker phone works ... |
03:21.26 | znoG | UnixDawg: please let me know how you go with th ephone.. very interested in the feedback |
03:21.43 | UnixDawg | ok will do |
03:21.55 | UnixDawg | <PROTECTED> |
03:21.56 | Stealth_Man | znoG: GS works good for sure ... a lot of people are using it .. |
03:22.06 | UnixDawg | I am writing my extensions.conf now |
03:22.22 | Stealth_Man | znog: you can take a look for some Sipura-2000 adapter maybe ... |
03:25.08 | Stealth_Man | Capabilities: us - 524302, them - 281/0, combined - 8 |
03:25.08 | Stealth_Man | Non-codec capabilities: us - 1, them - 0, combined - 0 |
03:25.08 | Stealth_Man | Looking for 7001 in sip |
03:25.08 | Stealth_Man | Transmitting (NAT): |
03:25.08 | Stealth_Man | SIP/2.0 404 Not Found |
03:25.17 | Stealth_Man | why is that ? |
03:26.02 | Adam_ | cause the name of the ata isn't really 7001 |
03:26.28 | znoG | are there any large sites with GS deployed on every desk? |
03:26.35 | Stealth_Man | Adam ??? |
03:27.00 | Stealth_Man | znoG: possible ... but large sites are usually with Cisco phones ... |
03:27.19 | Stealth_Man | Adam: what do you mean by ATA name not 7001 ??? |
03:27.28 | znoG | ah okay |
03:27.36 | znoG | Stealth_Man: how much is the Sipura adapater? |
03:27.44 | Stealth_Man | 110$ |
03:27.47 | znoG | ouchie |
03:27.49 | Stealth_Man | dual port |
03:27.51 | znoG | i'll stick to the GS :) |
03:28.09 | Stealth_Man | 2 FXS port ... |
03:28.22 | Stealth_Man | but you need actual phones to get connecteed |
03:28.30 | Corydon76-home | Wow, anybody else notice how _polite_ the Asterisk lists are as of late? |
03:28.51 | Stealth_Man | Coryd: yepp ... just very polite and nice talking .. hah ?:)) |
03:29.04 | Corydon76-home | It's the very model of civility... |
03:30.41 | Stealth_Man | Adam: what you said about name of ATA ? |
03:31.24 | znoG | Stealth_Man: so all the features of the GS as advertised should work, right? |
03:31.49 | Stealth_Man | znog: HARDWARE PHONE !!! |
03:31.50 | znoG | it'll probably become troublesome to have my phone on my laptop :) |
03:31.52 | Stealth_Man | it is much better |
03:32.03 | znoG | reliability wise or because its a pain in the ass? |
03:32.11 | Stealth_Man | znog: I was travelling with Cisco ATA no prob:)) |
03:32.17 | znoG | nice :) |
03:32.23 | Stealth_Man | Znog: for this money you cna try it :) |
03:32.29 | znoG | how much are the ATA's worth anyway? |
03:32.30 | Stealth_Man | many many people are using it |
03:32.36 | znoG | using what? the Cisco ATA? |
03:32.41 | Stealth_Man | znog : around 130-150 depends |
03:32.42 | znoG | oh the GS |
03:32.54 | Stealth_Man | GS are chepest phones |
03:33.00 | znoG | yeah i'm gonna buy one to test anyway.. they look good |
03:33.09 | Stealth_Man | so you meed to make decision ... to spent 100$ let's and to check it ... |
03:33.11 | znoG | have you tried any phones from www.sipphone.com ? |
03:33.19 | Stealth_Man | znog: same GS phones |
03:33.23 | Stealth_Man | as you see on picture |
03:33.23 | znoG | hehe okay |
03:33.32 | *** join/#asterisk Mike-69 (~mike@dsl-200-67-40-148.prodigy.net.mx) |
03:34.17 | Stealth_Man | znog: get phone, I will get you some free Long distance minutes with it ;-) |
03:34.50 | znoG | uhhh :) |
03:35.01 | znoG | here's a stupid question - why would you need dual ethernet on a phone? |
03:35.07 | Stealth_Man | simple |
03:35.16 | Stealth_Man | GS has 2 versions of phones ... |
03:35.26 | Stealth_Man | 1 with 1 eth and another with 2 ethr ports |
03:35.35 | Stealth_Man | znog : have you seen modem any time ? |
03:35.41 | Stealth_Man | internal modem ... |
03:35.43 | Stealth_Man | for PC |
03:35.48 | Stealth_Man | not for notebook ... |
03:35.59 | Stealth_Man | it has 2 ports too rigth ? |
03:36.08 | marrandy | Stealth_Man: do you have it registering yet ? |
03:36.31 | Stealth_Man | marrandy: i am wlroking on config file .. they are registered , i have jsut mess in extensions.conf |
03:36.37 | znoG | Stealth_Man: yeah but whats the use of the 2nd ethernet? |
03:36.40 | UnixDawg | chan_aix right |
03:36.46 | Stealth_Man | znog : same as a telephone port on modem ... |
03:36.48 | marrandy | computer |
03:36.53 | znoG | oh ok gotcha |
03:37.04 | marrandy | one for the sip phone, the other for the computer |
03:37.14 | Stealth_Man | :) |
03:37.25 | znoG | http://www.grandstream.com/images/BudgeTone.jpg << they are all budgetone 100 phones but they look different and some have more buttons than others - whats the diff? |
03:37.28 | Stealth_Man | LAN connection goes into phone and from phone into PC ... |
03:37.34 | znoG | yep i see |
03:37.43 | Stealth_Man | znog : you need simple phone .. common stop thinking |
03:37.50 | Stealth_Man | BT-100 one thernet |
03:38.06 | UnixDawg | ok iax is working |
03:38.36 | marrandy | Stealth_Man: so what is your sip.conf file as you have a static address. |
03:39.12 | UnixDawg | in modules.con how do you turn off the iax codec |
03:39.15 | Stealth_Man | marrandy:check email ... |
03:39.26 | UnixDawg | noload=chan_iax right |
03:39.55 | znoG | Stealth_Man: but does the BT-100 have the hold/transfer/conference features? the second phone in BudgeTone.jpg has the mute feature too. I want that one.. :) |
03:40.20 | UnixDawg | ok got it |
03:40.32 | UnixDawg | man once you learn it comes quick |
03:41.41 | Stealth_Man | znog: ye it does ... buttons :) |
03:43.55 | znoG | Stealth_Man: last question, do you know why only the right 2 phones have the mute button in that BudgeTone.jpg pic? they are all BT-100's |
03:44.05 | UnixDawg | well I am almost done with extensions |
03:44.06 | UnixDawg | .conf |
03:44.08 | UnixDawg | then to reload and test |
03:44.12 | UnixDawg | hell all my outbound are gone |
03:44.17 | UnixDawg | grr ok this will be fun |
03:44.27 | znoG | good luck UnixDawg .. let us know how you go :) |
03:46.34 | UnixDawg | ok |
03:47.13 | Stealth_Man | unixdawg: yep ... let's make it working ... |
03:48.12 | *** join/#asterisk yifang (~yifang@ip68-9-77-241.ri.ri.cox.net) |
03:48.27 | *** join/#asterisk Rave (~chatzilla@65.123.139.62) |
03:48.29 | znoG | so if you want a hardcore phone, you get a SNOM 105 or a Cisco.. right? |
03:48.29 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
03:48.43 | *** join/#asterisk battlej (~battlej@66-65-184-236.nyc.rr.com) |
03:49.17 | Stealth_Man | shit |
03:49.24 | Stealth_Man | znog: GS too :) |
03:49.25 | Stealth_Man | Capabilities: us - 524302, them - 281/0, combined - 8 |
03:49.25 | Stealth_Man | Non-codec capabilities: us - 1, them - 0, combined - 0 |
03:49.25 | Stealth_Man | Looking for 7001 in sip |
03:49.25 | Stealth_Man | Transmitting (NAT): |
03:49.25 | Stealth_Man | SIP/2.0 404 Not Found |
03:49.27 | Stealth_Man | Via: SIP/2.0/UDP 192.168.1.103;received=192.168.1.103 |
03:49.29 | Stealth_Man | From: <sip:7000@192.168.1.99>;tag=e9510380-b785-aab9-79b9-608cce612bf1 |
03:49.34 | Stealth_Man | anyone nay ideas why am i getting this ? |
03:52.43 | *** join/#asterisk frollo (~frollo@d81-211-254-242.cust.tele2.it) |
03:52.53 | frollo | Hi guys |
03:53.12 | frollo | Someone of you is testing the x.lite phone? |
03:54.02 | frollo | ok no one :) |
03:54.40 | battlej | ok, frollo, I'll bite. I've played with it some. |
03:54.50 | znoG | i have too |
03:55.08 | frollo | ok, did you experienced some issues in sending the mic audio? |
03:55.37 | battlej | I've used it on pocketpc and mac platforms, and found it... lacking. |
03:55.49 | frollo | everything is working perfect the only issue is sending mic audio on SIP |
03:56.01 | frollo | ok |
03:56.01 | battlej | I'm guessing they codecs haven't really been optimized for those platforms... |
03:56.21 | battlej | but it's just a guess. and I'm not using particularly intense codecs... |
03:56.33 | znoG | frollo: i had that problem, and i did disallow=all, allow=gsm .. and it worked |
03:56.34 | frollo | i tested it on windows and is fine even if i call out trough the fxo of asterisk |
03:57.01 | frollo | i'll try just now :) |
03:57.04 | frollo | let's see |
04:01.57 | frollo | znoG: I LOVE YOU :) IT WORKS!!!!!! |
04:02.09 | Stealth_Man | heh |
04:02.19 | battlej | excellent! |
04:02.31 | znoG | :-) |
04:03.24 | frollo | i was looking for a day in the internet in google asterisk and so on but everyone was saying about the default interface :/ |
04:03.57 | frollo | the issue was not there, it is strange that there is, no info, nowhere :/ |
04:04.25 | znoG | yeah, it is strange that the default codecs don't work |
04:04.44 | frollo | gsm is better i think |
04:05.06 | frollo | i heard that g711u takes 20k more bandwidth |
04:05.29 | Stealth_Man | frollo: g711 compare to Gsm ... is big different |
04:05.31 | wasim | frollo: 20k more than what? |
04:05.38 | Stealth_Man | wasim: good morning |
04:05.39 | Stealth_Man | :)) |
04:05.41 | frollo | more than gsm |
04:05.44 | Stealth_Man | you are not sleeping ? |
04:05.52 | wasim | morning Stealth_Man, just wakey wakey |
04:05.52 | Stealth_Man | g711 =64kbps + overhead |
04:06.11 | Stealth_Man | wasim: i have impression that you are not sleeping at all and only working working worling ... |
04:06.29 | frollo | what's the time there? here is 5 am an in a few our i should go to work without sleeping :/ |
04:06.46 | wasim | its just after 9 am |
04:07.12 | Mike | Stealth_Man: why is it a big difference? |
04:07.30 | frollo | yesterday night I was fighting with a SAP DB upgrade at work and I had to rollback the upgrade (shitty NT4) |
04:07.59 | Stealth_Man | frollo: hehe |
04:08.00 | Stealth_Man | :) |
04:08.45 | frollo | anyway now asterisk is making me happy now :) Is possible conferencing on asterisk only using SIP phones ? |
04:09.19 | Stealth_Man | frollo : why only sip phones ... |
04:09.42 | frollo | I mean is it possible to think an extension as a meeting room for SIP phones |
04:10.02 | frollo | ? |
04:10.08 | Mike | Stealth_Man: are you ignoring me? |
04:10.18 | Stealth_Man | Mike ? |
04:10.26 | Mike | <Mike> Stealth_Man: why is it a big difference? |
04:11.29 | Stealth_Man | Mike: why not ?! G711 is using 64 kbps |
04:11.39 | Stealth_Man | more then twice as a gsm ... |
04:12.20 | Stealth_Man | Mike: your GS phone |
04:12.24 | Stealth_Man | is it working ? |
04:12.25 | *** join/#asterisk sack (~sack@polar.es3.egwn.net) |
04:13.14 | Stealth_Man | Mike: are you igniring me ? |
04:14.04 | *** join/#asterisk idiot (~asterisk@ip-wv-68-119-143-145.charterwv.net) |
04:14.42 | idiot | ok.. as my nick says, i'm an idiot :) having two problems, both would be small if i weren't so stupid :) |
04:14.53 | znoG | you idiot, go ahead |
04:14.54 | Mike | Stealth_Man: sorry phone |
04:15.06 | Stealth_Man | Mike: :)) no problem |
04:15.09 | Mike | Stealth_Man: yes my GS phone is working |
04:15.12 | idiot | Nov 17 23:03:49 NOTICE[139478016]: File chan_sip.c, Line 5074 (handle_request): Registration from '<sip:carolyn@10.0.0.1>' failed for '10.0.0.2' |
04:15.14 | Mike | Stealth_Man: why? |
04:15.27 | idiot | znoG was typing :) |
04:15.43 | Stealth_Man | Mike:you have one GS behind NAT correct ? |
04:15.53 | Mike | well my asterisk server |
04:15.55 | Stealth_Man | can you put here your sip.conf and extenssions.conf ? |
04:15.56 | idiot | anyone got an example for what to put in sip.conf for incoming connections? :) |
04:15.58 | Mike | has the pppoe conextion |
04:16.02 | *** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) |
04:16.06 | Mike | i conect my GS directly to the internal ip address |
04:16.25 | Stealth_Man | Mike: ok |
04:16.31 | Stealth_Man | post your config please ... |
04:16.32 | idiot | any way to get better error message as for what is causing failure? |
04:16.40 | Stealth_Man | for sip.conf part and extensions.conf |
04:16.45 | frollo | idiot: http://www.onlamp.com/lpt/a/3956 |
04:16.49 | Mike | Stealth_Man: i got the on the web |
04:16.57 | Stealth_Man | ok what os the address ? |
04:17.21 | idiot | fwiw, carolyn is just a softphone (sjphone) on a laptop |
04:18.12 | UnixDawg | ok who has the libpri patch for fbsd |
04:18.16 | UnixDawg | I need it |
04:19.26 | znoG | so, with a mic/speakers headset and X-Lite, is it really that bad to use X-Lite for everyday phone use? |
04:19.32 | znoG | if so, whats so bad about it? |
04:19.42 | idiot | UnixDawg all you have to do is comment out the ldconfig invokation in makefile, and replace references to endian.h with machine/endian.h |
04:19.46 | znoG | (replace x-lite with <any software phone>) |
04:19.51 | idiot | znoG I use it daily, works great |
04:20.00 | znoG | yeah, but you're an idiot |
04:20.03 | znoG | only joking :) |
04:20.07 | frollo | znoG: x-lite is good and it's free :) |
04:20.32 | znoG | i'm getting ISDN put in soon.. just wondering if we can survive with X-Lite as phones for a while until we get some grandstream phones in |
04:20.36 | UnixDawg | ok |
04:20.37 | frollo | sjphone does not look nice |
04:21.13 | *** join/#asterisk _gorman (~lehmann@pD9E4E223.dip.t-dialin.net) |
04:21.17 | frollo | why buy expensive phones when everyone has a pc :) |
04:21.31 | Mike | frollo: because... |
04:21.45 | Mike | frollo: pcs have bad sound cards or different setting for sound cards |
04:21.51 | Mike | frollo: eco with speakers and mic |
04:22.05 | znoG | frollo: i'll be using X-Lite for everything, not just calling PCs |
04:22.05 | Mike | frollo: you need a compiuter on 24/7 to be accesible |
04:22.06 | frollo | ok the old foggy office assistant does not know PCs :) |
04:22.21 | Mike | frollo: and you dont always listen when people calls you |
04:22.36 | Mike | frollo: using a hardware phone is a standalone system no need for a computer |
04:22.44 | Mike | frollo: hardware works 100% and really good |
04:23.04 | Mike | frollo: you just need a small router in most cases that you wont even notice if its on all day and all night |
04:23.05 | frollo | yah it's true :) but may be because i'm used to work in a call center :) |
04:23.12 | Mike | frollo: your accessible 24/7 with your phone |
04:24.10 | frollo | we don't have anymore phones :) only VoIP and software everywhere no hardphones :) |
04:24.21 | znoG | frollo: works well? |
04:24.40 | idiot | brb, hacking on this |
04:24.44 | UnixDawg | make: don't know how to make pri.lo. Stop |
04:24.57 | Mike | frollo: that mean you have to be infront your computer to recibe a call |
04:25.12 | Mike | frollo: if your not infront it?? if your sleeping or if your eating |
04:25.13 | znoG | so? you have to be in front of the phone to recieve a call :) |
04:25.14 | frollo | well of course but in the call center we have AVAYA Definity = million dollars ACD cluster :) |
04:25.16 | UnixDawg | night boys |
04:25.21 | znoG | if im sleeping or eating, i dont want to take any calls |
04:25.43 | frollo | ahahaha znoG :) |
04:25.56 | frollo | Mike the speakers!!!!! |
04:26.14 | UnixDawg | ok got it |
04:26.15 | frollo | Mike the speakers!!!!! They ring when telephone rings!!! |
04:26.17 | UnixDawg | cls |
04:26.30 | znoG | problem is i'll have to unplug my headset so i can hear the calls |
04:26.38 | Mike | frollo: eco problem |
04:26.45 | znoG | Mike: not if you use a headset |
04:26.53 | Mike | znoG: hes talking about speakers |
04:27.00 | Mike | znoG: and most headset still makes eco |
04:27.25 | znoG | i was talking with my headset here and it didnt have the echo problem |
04:27.27 | Mike | znoG: good headsets like futball headset motorola are about 200usd |
04:27.36 | Mike | znoG: buy a 65dlls hardware phone that it |
04:28.42 | frollo | ok Mike you are right |
04:29.01 | frollo | you convinced me and mi bos too about money :) |
04:29.06 | frollo | boss |
04:30.04 | idiot | still not working, znoG |
04:30.41 | idiot | REGISTER sip:10.0.0.1 SIP/2.0 |
04:30.55 | idiot | From: <sip:1000@10.0.0.1>;tag=36940285 |
04:30.59 | idiot | that looks wrong for some reason? |
04:31.16 | idiot | asterisk box is 10.0.0.1, carolyn is 10.0.0.2 presently |
04:40.04 | frollo | :) Conferences also work using SIP Phones :) I love ASTERISK :) |
04:40.11 | *** join/#asterisk chayewala (chayewala@adsl-66-122-206-141.dsl.sntc01.pacbell.net) |
04:40.17 | chayewala | hello |
04:40.56 | frollo | ok guys. thanks for your precious help. i'll have to go to sleep a few hours before going back to work. Ciao! |
04:41.52 | chayewala | Hello Jerjer |
04:42.11 | chayewala | is stealthman there. |
04:42.31 | chayewala | I finally was able to setup my grandstream phone. |
04:43.20 | znoG | Mike: yeah, we're probably going to end up with a hardware phone |
04:43.26 | chayewala | he he h |
04:43.44 | chayewala | there are few nice one now within range of ordinary folks like us. |
04:44.11 | znoG | yeah, the grandstreams look ok |
04:44.28 | JerJer | barbietone |
04:44.37 | Mike | znoG: 65dlls for a 100% compatible phone |
04:44.41 | Mike | znoG: its a good deal |
04:47.15 | chayewala | he he he |
04:47.18 | chayewala | Jerjer hello |
04:47.48 | bkw_ | 7960s ROCK! |
04:48.33 | chayewala | Jerjer - can I make multiple calls from chan_H323 code. I am haveing a lot of success with your code. |
04:48.54 | bkw_ | chayewala yes you sure can |
04:49.14 | chayewala | Jerjer: What will it take to make your chan_h323 code a gateway instead of an endpoint only. |
04:49.21 | chayewala | nkw: Have you tried it? |
04:49.28 | chayewala | bkw |
04:49.30 | bkw_ | kill h323 NOW |
04:49.33 | bkw_ | dont use it |
04:49.36 | chayewala | why? |
04:49.47 | chayewala | I am routing all h323 calls to * |
04:49.57 | chayewala | a lot of gateways only use H323 |
04:50.05 | chayewala | so be considerate. |
04:50.08 | chayewala | he hehe |
04:50.21 | bkw_ | inbound to * via h323 |
04:50.25 | Farooq | yes |
04:50.38 | Farooq | those calls we get on H323, I rout them to *. |
04:50.44 | Farooq | and then viola |
04:50.48 | Farooq | voila |
04:50.55 | bkw_ | viola? |
04:50.56 | bkw_ | haha |
04:50.56 | Farooq | Jerjer's code is good |
04:51.03 | Farooq | haha |
04:51.06 | bkw_ | good... DAMN GOOD! |
04:51.23 | Farooq | yes but why is it an end_point only. |
04:51.34 | bkw_ | its not.. it will receive and make calls |
04:51.39 | Farooq | yes |
04:51.45 | bkw_ | you just set a context and it works |
04:51.45 | Farooq | will it make multiple calls ? |
04:52.01 | Farooq | how does it know how many channels. |
04:52.08 | Farooq | can I limit channels? |
04:52.19 | Farooq | jerjer? |
04:52.28 | bkw_ | no |
04:52.39 | Farooq | no what? |
04:52.47 | Farooq | will not make multiple calls to and form h323? |
04:52.53 | bkw_ | I don't think you can limit |
04:52.56 | Farooq | ok |
04:52.58 | bkw_ | you can make as many as the box will take |
04:53.06 | Farooq | ok |
04:53.10 | Farooq | and another question. |
04:53.29 | Farooq | I am trying to setup a simple generic office setup. |
04:53.32 | Farooq | with voice mail |
04:53.44 | bkw_ | pay me and I will h00k you up |
04:53.49 | Farooq | like respond with a greeting and etc etc |
04:53.51 | Farooq | he hehe |
04:53.54 | Farooq | with what? |
04:54.02 | Farooq | you mean with jerjer? |
04:54.18 | Farooq | alas! jerjer is a guy and me too :-) |
04:54.18 | Farooq | hehehe |
04:55.00 | Farooq | can someone provide me some good voice mail + generic office setup |
04:55.21 | Farooq | I mean configuratin files |
04:55.31 | Mike | bkw_: for some reason asterisk and my gs doesnt like |
04:55.35 | Mike | g723 |
04:55.44 | Mike | g729 |
04:55.45 | Mike | i mean |
04:55.49 | Farooq | yeah |
04:56.04 | Farooq | can someone develope g723.1 support using Intell IPP. |
04:56.11 | bkw_ | nope |
04:56.14 | bkw_ | still have to lic. it |
04:56.15 | Farooq | it is freely available for development |
04:56.24 | Farooq | why not? |
04:56.38 | Farooq | then who ever wants to use it can pay for it :-) |
04:57.04 | idiot | hmm |
04:58.10 | Farooq | I downloaded IPP and was able to compile the code and save voice files in g723.1 |
04:58.27 | Farooq | but I dont know how to add that support to Asterisk |
05:00.05 | bkw_ | give up now.. you will have to buy the rights to use it |
05:00.26 | Mike | g729? |
05:00.27 | Farooq | yes but not to develop it - right? |
05:00.39 | Farooq | g723.1 - i am talking about. |
05:01.22 | Farooq | Can any one assign 1700 and 1777 numbers or does it require some sort of permission ? |
05:04.28 | atacomm | bkw: when i enable a macro that is supposed to fix some caller id quirks (i found it else where), Asterisk hangs up the call.....can you look it over quickly? |
05:04.30 | idiot | anyone got * working w/ fwd? |
05:05.56 | atacomm | Farooq: everyone that does it does it internally only. The PSTN does not recognize them |
05:08.34 | h3x | 700 is reserved for the IXC's use. typically 700-555-4141 tells you which ld carrier you have selected as your PIC |
05:09.08 | *** join/#asterisk maharzan (~arun@202.51.76.140) |
05:09.18 | maharzan | hi |
05:09.24 | maharzan | i have a small problem |
05:09.37 | maharzan | i am currently using netgear router |
05:09.45 | h3x | 777 is assigned by nanpa as "easily recognizable code" |
05:09.51 | maharzan | which uses public IP |
05:10.12 | maharzan | in my network i.e 192.168.0.x , i have set up my * |
05:10.19 | *** join/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net) |
05:10.30 | maharzan | i cannot call to US using IP with that configuration? |
05:10.37 | maharzan | is it something to do with NAT? |
05:10.46 | maharzan | i think netgear uses NAT |
05:11.09 | idiot | using SIP? |
05:11.16 | maharzan | IAX |
05:11.26 | maharzan | using voicepulse account |
05:11.49 | idiot | i dunno about iax but for sip you must forward your RTP ports |
05:12.05 | maharzan | ok |
05:12.11 | maharzan | how can this be done? |
05:12.23 | maharzan | it should be similar |
05:12.27 | *** join/#asterisk lexluther (~admin@80.88.142.221) |
05:12.55 | *** part/#asterisk lexluther (~admin@80.88.142.221) |
05:13.21 | *** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net) |
05:14.14 | mishehu | a ratings engine? |
05:14.18 | bkw_ | rating |
05:14.24 | mishehu | rating what? |
05:14.32 | bkw_ | calls |
05:14.45 | bkw_ | i'm sure as hell not talking about movies! |
05:14.50 | idiot | maharzan check your rtp.conf |
05:14.57 | Farooq | thanks atacomm |
05:15.04 | idiot | and forward that port range to the asterisk box's ip |
05:15.05 | Farooq | you are very knowledgable |
05:15.12 | mishehu | bkw_: are you recording the conversations to rate them? ;-) |
05:15.20 | Farooq | how about 1 777 numbers? |
05:15.27 | idiot | Anyone using free world dialup with *? :) |
05:15.34 | Farooq | he he he |
05:15.37 | Connor | were do you get the channel h323 code at? |
05:15.49 | Farooq | it is part of asterisk |
05:15.52 | bkw_ | mishehu um no |
05:15.56 | Farooq | you just need to compile it |
05:16.02 | Farooq | connor: |
05:16.24 | bkw_ | JerJer gots a question |
05:16.26 | Farooq | for that you need to get pwlib and h323 from www.openh323.org |
05:17.04 | JerJer | yep |
05:17.08 | Farooq | instructions are in a file in channels directory about h323 |
05:17.20 | bkw_ | haha |
05:17.36 | Connor | How well does it work with Cisco Call Manager? |
05:17.45 | Farooq | bkw : have you tried multiple h323 channels with * |
05:17.46 | Farooq | ? |
05:17.52 | bkw_ | Farooq daily |
05:17.58 | Farooq | ok |
05:18.22 | Farooq | can you provide your configuration files? You can remove passwords etc :-) |
05:18.33 | Farooq | I can only use one channel. |
05:19.02 | Farooq | what type of work do you do bkw? |
05:20.02 | Farooq | atacom? |
05:20.18 | decode | bkw is an evil hacker |
05:20.45 | decode | bkw_ isn't that right? |
05:20.52 | bkw_ | um no |
05:20.54 | bkw_ | [general] |
05:20.54 | bkw_ | port = 1720 |
05:20.54 | bkw_ | bindaddr = 0.0.0.0 |
05:20.54 | bkw_ | tos=lowdelay |
05:20.54 | bkw_ | allow=all |
05:20.55 | bkw_ | noSilenceSuppression = yes |
05:20.57 | bkw_ | dtmfmode=inband |
05:20.59 | bkw_ | context=autoattend |
05:21.01 | bkw_ | that my h323.conf |
05:21.03 | bkw_ | PURE AND FUCKING SIMPLE |
05:23.04 | decode | bkw_ sire, what is 'black friday' prices? |
05:23.11 | decode | i like the 5$ ups |
05:23.24 | Farooq | well how do you transfer the calls to sip devices? |
05:23.26 | *** part/#asterisk lucifuge3 (trilluser@pcp05047461pcs.ivylnd01.pa.comcast.net) |
05:23.31 | Farooq | BKW? |
05:23.38 | bkw_ | decode its what bestbuy used the DMCA to quash from fatwallet.com |
05:23.41 | decode | Farooq Call(SIP/port) |
05:23.49 | decode | bkw_ Huh? |
05:23.53 | decode | was that bs prices? |
05:23.54 | bkw_ | I would like to see them try to make me take them down.. because they better have a gun to my head and kill be first. |
05:23.57 | decode | 5$ for the ups? |
05:24.01 | bkw_ | haha |
05:24.09 | decode | i am confused |
05:24.15 | bkw_ | not sure.. I just posted the full list from the slashdot links |
05:25.50 | decode | ahh |
05:25.58 | decode | was wondering if i have to take 500$ to bestbuy ;) |
05:27.44 | bkw_ | decode i'm just being evil.. |
05:27.55 | decode | bkw heh |
05:28.07 | Connor | where can I find the versons of Open h323 and pwlib that's needed for h323 ? |
05:28.08 | decode | bkw happen to use free world dialup w/ *? :) |
05:28.41 | bkw_ | decode mine works |
05:28.43 | bkw_ | why? |
05:31.48 | bkw_ | now what was my fwd number |
05:32.01 | bkw_ | FWDUSERID=51991 |
05:33.04 | decode | because i cant get it to work on incoming calls? :) |
05:33.46 | bkw_ | behind nat? |
05:33.52 | decode | yea, sadly |
05:34.00 | bkw_ | good fuckin luck! god be with ya |
05:34.04 | decode | can punch holes as needed |
05:34.05 | atacomm | lol |
05:34.17 | bkw_ | atacomm you got 500 bucks |
05:34.32 | atacomm | ? i had to borrow money today so i could go park, why? |
05:34.40 | bkw_ | haha |
05:34.42 | bkw_ | just askin |
05:35.36 | decode | heh.. i think i'm going to be unemployed soon.. |
05:35.44 | decode | one of my gf's came to where i work |
05:35.58 | decode | and my boss made the mistake of calling her a "tasty treat" so i sliced his tires.. heh.. heh.. |
05:38.01 | *** join/#asterisk maharzan (~arun@202.51.76.140) |
05:40.35 | tessier__ | decode: Glad you don't work for me. You're a nut. |
05:40.49 | tessier__ | "one" of your gf's? |
05:41.01 | decode | tessier__ my gf has a gf :) |
05:41.08 | tessier__ | Sure. |
05:41.09 | decode | and therefore her gf is my gf too |
05:43.00 | Mike | g729 doesnt work!! |
05:43.02 | Mike | grrr |
05:43.14 | Mike | can dial 1800 when i call it rings when they pick up the call gets hang up |
05:43.17 | Mike | what can i do? |
05:45.00 | *** join/#asterisk Pids (~pids@adsl-67-121-190-78.dsl.sntc01.pacbell.net) |
05:45.13 | *** join/#asterisk clive- (~pirch@rndf-ip-nas-2-p217.telkom-ipnet.co.za) |
05:49.45 | *** join/#asterisk danielq (~danielq@nat154.gw1.bne.webcentral.com.au) |
05:52.56 | danielq | anyone awake? |
05:53.32 | Farooq | pin drop silence |
05:53.44 | Farooq | sprint rules :-) |
05:56.10 | *** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net) |
05:56.10 | *** topic/#asterisk is Asterisk: The Linux of Telephony |
05:56.14 | bkw_ | blah |
05:56.30 | wasim | blah timmah |
05:56.56 | wasim | what's a timmah, bkw_ ? |
05:57.38 | Connor | hey, bkw... In that new vocemail enhancment patch.. |
05:57.50 | danielq | having fun and games getting chan_oh323 to compile |
05:57.50 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
05:58.12 | danielq | anyone played with it lately? |
05:58.18 | sizzzung | yo momma |
05:58.23 | sizzzung | she likes my TESTICLES. |
05:58.26 | sizzzung | aklsjfklajdf |
05:58.29 | Connor | why the heck did they go overboard with the cmd = ast_waitfordigit(chan, 600) between each and every prompt ? |
05:58.32 | sizzzung | HLAGHLAGHLAG |
05:58.42 | ReG-Hexer | hi |
05:58.42 | bkw_ | who is they |
05:58.47 | ReG-Hexer | i have a question |
05:58.51 | Connor | Who ever coded the patch. :) |
05:58.54 | sizzzung | ReG-Hexer: no i am not gay |
05:58.57 | sizzzung | ReG-Hexer: but blll is |
06:09.54 | *** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net) |
06:09.54 | *** topic/#asterisk is Asterisk: The Linux of Telephony |
06:09.56 | cman | i cannot call with * to US using IAX voicepulse |
06:10.32 | cman | <PROTECTED> |
06:10.33 | cman | <PROTECTED> |
06:10.33 | cman | WARNING[7176]: File chan_iax2.c, Line 1124 (attempt_transmit): Max retries exceeded to host 66.234.228.132 on IAX2[voicepulse]/1 (type = 6, subclass = 1, ts=1, seqno=0) |
06:10.33 | cman | <PROTECTED> |
06:10.33 | cman | <PROTECTED> |
06:11.27 | cman | my config files are alrite |
06:12.27 | Farooq | what is your phone number cman? |
06:12.28 | cman | i am using netgear router which has an IP of 202.xx.xx.xxx |
06:12.47 | cman | i am in nepal.. |
06:12.58 | Farooq | OK |
06:13.02 | cman | you cannot call from there as i have no account |
06:13.05 | Farooq | what is your voicepulse number? |
06:13.10 | cman | i can only mak calle to US |
06:13.11 | cman | not incoming |
06:13.17 | Farooq | hmm |
06:13.27 | Farooq | what type of device do you have? |
06:13.31 | cman | i am trying to set up * to call to US |
06:13.33 | Farooq | SIP phone? |
06:13.39 | Farooq | OK |
06:13.49 | cman | i am using X101P and TDM400P |
06:13.54 | Farooq | ok |
06:13.57 | Farooq | wll |
06:14.01 | Farooq | well |
06:14.07 | cman | i have normal phone |
06:14.15 | Farooq | OK |
06:14.21 | Farooq | I dont really know then |
06:14.23 | cman | i can make internal calls, local calls but not to US |
15:31.11 | *** join/#asterisk jbot (ibot@c-24-1-99-18.client.comcast.net) |
15:31.11 | *** topic/#asterisk is Asterisk: The Linux of Telephony |
15:32.38 | opsys | Pj_ I ran gdb core file then bt and got 'no stack' Any Ideas?? |
15:33.20 | RoyK | Ares: apt-get install icecast-client |
15:33.25 | RoyK | ...or something |
15:33.50 | Ares | why install a new player |
15:33.56 | RoyK | perhaps |
15:34.00 | RoyK | *checking* |
15:34.16 | Ares | let check |
15:34.39 | Ares | but icecast should normally work well with shoutcast |
15:35.02 | *** join/#asterisk dutch_ (~dutch@a80-126-102-2.adsl.xs4all.nl) |
15:35.18 | Ares | Couldn't find package icecast-client |
15:35.26 | RoyK | don't mind |
15:35.37 | RoyK | it was to feed mpeg streams to an icecast server |
15:36.11 | *** part/#asterisk kapejod (~kapejod@pD9E81940.dip.t-dialin.net) |
15:36.28 | Ares | well, do you think possible to use an other player than mpg123 with asterisk |
15:36.43 | *** join/#asterisk Powerkill (~powerkill@195.68.105.195) |
15:37.06 | *** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) [NETSPLIT VICTIM] |
15:37.06 | *** mode/#asterisk [+bb *!*@og.latency.net *!*dan@194.158.*.*] by simak.freenode.net |
15:37.06 | *** mode/#asterisk [+q sant!*@*] by simak.freenode.net |
15:37.38 | Powerkill | RoyK are you here ? |
15:37.52 | *** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-131-106.NYCMNY83.dynamic.covad.net) |
15:37.52 | RoyK | Powerkill: sure :) |
15:38.03 | RoyK | Powerkill: I found the entries in a header on a redhat box |
15:38.10 | Powerkill | tiff.h:#define COMPRESSION_CCITT_T4 3 /* CCITT T.4 (TIFF 6 name) */ |
15:38.10 | Powerkill | tiff.h:#define COMPRESSION_CCITT_T6 4 /* CCITT T.6 (TIFF 6 name) */ |
15:38.14 | RoyK | so I just copied the entries over to my header file |
15:38.27 | Powerkill | ok so it's working now ? |
15:38.39 | RoyK | well - compiled, at least |
15:38.48 | RoyK | but I haven't sent any fax yet |
15:39.27 | *** join/#asterisk cc-reel (~cc-reel@hoochie.digium.com) |
15:40.57 | *** join/#asterisk sjoep803 (~sjoerd@213-84-218-42.adsl.xs4all.nl) |
15:41.58 | cc-reel | Has anyone had trouble with GrandStream Budge Tone 100 phones with the new versions of Asterisk? |
15:42.25 | bkw_ | cc-reel its the barbietones |
15:42.27 | bkw_ | its not * |
15:42.33 | bkw_ | barbietones SUCKK |
15:42.37 | bkw_ | with a double K |
15:42.38 | bkw_ | :P |
15:43.19 | cc-reel | What is the cheapest IP phones that you would give a good review? |
15:43.42 | cypromis | cheap + good review = non existant |
15:44.02 | cc-reel | hahaha .... Ok, So what phones do you like? |
15:44.16 | cypromis | cicsco 7960 |
15:44.16 | cypromis | or |
15:44.24 | cypromis | farfon when it will be ready |
15:44.52 | cc-reel | We have had good luck with the Snom 200's. |
15:45.42 | Powerkill | i try to fax from tx to rx and it's doesn't work |
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15:47.24 | RoyK | Ares: still here? |
15:47.37 | RoyK | bkw_: what's so bad about them? |
15:47.39 | Ares | yes |
15:48.06 | Ares | I was ready to hang my-self |
15:48.08 | RoyK | Ares: that jazz station works fine with MP3Player() |
15:48.41 | Ares | RoyK: that doesn't work |
15:48.57 | Ares | probably because that doesn't work with mpg123 |
15:49.28 | Corydon76-work | Hrm, nobody has commented on bug #543 yet... |
15:49.36 | Ares | actually, I m not greazy about this jazz station, but I would like to know which stream I can use and why ? |
15:49.46 | Pj_ | Ares: if you find another player which support changing output rate and forcing mono output |
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15:49.57 | *** join/#asterisk daork (~daork@202.89.35.252) [NETSPLIT VICTIM] |
15:49.57 | *** mode/#asterisk [+bb *!*@og.latency.net *!*dan@194.158.*.*] by asimov.freenode.net |
15:49.57 | *** mode/#asterisk [+q sant!*@*] by asimov.freenode.net |
15:50.17 | Ares | why my mother hit me ? and also if there is a solution to make working an other player ? |
15:50.25 | *** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) [NETSPLIT VICTIM] |
15:50.25 | *** join/#asterisk daork (~daork@202.89.35.252) [NETSPLIT VICTIM] |
15:50.25 | *** mode/#asterisk [+bb *!*@og.latency.net *!*dan@194.158.*.*] by asimov.freenode.net |
15:50.25 | *** mode/#asterisk [+q sant!*@*] by asimov.freenode.net |
15:50.28 | bkw_ | Corydon76 its a nice idea |
15:50.30 | tholo_ | Only mpg123 is supported by *. |
15:50.30 | Ares | Pj_ : I was looking for it |
15:50.30 | *** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net) |
15:50.30 | RoyK | hm |
15:50.38 | Ares | there is a version xmms without gui |
15:50.50 | Ares | but I did find way to get the output |
15:50.51 | Pj_ | Ares: I worked on supporting mpg321 but it doesn't support changing outptu rate |
15:50.52 | RoyK | Ares: mpg123 can download it, but not decode it afacs |
15:50.55 | tholo | But still only mpg123 is supported by *. |
15:51.33 | Pj_ | If you find a player that works better, I'll give you a patched version to use with |
15:52.23 | Pj_ | just mail me at pj@trouarat.com, with the player you've found... Or if you don't find any gimme urls that don't work with mpg123 and works with xmms and I'll investigate |
15:52.38 | bkw_ | um whats wrong with mpg123? |
15:52.41 | bkw_ | it works |
15:53.08 | Pj_ | bkw_: he says some streams don't work with it |
15:53.18 | Pj_ | and do work with xmms |
15:53.27 | Ares | Pj_: many thanks, I will try to investigate first and give you feedback soon |
15:53.36 | Pj_ | but maybe it's just a problem of URL malformation, that kinda things |
15:53.48 | Pj_ | Ares: I already have patched asterisk to support multiple players |
15:53.58 | Pj_ | just hadn't found any player interesting enough to add ;) |
15:54.32 | bkw_ | smokin crack trying to use a stream for MOH |
15:54.35 | RoyK | bkw_: it doesn't work with ogg |
15:54.41 | Ares | We will try to hack noxmms to get the output, i think that would be great |
15:54.43 | RoyK | bkw_: that's what's wrong with it |
15:54.44 | bkw_ | their is a patch on bugs.digium.com |
15:54.51 | bkw_ | oh ogg |
15:54.55 | *** join/#asterisk olivier_ (~olivier@APuteaux-105-2-1-45.w217-128.abo.wanadoo.fr) |
15:54.57 | bkw_ | why use something that isn't widely used? |
15:55.07 | RoyK | Ares: just use mpg123 to pipe the output to something else - like sox |
15:55.17 | Pj_ | bkw_: because it's freeeeeee |
15:55.19 | Pj_ | ;) |
15:55.24 | bkw_ | so is mpg123 |
15:55.30 | bkw_ | god you people make me laff |
15:55.34 | bkw_ | never happy |
15:55.38 | Pj_ | mpeg 2 layer 3 is a proprietary format |
15:55.42 | bkw_ | always gotta try to break something |
15:55.50 | bkw_ | its mpeg 1 layer 3 |
15:55.58 | Pj_ | still proprietary ;) |
15:56.05 | bkw_ | and its hardly far from proprietary |
15:56.18 | bkw_ | every os on the planet can pretty play them out of the box |
15:56.24 | Pj_ | (and I don't give a damn about mp3 being proprietary either :) |
15:56.24 | bkw_ | er /pretty much/ |
15:56.55 | bkw_ | Pj_ in your little world... you might think its proprietary.. but at some level EVERYTHING is |
15:57.02 | Pj_ | play them yeah, but not encode them, since you have to pay the Fraunhofer institute for hardware compression... something like this |
15:57.17 | bkw_ | you dont need to encode mp3's to fucking use hold music |
15:57.46 | Corydon76-work | bkw_: why don't you tell us how you really feel? |
15:57.50 | Pj_ | bkw_: calm down, I have _no_ ogg vorbis on my hd |
15:58.01 | bkw_ | Corydon76 I might piss people off |
15:58.09 | Pj_ | It's Ares who wanted to get an ofgg vorbis stream for his moh ;) |
15:58.17 | bkw_ | oh fuck it.. you're all idiots |
15:58.21 | bkw_ | :P |
15:58.23 | RoyK | http://www.apple.com/imac/ <-- 20" new iMac |
15:58.37 | Pj_ | GPL idiots, please |
15:58.56 | Pj_ | or BSD idiots, as you like |
15:59.05 | Corydon76-work | Oooh, that'd be nice to replace my 15" iMac... |
15:59.29 | bkw_ | corydon you can sed that 15 inch imac my way |
15:59.53 | bkw_ | me luv you long time^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H^H |
16:00.07 | *** join/#asterisk benjk_ (~Benjamin@f8a01-0359.din.or.jp) [NETSPLIT VICTIM] |
16:00.19 | Ares | I don't understand what is wrong to try to get asterisk working better |
16:00.36 | *** join/#asterisk anoosh (~anoosh@217.219.56.183) |
16:00.45 | bkw_ | whats better about ogg? |
16:00.48 | Corydon76-work | Ares: if it ain't broke... |
16:00.54 | bkw_ | we sould just convert our moh to gsm |
16:01.02 | bkw_ | and then we wouldn't need mpg123 |
16:01.52 | Ares | Corydon76-work: we will try to fix till we get it |
16:02.04 | anoosh | ok, sorry to jump in the middle of the discussion which i haven't read from the beginning ... but |
16:02.14 | anoosh | why not ogg? |
16:02.37 | bkw_ | ogg isn't standard |
16:02.39 | bkw_ | its open |
16:02.42 | bkw_ | but its far from a standard |
16:03.01 | anoosh | bkw said whats better with ogg, well, not having a license problem? |
16:03.23 | *** join/#asterisk super_st (~super_ste@hoochie.digium.com) |
16:04.19 | anoosh | ogg's not a standard? |
16:05.10 | *** join/#asterisk Art3d (~Art3d@vickesh01-6569.tbaytel.net) |
16:05.16 | bkw_ | Stealth_Man STOP IT |
16:05.33 | *** join/#asterisk bkw_ (~brian@ns.bkw.org) |
16:05.51 | dheckaman | ?????? |
16:06.06 | anoosh | bkw, you were saying ... (about ogg not being a standard) |
16:06.12 | Ares | Pj_: which id have the patch for support the multi player ? |
16:06.27 | *** join/#asterisk mortck (~test@p213.54.146.115.tisdip.tiscali.de) |
16:06.53 | bkw_ | anoosh its not a standard.. just like iax isn't |
16:07.04 | bkw_ | its open but far from a standard |
16:07.23 | *** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com) |
16:07.31 | Corydon76-work | bkw_, I think you're confused on exactly what makes a standard |
16:08.01 | Corydon76-work | Given that more than one entity agrees on the format and encoding, it's a standard |
16:08.08 | bkw_ | no it isn't |
16:08.13 | bkw_ | in that case we have ALOT of standards |
16:08.16 | Corydon76-work | sure it is |
16:08.19 | Corydon76-work | Yes, we do |
16:08.33 | bkw_ | just because two geeks agree to encode or decode with X format doesn't make it a standard |
16:08.36 | bkw_ | IAX isn't a standard |
16:08.40 | Corydon76-work | Sure it is |
16:08.55 | bkw_ | not it isn't |
16:09.05 | Exomorph | Morning Peoples |
16:09.14 | Corydon76-work | You think a standards organization has to approve it before it becomes a standard? Hogwash |
16:09.19 | bkw_ | nope |
16:09.28 | anoosh | well it really doesn't matter if it's a standard or not, it works and you can convert audio from mp3 format to ogg |
16:09.29 | Ares | Pj_: still here ??? |
16:09.34 | Corydon76-work | Standards organizations hammer out differing interpretations |
16:09.38 | anoosh | therefore |
16:09.55 | anoosh | anybody that wants to use asterisk could do just that, and get rid of mp3 license problems |
16:09.55 | Corydon76-work | Actually, while you can convert audio from mp3 to ogg, it's not recommended |
16:10.17 | bkw_ | Corydon just because alot of people use something.. doesn't make it standard.. wide acceptance does. |
16:10.23 | bkw_ | and ogg and iax are far from that point. |
16:10.45 | Corydon76-work | So you're saying that because the world doesn't accept T1, it's not a standard? |
16:11.01 | bkw_ | T1 is a standard |
16:11.16 | anoosh | bkw_: do you think it matters to use a widely accepted standard in asterisk's case? newbie computer users are NOT the people that are using asterisk |
16:11.16 | bkw_ | its widly acceptted in many areas |
16:11.17 | Exomorph | He didn't say the whole world accepts something... He said wide acceptance. |
16:11.22 | Exomorph | Two diffrent things. |
16:11.25 | Corydon76-work | But T1 doesn't have world-wide acceptance, so in your definition, it's not a standard |
16:11.35 | Corydon76-work | How wide is wide, then? |
16:11.47 | bkw_ | no I did'nt FUCKING SAY WORLD WIDE ACCEPTANCE |
16:12.21 | bkw_ | I said WIDE ACCEPTANCE |
16:12.29 | bkw_ | you people dont listen to me |
16:12.30 | Corydon76-work | And how wide is that? |
16:12.37 | Corydon76-work | IAX has wide acceptance |
16:12.39 | bkw_ | more than a handful of people |
16:12.44 | Corydon76-work | Ogg has wide acceptance. |
16:12.51 | bkw_ | still not wide enuf to take off and be viable yet |
16:13.05 | bkw_ | once we see hardware IAX phones.. and devices.. it will be a step closer |
16:13.10 | Corydon76-work | Whoa, you said standard, not viability |
16:13.21 | dheckaman | eek |
16:13.24 | bkw_ | viability as a standard |
16:13.45 | Corydon76-work | It's already viable as a standard |
16:14.07 | *** join/#asterisk doughecka (~rooot@adsl-68-133.lou.bluegrass.net) |
16:14.08 | tholo | From Webster: something established by authority, custom, or general consent as a model or example: CRITERION |
16:14.10 | bkw_ | in your mind maybe... |
16:14.31 | *** join/#asterisk Stealth (~Stealth_M@h-67-101-129-177.NYCMNY83.dynamic.covad.net) |
16:14.51 | Corydon76-work | bkw_: apparently, if you want it to be standard, it's a standard, and if not, it's not. That's not a viable definition of a standard |
16:15.08 | anoosh | lol |
16:15.34 | bkw_ | tholo pasted the def. of a standard |
16:15.52 | Corydon76-work | Yes, and by that definition, both IAX and Ogg are standards |
16:15.54 | tholo | "General consent" is enough for it to be a standard. |
16:16.09 | bkw_ | I don't feel ogg is |
16:16.13 | bkw_ | IAX might be |
16:16.17 | bkw_ | but I don't think ogg is |
16:16.22 | Corydon76-work | Like, I said, it's all about what you think, bkw |
16:16.28 | bkw_ | :P |
16:16.28 | anoosh | ok, could we drop this 'standard' definition, cuz it's really not important in this case (using ogg for moh) doesn't anybody agree with me? |
16:16.31 | bkw_ | ok NEXT SUBJECT! |
16:16.50 | anoosh | hehe |
16:17.19 | RoyK | are you still discussing mp3 vs ogg? |
16:17.26 | anoosh | i am |
16:17.32 | RoyK | GIF RULES |
16:17.47 | anoosh | i think any fairly experienced linux user CAN use ogg if s/he wants to |
16:17.52 | Corydon76-work | If you want to create a patch which allows using ogg for playing MOH without breaking mp3 support, I'd be all for it |
16:18.00 | anoosh | therefore it's not important how widely it's accepted |
16:18.22 | anoosh | it's good enough for solving the problem at hand: mp3 license issue |
16:18.23 | RoyK | I started to change MP3Player to use sox instead but was never finished... |
16:18.27 | RoyK | that's the solution |
16:18.30 | RoyK | use sox |
16:18.48 | zwi | just my .02...Ogg *is* becoming more standard..now that they have implemented an int based decoder there are several HW players avail...that to me would indicated acceptance of ogg in general |
16:18.51 | anoosh | where's bkw_ |
16:18.57 | anthm | maybe the standard belongs in the * itself a standardized interface to allow any external music playing app to register itself as a moh agent |
16:19.22 | anoosh | that would be nice anthm |
16:19.26 | Corydon76-work | Well, you'd need to expand how moh works, then |
16:19.36 | RoyK | it'd work with sox |
16:19.45 | anthm | better to got from 1 to infinity that 1 to 2 |
16:19.47 | Corydon76-work | Right now, everything is hardcoded to use mpg123 |
16:19.52 | RoyK | sox can have any sort of plugins |
16:20.01 | RoyK | Corydon76-work: I know - that sux |
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16:20.35 | *** join/#asterisk zeta_ (~zeta@207.88.150.254.ptr.us.xo.net) |
16:20.39 | anoosh | royk any resources on sox? |
16:20.43 | Corydon76-work | And speaking from experience, Mark will not accept any patch which breaks existing functionality... |
16:21.06 | RoyK | anoosh: man sox |
16:21.07 | RoyK | :) |
16:21.44 | anoosh | oh, lol |
16:22.11 | anoosh | thanks royk |
16:22.35 | jets | is there any _good_ documentation on AGI? |
16:22.53 | bkw_ | not really |
16:22.53 | RoyK | jets: /dev/zero |
16:22.59 | anthm | could easily make the hard code be a variable that you can fill in form config and have no refrence to those config default back to mpg123 settings |
16:23.09 | anthm | instant pick your app support |
16:23.10 | zeta_ | heh |
16:23.50 | zeta_ | is there a spot people have submitted code that uses agi? I've seen a few files from voip-info's wiki, but not much |
16:26.04 | RoyK | anthm: perhaps you can just use AGI to code a sox interface or something :) |
16:29.20 | Corydon76-work | RoyK: yeah, right. Create a new fork everytime you need MOH? |
16:29.38 | RoyK | so? |
16:30.10 | Corydon76-work | I don't know if anybody else does this, but everytime I need something from AGI, I write an application which does the same thing that I need... everything gets done from the dialplan on my machines... |
16:30.27 | *** join/#asterisk TeKP (~TeK@lan.ciberlynx.net) |
16:30.30 | Corydon76-work | Threading rocks |
16:30.46 | bkw_ | yep |
16:30.52 | knight- | anyone have any AGI to pass incoming caller id to a script? |
16:31.13 | TeKP | whenever this one particular cisco sip phone trys to register I get this in my asterisk logfile (and it fails to register): File chan_sip.c, Line 5175 (sipsock_read): Recv error: Resource temporarily unavailable |
16:31.16 | Corydon76-work | Uh, that's what AGI is, knight-... a script... |
16:31.18 | TeKP | anyone know what could cause that? |
16:31.27 | GangBang | bkw |
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16:32.02 | *** part/#asterisk TimRiker (~timr@proxyle02.ext.ti.com) |
16:32.20 | Powerkill | someone have the patch of app_dial that return the status code (busy answered unanswanred to the agi script ? |
16:32.22 | knight- | I know AGI is a scripting language. |
16:32.42 | knight- | I need to pass it to a command line program, that will broadcast the callerid to the appropriate machines |
16:33.19 | knight- | nufone_rating? |
16:33.45 | bkw_ | yes i'm abusing the raing_engine jerjer wrote |
16:34.01 | knight- | ratings for what? |
16:34.19 | dheckaman | whats a free instant message app? |
16:34.42 | dheckaman | I cant find a working jabber server for windows |
16:34.50 | bkw_ | knight- you know when you get a bill from the phone company.. with a list of calls.. you need a rating engine for that |
16:34.51 | dheckaman | thats free |
16:35.01 | knight- | bkw, gotcha |
16:35.04 | zoa | <zoa> i have 144 instances of asterisk running :( |
16:35.09 | bkw_ | ouch |
16:35.21 | knight- | bkw, so nufone_rating.so queries nufone's database? |
16:35.28 | bkw_ | no mine boy |
16:35.29 | bkw_ | mine |
16:35.33 | knight- | gotcha |
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16:40.13 | Corydon76-work | knight-: actually, AGI is not a scripting language. It's just an interface. AGI can be used in many different scripting languages. |
16:40.37 | *** join/#asterisk tmoertel_moz (~chatzilla@pa-bethlprk-cad2-grp1-2d-170.pittpa.adelphia.net) |
16:41.06 | UnixDawg | ok where is everyone getting the latest flash for the grandstreams mine is outof dat compaired to others |
16:41.32 | UnixDawg | and the grandstream site seems to be otu of date |
16:41.41 | tholo | Grandstream made a special deal for you. |
16:42.00 | UnixDawg | compaired to others getting new phones |
16:42.02 | UnixDawg | no |
16:42.04 | tholo | Just to ensure less problems w/FreeBSD 5.1... |
16:42.08 | tholo | ;-) |
16:42.16 | UnixDawg | I am back to 4.9 |
16:42.20 | Stealth | :) |
16:42.21 | UnixDawg | after a hd crash |
16:42.31 | tholo | I didn't say better deal, I said special deal. |
16:42.52 | UnixDawg | I want the uptodate flash |
16:42.55 | Corydon76-work | Oh, good, it must have made an impression on you... |
16:42.57 | tholo | Seriously, ...3.81 is the last _official_ release. People running ...4.x are running pre-release stuff. |
16:43.11 | UnixDawg | ahh |
16:43.13 | Stealth | shit .... |
16:43.22 | decode | hmm |
16:43.41 | UnixDawg | Software Version: Program--1.0.4.17 Bootloader--1.0.0.11 HTML--1.0.0.19 |
16:43.42 | bkw_ | haha |
16:43.47 | UnixDawg | is what I am looking to test |
16:43.48 | bkw_ | 4.17 works for me |
16:43.48 | *** part/#asterisk sjoeperd (~sjoerd@213-84-218-42.adsl.xs4all.nl) |
16:44.13 | UnixDawg | thats what I want to get my hands on |
16:44.13 | UnixDawg | I am 3.18 |
16:44.44 | decode | *sighs* |
16:44.49 | UnixDawg | but thats me wanting to push the edge |
16:45.21 | tholo | So ask Grandstream if you can have access to their pre-release stuff, but don't complain when things don't work for you. |
16:45.31 | decode | hmm |
16:45.41 | blitzrage | morning all |
16:47.00 | UnixDawg | well my pos compiles and works |
16:47.04 | UnixDawg | I am happy |
16:47.57 | *** join/#asterisk bobman (~bobman@mube.psouth.net) |
16:48.15 | decode | what's the command to invoke the user directory? |
16:48.18 | decode | Directory? |
16:48.26 | *** join/#asterisk rpb (~rpb@clt74-104-173.carolina.rr.com) |
16:49.18 | Mike | hey guys the server that has asterisk uses upload bandwidth for remote clients |
16:49.28 | Mike | and remote clients dont use much upload right? |
16:50.17 | decode | I'm really bored.. |
16:50.20 | Mike | do i save bandwidth if i have to asterisk servers? |
16:50.25 | decode | anyways, what would be an easy way to access music-on-hold queues from the menu? :) |
16:51.46 | blitzrage | decode: you're bored? Here, take the keys to my car and go and get it e-tested. I'll go back to sleep |
16:52.06 | UnixDawg | ok test 2 of new add functions |
16:52.21 | blitzrage | I'm out. Latah |
16:52.58 | decode | blitzrage can i go pick my gf up first? :) |
16:53.24 | decode | UnixDawg how to call musiconhold from a context? :) |
16:53.32 | blitzrage | decode: sure, bring her over, I could always use a good romping :) |
16:53.42 | decode | exten => 8,1,MusicOnHold |
16:53.59 | decode | blitzrage only if you're female and hot ;) |
16:54.04 | UnixDawg | looks right |
16:54.21 | blitzrage | decode: why does it matter, you won't be involved :) |
16:58.05 | UnixDawg | hold sorry brb workign on 5 thigns at once |
16:59.20 | UnixDawg | trying to add to my pos software and get the pbx intergrated |
16:59.34 | UnixDawg | and got a compile error |
17:00.30 | UnixDawg | I dont have music on hold workon fbsd yet due to lack of zztdumy |
17:00.37 | UnixDawg | I am on on linux |
17:00.50 | *** join/#asterisk nicebub (~bob@216.253.86.210) |
17:00.51 | *** join/#asterisk n3wb33 (~kapejod@pD9E81940.dip.t-dialin.net) |
17:02.01 | UnixDawg | not on |
17:07.03 | decode | heh |
17:07.19 | decode | zztdumy? |
17:07.26 | decode | i'm using fbsd.. |
17:07.37 | decode | 4.9 to be exact |
17:07.53 | *** join/#asterisk outtolunc (~me@h-66-166-19-148.SNVACAID.covad.net) |
17:08.43 | ionix- | Hey guys, do you suggest Linux or FreeBSD for asterisk. I am fluent in both OS but I want to know the one that works better with drivers and asterisk and it's modules. Thx |
17:08.49 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
17:09.05 | bkw_ | ionix- use linux |
17:09.10 | bkw_ | if you wanna keep your sanity |
17:09.12 | denon | unfortunately |
17:09.23 | ionix- | ok :/ |
17:09.24 | Alric | I hate tech support. |
17:09.27 | bkw_ | fluent in both.. like they are that diffrent? |
17:09.38 | ionix- | bkw_: Very if you do more than bash :) |
17:09.42 | denon | ionix: of course, you could use FreeBSD and help the development effort |
17:09.52 | denon | talk to UnixDawg |
17:10.02 | denon | he'll give you the lowdown on freebsd support |
17:10.14 | ionix- | denon: I will learn then master it before considering developing. But I will probably ending up coding for asterisk |
17:10.15 | bkw_ | ionix- um no they aren't that diffrent |
17:10.40 | denon | ionix-: well, by saying help the dev effort, I didnt mean you had to code, just that you could find and isolate bugs |
17:11.04 | ionix- | bkw_: Well Linux is a posix kernel with a bunch of GPL software. FreeBSD is a complete base system in the *BSD family. It comes from unix |
17:11.20 | ionix- | they are as different as windows and Mac OS9. They both have icons but that's it |
17:11.36 | bkw_ | ionix- but they aren't functionaly diffrent from the user point of view |
17:11.58 | ionix- | denon: I understand but first I want to master it the easy way so I can be more efficient at isolating bugs |
17:12.27 | ionix- | bkw_: yes, if you stay in a bash-like environement and run freebsd with a linuxbase :) |
17:12.38 | ionix- | also /usr/ports is wonderful :) |
17:12.41 | bkw_ | ionix- no it isn't |
17:12.46 | *** join/#asterisk jtodd (~jtodd@207.141.153.205) |
17:12.49 | bkw_ | their are small diffrences |
17:12.54 | Art3d | Anyone ever tried OS X? |
17:12.57 | outtolunc | icons? since when does linux or freebsd have icons? <G> |
17:13.19 | ionix- | I won't argue the FreeBSD/Linux discussion anymore we just have different opinions |
17:13.45 | Powerkill | why accountcode is not set on my cdr log even if i specify it on my agi ? |
17:14.00 | UnixDawg | fbsd works file libpri works and * works we are waiting on zapata drivers |
17:14.04 | bkw_ | ionix- how long have you been using linux or freebsd? |
17:14.15 | bkw_ | but gentoo has really impressed me |
17:14.17 | bkw_ | its more bsdish |
17:14.31 | bkw_ | :P |
17:15.17 | ionix- | bkw_: Like 10 years |
17:15.18 | ionix- | never used gentoo |
17:15.18 | bkw_ | gentoo is very impressive |
17:15.37 | bkw_ | portage blows freebsd ports away |
17:15.54 | bkw_ | wish freebsd ports had the same features... but oh well.. i still like freebsd over linux anyday |
17:16.04 | ionix- | bbl |
17:16.27 | Powerkill | bkw any idea why $AGI->exec('SetAccount',$pin) don't set the pincode in the cdr ? |
17:16.35 | *** join/#asterisk lexluther (~You@212.247.106.52) |
17:16.45 | bkw_ | Powerkill it makes a new file $pin.csv |
17:16.47 | *** part/#asterisk lexluther (~You@212.247.106.52) |
17:16.50 | bkw_ | if you use csv stuffs |
17:17.13 | Powerkill | and how can I put it on mysql ? |
17:17.19 | bkw_ | cdr_mysql |
17:17.28 | Mike | bkw_: do i save bandwidth if i have 2 asterisk servers??? |
17:17.33 | Powerkill | my accountcode colum is empty |
17:17.43 | Mike | 3 extensions in my server 1 extension in the other server |
17:17.44 | Powerkill | bkw_ i'm using cdr_myslq |
17:17.47 | Mike | do i save bandwidth? |
17:17.52 | bkw_ | Mike I will answer that if you hire me as a consultant... but I and many others have already answered that question. |
17:18.06 | bkw_ | Powerkill it should be in there |
17:18.07 | decode | bkw_ portupgrade is really nice |
17:18.22 | bkw_ | Powerkill you need a rating engine |
17:18.24 | Powerkill | but it's not |
17:18.28 | n3wb33 | Mike: you will save bandwidth if you have 0 * servers! |
17:18.30 | Powerkill | rating ??? |
17:18.41 | bkw_ | decode portupgrade is nice |
17:18.44 | Mike | n3wb33: i have 1 asterisk server right now |
17:18.48 | Mike | n3wb33: 3 extensions |
17:18.59 | Mike | n3wb33: i want to put another extension on a remote point |
17:19.13 | Mike | n3wb33: but i could also put a server on the remote point |
17:19.20 | Mike | n3wb33: the thing is would it be worth? |
17:19.28 | bkw_ | n3wb33 want the cluebat? |
17:19.45 | n3wb33 | bkw_: please |
17:19.46 | Mike | bkw_: how much per hour/answer? |
17:20.11 | bkw_ | Mike depends on what you want! |
17:20.18 | bkw_ | ok that sounded kinda whorish |
17:20.32 | decode | hah |
17:20.33 | Mike | i just want to know if i would save bandwidth putting another server |
17:20.36 | Mike | on the remote point |
17:20.42 | decode | * is cool :) |
17:20.49 | decode | bbiab tho |
17:20.50 | Powerkill | bkw_ any other idea ? |
17:20.51 | decode | cleaning lights |
17:20.52 | bkw_ | Mike yes you could save bandwidth |
17:20.58 | UnixDawg | man I hate when a errror is due to case |
17:21.19 | bkw_ | Powerkill check your msges |
17:21.21 | Mike | bkw_: how much and why? |
17:21.43 | bkw_ | the ability to use lower bandwidht codecs between the * servers |
17:21.59 | Mike | without lossing quality? |
17:22.43 | dheckaman | ilbc isnt too bad quality |
17:22.52 | dheckaman | gsm is good too |
17:23.01 | dheckaman | what is it, 30 kb/s? |
17:23.06 | dheckaman | ~30/8 |
17:23.07 | | 3.75 |
17:23.13 | Mike | nice |
17:23.24 | Mike | and its as good as ulaw? |
17:23.26 | Mike | alaw? |
17:24.03 | dheckaman | not AS good |
17:24.03 | dheckaman | but not noticable |
17:24.03 | cypromis | nothing is as good as ua/law |
17:24.03 | dheckaman | try it out |
17:24.03 | dheckaman | and see for your self |
17:24.32 | ares_ | i getting like really bad echo on mine and low volume... increasing the gain increases the echo. :\ |
17:24.38 | Mike | ulaw and alaw simply are really as good quality as landlines |
17:24.49 | dheckaman | yep |
17:24.56 | dheckaman | Ares: I have the same problem |
17:25.01 | dheckaman | exept no echo |
17:25.05 | ares_ | i read the mailing list...same problem, no solution.. |
17:25.08 | dheckaman | but horrible volume |
17:25.11 | Mike | so you guys recommend gsm? |
17:25.18 | dheckaman | gsm and ilbx |
17:25.19 | dheckaman | ilbc |
17:25.25 | Mike | whats better gsm? |
17:25.31 | dheckaman | ilbc uses slightly less BW |
17:25.32 | ares_ | no solution? |
17:25.36 | dheckaman | but uses more CPU |
17:25.45 | Mike | dheckaman: but the quality |
17:25.47 | dheckaman | Ares: there IS a solution, just havnt found one yet :) |
17:26.02 | Mike | well i dont think 4 extensions in total can use alot of cpu:) |
17:26.03 | ares_ | did you try increasing the gain? |
17:26.03 | dheckaman | Mike: cant say, try it out your self and see how it sounds |
17:26.12 | dheckaman | Mike: indeed |
17:26.13 | Mike | im not really setting up AT&T network or something |
17:26.18 | dheckaman | yea |
17:26.45 | dheckaman | ilbc has better packet loss recovery too, I think |
17:26.56 | Mike | the problem is grandstream phones dont support |
17:26.56 | Mike | gsm |
17:26.57 | Mike | tho |
17:27.07 | dheckaman | they dont? |
17:27.21 | dheckaman | oh |
17:27.25 | Mike | plus i have no idea how to link 2 asterisk servers |
17:27.27 | dheckaman | they must support ulaw |
17:27.27 | dheckaman | :) |
17:27.31 | dheckaman | me niehter |
17:27.34 | dheckaman | thats for the higher ups |
17:27.35 | Mike | yes they support ulaw |
17:27.41 | Mike | but asterisk converts it back to ulaw? |
17:27.46 | dheckaman | sure |
17:27.46 | ares_ | i used IAX to link, it's so easy ... |
17:27.57 | Mike | ares IAX? |
17:28.00 | ares_ | just use Dial/IAX/EXTENSION |
17:28.00 | Mike | and what codec do you use? |
17:28.01 | dheckaman | it can do ilbc to ulaw or whatever |
17:28.06 | ares_ | yea...read up on that. |
17:28.07 | ares_ | it's like SIP |
17:28.13 | ares_ | or you can even use SIP too |
17:28.14 | dheckaman | but better |
17:28.15 | RoyK | Dial(AIX,ibm) |
17:28.28 | dheckaman | if its between 2 * servers, use IAX |
17:28.38 | dheckaman | RoyK: hah |
17:28.52 | Mike | ok ill use IAX |
17:28.59 | Mike | anyone has any url on where to read |
17:29.01 | Mike | an example |
17:29.03 | Mike | or a howto |
17:29.05 | Mike | explanation |
17:29.10 | UnixDawg | ok the pos works |
17:29.11 | Mike | or what ever that can help me figure it out |
17:29.11 | Mike | ? |
17:29.18 | UnixDawg | but not the new addin |
17:29.21 | UnixDawg | grr |
17:29.21 | dheckaman | read the email list archive |
17:29.38 | dheckaman | and digium.com has lots of info in the documentation section |
17:30.52 | UnixDawg | who know the url for the webpage where you type in text and it creates a wav file with a voice reeading the text |
17:31.06 | dheckaman | ~att |
17:31.13 | dheckaman | ~rhetorical |
17:31.17 | dheckaman | google for it |
17:31.27 | bde | UnixDawg: http://www.research.att.com/projects/tts/demo.html |
17:31.29 | UnixDawg | I have not finding it |
17:31.38 | dheckaman | rhetorical is better |
17:31.39 | UnixDawg | thnks bde |
17:32.44 | Powerkill | ~seen JerJer |
17:32.47 | | jerjer is currently on #asterisk |
17:32.52 | Powerkill | JerJer are you here ? |
17:34.41 | decode | omg |
17:34.50 | decode | i just found a trashbag full of marijuana in my attic :) |
17:35.26 | *** join/#asterisk h3x (~hex@user57.net472.lv.sprint-hsd.net) |
17:35.54 | decode | dheckaman if it were illegal here :) |
17:36.07 | decode | my hostname is not where i am :) |
17:36.41 | dheckaman | heh |
17:37.02 | *** join/#asterisk Exomorph (Greg@134-9.bvcompuworks.com) |
17:37.23 | dheckaman | anyone know of a free instant message server? |
17:37.31 | carrar | ICQ |
17:37.34 | dheckaman | I cant find a free windows server that works |
17:37.42 | dheckaman | I said SERVER :P |
17:37.47 | carrar | ircd |
17:37.52 | dheckaman | ugh |
17:37.57 | carrar | heh |
17:38.04 | dheckaman | I want something like MSN or something |
17:38.08 | carrar | it's instant |
17:38.09 | carrar | and |
17:38.18 | carrar | You can use the windows trillian IM client with it |
17:38.21 | cypromis | jabber |
17:38.27 | cypromis | :) |
17:38.27 | tzanger | jabber kicks ass |
17:38.29 | Powerkill | dheckaman jabber |
17:38.31 | anthm | jabber |
17:38.32 | tzanger | psi is my client of choice |
17:38.53 | carrar | write one in perl |
17:39.18 | dheckaman | I cant find a free jabber windows server |
17:39.21 | dheckaman | that works |
17:39.26 | dheckaman | psi? |
17:40.45 | tzanger | psi.sf.net |
17:40.46 | tzanger | I think |
17:40.55 | tzanger | psi.affinix.com rather |
17:41.16 | bkw_ | pisy |
17:41.20 | tzanger | multiplatform, qt-based and does NOT take up all your screen real estate.. it behaves very much like the old ICQ client, which is IMO the best |
17:41.25 | tzanger | hahahha |
17:41.33 | dheckaman | hmm, but thats not the server |
17:41.34 | dheckaman | :( |
17:41.42 | tzanger | dheckaman: well use jabberd then |
17:42.05 | dheckaman | on windows? |
17:42.29 | decode | "Chong's Bong. Press 1 for sales, 2 for seed catalog orders, or 3 for order information" |
17:42.39 | dheckaman | lol |
17:42.49 | decode | Chong's Bongs that is |
17:43.01 | dheckaman | oh cool |
17:43.04 | dheckaman | I found it |
17:43.16 | decode | heh.. |
17:43.23 | decode | My dialplan is all screwed up |
17:44.04 | heller | ah. you need the 'unscrewup' command. |
17:44.49 | decode | still cant get fwd stuff working, yet.. |
17:44.53 | decode | but that's to be fixed tonight |
17:44.58 | decode | as are incoming fwd calls :) |
17:45.28 | decode | bbl, gf and i are off to the shower >:) |
17:46.17 | carrar | bf? you kick out your bf |
17:46.23 | carrar | err gf |
17:46.27 | carrar | haha |
17:46.28 | dheckaman | LOL |
17:53.36 | *** join/#asterisk zatu (~zatu@6-VALL-X5.libre.retevision.es) |
17:55.09 | *** part/#asterisk mortck (~test@p213.54.146.115.tisdip.tiscali.de) |
18:00.47 | decode | ok, question |
18:00.58 | decode | i wanna send calls to 411 to 411 via fwd, how? :) |
18:01.00 | carrar | sorry no more questions today |
18:01.53 | *** join/#asterisk Simon_ca (~sedgett@h24-81-97-130.vc.shawcable.net) |
18:04.20 | *** part/#asterisk pablo_slw (~zatu@6-VALL-X5.libre.retevision.es) |
18:04.54 | *** join/#asterisk rajo_home_ (~rainer@p508AEFB0.dip.t-dialin.net) |
18:06.27 | *** join/#asterisk ctooley (~ctooley@199.89.146.58) |
18:06.46 | Connor- | anyone have problems with compiling pwlib? |
18:06.53 | ctooley | Which of the VoIP providers that offer an unlimited call plan is most likely to work with Asterisk |
18:11.08 | *** join/#asterisk UnixDawg (~UnixDawg@ny-lasalle6c-203.buf.adelphia.net) |
18:16.59 | RoyK | ctooley: what do you mean - unlimited call plan? |
18:17.10 | jsharp | Flat rate. No per minute charges. |
18:17.24 | ctooley | RoyK: $19.95/month no per minute fees |
18:17.42 | bkw_ | all of em work with * |
18:17.52 | bkw_ | just gotta use an FXO port on some of em |
18:18.12 | ctooley | bkw_: :) That's not really what I meant. |
18:18.36 | RoyK | unless they're running JHSVOIPP |
18:18.37 | ctooley | I'm trying to avoid the whole analog conversion part. |
18:19.55 | bkw_ | jsharp ahhahahahah funny |
18:19.57 | *** join/#asterisk rusty (~rusty@65-101-254-85.dnvr.qwest.net) |
18:20.27 | ctooley | jsharp: I've got that! |
18:21.46 | UnixDawg | grrrr |
18:21.49 | bkw_ | we need sip.conf per peer agent strings |
18:22.16 | Connor- | need a way to use Radius for sip.conf... |
18:22.21 | zoa | yes |
18:22.25 | ctooley | I've written an HTML-RPC language, for switch to switch updats, that has all of the negatives of a verbose slow communication protocol without the flexibilty of XML-RPC and thats an "HTML Networking Technology" |
18:22.41 | Connor- | I don't give a phuck what JerJer says. :) |
18:22.47 | RoyK | Connor-: why the fsck do you want radius? |
18:22.54 | zoa | top - 22:22:34 up 8 days, 11:21, 3 users, load average: 20.98, 20.97, 20.91 |
18:22.56 | zoa | Tasks: 198 total, 22 running, 174 sleeping, 0 stopped, 2 zombie |
18:22.56 | zoa | Cpu(s): 2.3% user, 0.6% system, 0.0% nice, 97.1% idle |
18:23.05 | RoyK | Connor-: jerjer's a jerk. don't bother about what he says |
18:23.33 | Connor- | Because, I've got a very nice frontend billing and controll system I use for my ISP which works with Radius. |
18:23.41 | Connor- | No need to re-invinte the wheel for VoIP |
18:23.52 | nicebub | i agree |
18:23.56 | nicebub | same for me |
18:23.57 | RoyK | agreed |
18:24.17 | RoyK | so - what you need is an app_radius |
18:24.19 | RoyK | or something |
18:24.20 | Connor- | I just got my cisco working for ANI authentication with my radius and it works great. |
18:24.31 | jsharp | Or pam_radius. |
18:24.35 | zoa | top - 22:24:14 up 8 days, 11:22, 3 users, load average: 21.10, 21.00, 20.92 |
18:24.36 | zoa | Tasks: 198 total, 22 running, 174 sleeping, 0 stopped, 2 zombie |
18:24.36 | zoa | <PROTECTED> |
18:24.36 | zoa | <PROTECTED> |
18:24.36 | zoa | <PROTECTED> |
18:24.36 | zoa | <PROTECTED> |
18:24.38 | zoa | Mem: 2068704k total, 1261584k used, 807120k free, 17008k buffers |
18:24.40 | zoa | Swap: 2097136k total, 0k used, 2097136k free, 1087992k cached |
18:24.41 | Connor- | Well.. need a app.. but how would it work to replace sip.cof ? |
18:24.44 | zoa | any suggestions ? :((( |
18:24.59 | RoyK | zoa: you have too many CPUs |
18:25.03 | Connor- | zoa, you using software raid ? |
18:25.10 | Connor- | Nah.. he's dual CPU with HT |
18:25.15 | zoa | nopez, hardware raid |
18:25.23 | *** join/#asterisk adam_gafachi (~diddy@69-55-69-130.da.netsville.net) |
18:25.26 | RoyK | zoa: what is it that uses all that? |
18:25.30 | RoyK | load avg 21????? |
18:25.30 | zoa | asterisk |
18:25.37 | zoa | 144 instances of them |
18:25.41 | bkw_ | zoa did you compile everythign to use SMP |
18:25.41 | RoyK | shite |
18:25.44 | bkw_ | zaptel and all the goodies? |
18:25.47 | zoa | yes |
18:25.57 | RoyK | tried disabling smp or ht? |
18:26.06 | zoa | it normally is max 3% usage |
18:26.13 | zoa | but after an hour this is what happens |
18:26.23 | RoyK | zoa: after an idle hour? |
18:26.36 | zoa | nopez, after an hour of 20 simultaneous calls |
18:26.37 | RoyK | zoa: connect to it with gdb and see what's going on |
18:26.52 | bkw_ | zoa get kram that login info and let him look at it |
18:26.58 | zoa | k |
18:28.07 | Connor- | So anyway, I think we need to have a way to do Radius for sip.conf |
18:28.08 | RoyK | Connor-: how does radius work? do you have a radius server everyone authenticates to, and then anyone can ask that server 'is tom really tom and should he be there?' |
18:28.34 | Connor- | 2 ways, Auth only, or Auth and pass attributes needed |
18:28.43 | Connor- | For the AUTH side.. |
18:28.50 | Connor- | then there is accounting, when you |
18:29.02 | Connor- | have auth'd, the radius gets a accounting START record.. |
18:29.05 | bkw_ | ewww |
18:29.10 | Connor- | when the session ends, you get a stop record. |
18:29.22 | bkw_ | Connor dont strat the radius flame.. please |
18:29.28 | bkw_ | if you want it.. write it.. |
18:29.35 | Connor- | Not wanting it for tracking calls.. |
18:29.53 | Connor- | I'm wanting it for a replacement for sip.conf |
18:29.53 | bkw_ | I know but it will lead to call tracking |
18:30.09 | bkw_ | someone was working on a pam module for * |
18:30.11 | bkw_ | who was that |
18:30.16 | bkw_ | then you coud use a pam radius module |
18:30.25 | Connor- | That's easy enough anyway.. We do that with or Term Server anyway.. What's the diff? |
18:30.44 | jsharp | That would be me. |
18:30.46 | Connor- | That's what Radius was designed for.. Authenticating and Accounting. |
18:30.56 | jsharp | I'm hacking on the code right now, in fact. |
18:31.03 | bkw_ | jsharp what will it allow? |
18:31.15 | jsharp | Right now, I'm just using it for authentication. |
18:31.26 | zoa | kram has been idle 3mins 53secs, signed on Fri Oct 31 23:31:55 |
18:31.30 | zoa | -kram- I left 14h 4m 40s ago it is 12:29PM right now. info: Not here right now, sorry... |
18:31.39 | bkw_ | but in theory he could use radius to auth then right? |
18:31.41 | zoa | yeah right kram :p |
18:31.44 | jsharp | Aye, he could. |
18:31.57 | h3x | zoa: probably anti-idle or something |
18:32.04 | RoyK | if someone were to use asterisk to allow 500 users to use it, 120 concurrently... is this possible with a single server? |
18:32.29 | bkw_ | RoyK should be possible |
18:32.35 | jsharp | With a beefy enough machine, sure. |
18:32.43 | bkw_ | ya what he said |
18:32.55 | RoyK | where would the limit go, if using analog phones? |
18:33.00 | zoa | royk: im getting a quad opteron |
18:33.05 | zoa | ill let you know |
18:33.14 | *** join/#asterisk jtodd2 (~jtodd@207.141.153.205) |
18:33.19 | Connor- | Can I ask a question.. Why are so many people against Radius? |
18:33.34 | jsharp | Cause a lot of people think of it as a really ugly hack. |
18:33.39 | zoa | i can do 80 with ilbc on a dual xeon max |
18:33.56 | jsharp | Will * build and run on an opteron? |
18:33.59 | Connor- | jsharp: explain ? |
18:34.08 | zoa | why would it not build and run : |
18:34.09 | zoa | ? |
18:34.13 | high-rez | jsharp: in 32 bit mode for sure. |
18:34.14 | cypromis | jsharp: it should |
18:34.15 | jtodd2 | bkw_: re: your per peer agent strings: see my long "why sip.conf needs to see some re-work" post of a few months ago. It's in there, with a bunch of other issues that probably would improve *'s scalability for sip. |
18:34.23 | cypromis | I am just ordering a couple to work as codec translation engines |
18:34.55 | zoa | cypromos: when will they arrive ? |
18:34.57 | cypromis | RoyK: a dual xeon will do 120 concurrent calls |
18:35.04 | cypromis | zoa: no idea, I am in poland :))) |
18:35.11 | zoa | cypromis: depends on the codec |
18:35.28 | cypromis | zoa: ok |
18:35.33 | RoyK | cypromis: is that a xeon with 1MB L3 cache? |
18:35.50 | cypromis | nope |
18:35.53 | cypromis | we use the 512 ones |
18:35.58 | zoa | the l3 cache is non relevant according to toms hardware |
18:36.34 | RoyK | sounds strange ... |
18:36.49 | Bonbon | don't the allow/disallow codec commands work within an iax context? |
18:36.50 | RoyK | more cache usually gives speed on at least databases and such |
18:36.52 | zoa | the 1mb version is not a lot faster i mean |
18:37.05 | RoyK | at least not for asterisk use... |
18:37.08 | zoa | bonbon: i have the same impression |
18:37.16 | Bonbon | the don't work |
18:37.42 | Bonbon | i have disallow=all, allow=ilbc in the general context |
18:37.49 | zoa | pfft, i cant even do 40 concurrent pass thru calls atm :( |
18:37.55 | Bonbon | then i have disallow=all, allow=ulaw in one of the contexts. |
18:38.04 | zoa | and i need 350 concurrent calls |
18:38.05 | Bonbon | yet, it still negotiates ilbc |
18:38.08 | cypromis | I think you cannot allow a codec per peer that you disallow in the general context |
18:38.10 | cypromis | besides that |
18:38.11 | cypromis | it works fine |
18:38.13 | zoa | bonbon: same here |
18:38.14 | RoyK | zoa: 350??? |
18:38.16 | RoyK | how many phones? |
18:38.19 | zoa | 350 |
18:38.25 | UnixDawg | busniess |
18:38.30 | *** join/#asterisk ajh (~ajh@c-24-0-23-89.client.comcast.net) |
18:38.32 | RoyK | yeah - but that doesn't mean 350 concurrent calls. |
18:38.33 | cypromis | zoa: on what box |
18:38.41 | cypromis | we have far far far more than 40 passthrough calls in parallel |
18:38.42 | RoyK | that means some ~150 or so |
18:38.43 | zoa | that means 350 concurrent calls :) |
18:38.44 | Bonbon | cypromis: i've allowed all codecs in the general context |
18:38.54 | cypromis | Bonbon: strange |
18:38.58 | cypromis | but on the other hand |
18:39.11 | cypromis | we don't use ilbc anymore sinc it is somehow buggy |
18:39.12 | RoyK | zoa: what chance is there that everyone in the building are speaking on the phone at the same time? |
18:39.24 | RoyK | cypromis: stopped using it? |
18:39.25 | zoa | its an outbound callcenter |
18:39.35 | zoa | quite high id say :) |
18:39.38 | RoyK | oh |
18:39.38 | RoyK | ok |
18:39.41 | Bonbon | cyypromis: disallow=all |
18:39.41 | Bonbon | allow=ilbc |
18:39.41 | Bonbon | allow=gsm |
18:39.41 | Bonbon | allow=ulaw |
18:39.41 | Bonbon | allow=alaw |
18:39.47 | zoa | they want me to go up to 3000 |
18:39.52 | decode | blah |
18:39.58 | zoa | they are insane |
18:40.00 | Bonbon | cypromis: in specific context: |
18:40.05 | UnixDawg | ok now to conver to gsm |
18:40.05 | nicebub | did nufone write their own AAA system? |
18:40.06 | Bonbon | disallow=all |
18:40.06 | Bonbon | allow=ulaw |
18:40.08 | Connor- | Multiple boxes man. Lots of them |
18:40.11 | cypromis | hmm I have nothing in general and do specific only |
18:40.22 | decode | can't get this damn thing working :( |
18:40.47 | decode | (freeworld dialup incoming forwarded to sipphone of name xlite) |
18:41.02 | UnixDawg | sox file.wav -??? 8000 file.gsm |
18:41.17 | Connor- | sox file.wav file.gsm |
18:41.20 | cypromis | 3.000 phones ? |
18:41.23 | cypromis | or parallel calls ? |
18:41.34 | RoyK | on a duron 1200 |
18:41.55 | cypromis | nah we don't use amd 32 bit stuff since they get too hot |
18:41.59 | zoa | scxgk3:/var/log/asterisk# lsof | wc -l |
18:42.00 | zoa | <PROTECTED> |
18:42.19 | RoyK | zoa: wtf? |
18:42.47 | zoa | dont ask me |
18:42.51 | zoa | its asterisk :) |
18:43.05 | RoyK | lsof| cut -c67-|grep ^/ | wc |
18:43.22 | RoyK | just want to know how many _files_ there are |
18:44.07 | zoa | scxgk3:/var/log/asterisk# lsof| cut -c67-|grep ^/ | wc |
18:44.07 | zoa | <PROTECTED> |
18:44.14 | zeta_ | wow |
18:44.48 | *** join/#asterisk Chris_DE (~Chris_DE@p5083053C.dip0.t-ipconnect.de) |
18:47.05 | RoyK | zoa: meaning what's a _lot_ is ! files |
18:47.38 | RoyK | zoa: lsof| cut -c67-|grep -w pipe | wc |
18:48.02 | zoa | lots pipes |
18:48.05 | zoa | lotsa |
18:48.13 | RoyK | zoa: lsof| cut -c67-|egrep -v "^/|pipe" | wc |
18:48.43 | zoa | just a sec |
18:49.26 | zoa | scxgk3:/var/log/asterisk# lsof | cut -c67-|egrep -v "^/|pipe" | wc |
18:49.26 | zoa | <PROTECTED> |
18:49.27 | RoyK | zoa: or netstat --inet |
18:49.33 | RoyK | heh |
18:49.34 | *** join/#asterisk simonaut (~simonaut@hoochie.digium.com) |
18:49.39 | RoyK | try netstat --inet|less |
18:49.55 | zoa | only 2 ssh connections |
18:50.00 | RoyK | huh? |
18:51.21 | zoa | scxgk3:/var/log/asterisk# lsof | grep asterisk | wc -l |
18:51.21 | zoa | <PROTECTED> |
18:51.44 | RoyK | ps ax|grep -w asterisk|grep -v grep|wc |
18:51.55 | RoyK | -l |
18:52.20 | zoa | scxgk3:/var/log/asterisk# ps ax|grep -w asterisk|grep -v grep|wc -l |
18:52.21 | zoa | <PROTECTED> |
18:52.32 | UnixDawg | nope need sox file.wav -?s? 8000 file.gsm and I cant recall it all |
18:52.36 | UnixDawg | ~sox |
18:52.38 | | somebody said sox was Sound Processing Tool. URL: http://sox.sourceforge.net/ |
18:52.42 | UnixDawg | ~sox info |
18:52.52 | UnixDawg | ~sox command |
18:52.57 | denon | !info sox |
18:52.57 | denon | ? |
18:52.59 | RoyK | UnixDawg: man sox |
18:53.08 | RoyK | zoa: how many concurrent calls do you have now? |
18:53.09 | denon | hrm, Ainfo is gone |
18:53.29 | zoa | royk: nobody |
18:53.34 | zoa | it stopped working |
18:53.35 | Bonbon | zoa: if a call comes into my server via iax, with no defaulty context or anything then where would it go? |
18:53.39 | zoa | :) |
18:53.43 | RoyK | gdb -p ... |
18:54.08 | UnixDawg | wo knows what the sox output options are to conver from wav to gsm |
18:54.14 | RoyK | UnixDawg: sox file.wav -r 8000 file.gsm |
18:54.29 | RoyK | UnixDawg: it just looks at the extension |
18:54.31 | RoyK | that's all |
18:54.35 | UnixDawg | sounds to slow |
18:54.40 | RoyK | slow? |
18:54.40 | UnixDawg | ok |
18:54.54 | UnixDawg | yes sounds liek a 45 on 33 |
18:54.56 | zoa | 0x08051ad3 in ast_sched_add (con=0x4023a1a8, when=0, |
18:54.56 | zoa | <PROTECTED> |
18:54.56 | zoa | 167 if (SOONER(s->when, current->when)) |
18:55.03 | zoa | is the last thing it gives me |
18:56.11 | tmoertel_moz | Anybody know what " File app_dial.c, Line 518 (dial_exec): Unable to create channel of type 'sip'" means? |
18:56.22 | tmoertel_moz | I can't bridge a Zap -> SIP call. |
18:56.24 | nicebub | run sip debut |
18:56.30 | nicebub | sip debug |
18:56.36 | af_ | which port I have to open to permit h323 traffic between * and an ip public gnomemeeting? |
18:56.36 | nicebub | or reload module |
18:56.37 | RoyK | does SIP have silence detection? |
18:56.42 | af_ | ports |
18:56.55 | RoyK | af_: through nat? |
18:57.05 | tmoertel_moz | nicebub: I am. |
18:57.13 | af_ | no RoyK * is public, and so gnomemeeting |
18:57.27 | RoyK | af_: the RTP stream is chosen at run time, so you need an agent that speaks h323 |
18:57.43 | RoyK | er |
18:57.49 | RoyK | the RTP stream's ports ... |
18:58.49 | af_ | which are those ports? |
18:58.50 | UnixDawg | ok they sound better |
18:59.24 | RoyK | af_: they are determined at run time |
18:59.30 | RoyK | af_: something > 1024 |
18:59.50 | UnixDawg | ok my inbound iax nmbr is now not working |
18:59.50 | af_ | mhhh |
18:59.53 | UnixDawg | grrr |
19:00.04 | af_ | udp or tcp, RoyK? |
19:00.12 | RoyK | rtp uses udp |
19:00.19 | RoyK | voip over tcp??? |
19:00.20 | af_ | thanks |
19:00.21 | RoyK | don't think so |
19:00.32 | RoyK | af_: what sort of router/firewall? |
19:01.00 | af_ | iptables |
19:01.19 | RoyK | there's an h.323 iptable conntrack patch in patch-o-matic |
19:01.31 | nicebub | conntrack? |
19:01.33 | RoyK | but it's not stable yet, so YMMW |
19:01.35 | tmoertel_moz | nicebub: there's no sip traffic |
19:01.44 | Bonbon | what is the [guest] part for in iax.conf? |
19:01.51 | RoyK | nicebub: linux connection tracking |
19:01.58 | nicebub | rework your config file...try unloading and reloading the module |
19:02.12 | af_ | I don't do nat |
19:02.22 | nicebub | Royk: reports status on current connections? |
19:02.25 | zoa | royk: do i have to type anything else in gdb ? |
19:02.26 | RoyK | Bonbon: it allows me to use your PBX to call sex phones |
19:02.50 | RoyK | zoa: I don't really know how to debug this. sorry |
19:02.58 | Bonbon | royk: but i can't get it to work. if i make an iax connection to a box, with no username or secret, then wouldn't it go to this context? |
19:02.59 | zoa | strace is not giving anything |
19:03.09 | tmoertel_moz | nicebub: Dial("Zap/1-1","sip/5001|20") causes the problem |
19:03.19 | Bonbon | royk: it seems to go to the first non-guest context that it can successfully authenticate |
19:03.25 | tmoertel_moz | nicebub: restarting all of * doesn't solve the problem |
19:03.33 | tmoertel_moz | nicebub: no errors in log, either |
19:03.49 | RoyK | nicebub: yes. /proc/net/ip_conntrack |
19:04.05 | nicebub | t_moz:when you list your mdules does it show up |
19:04.13 | tmoertel_moz | <PROTECTED> |
19:04.33 | RoyK | nicebub: with connection tracking enable for a protocol, you can NAT it even if it's 'dirty', like FTP |
19:04.41 | tmoertel_moz | nicebub: chan_sip.so |
19:04.49 | UnixDawg | I have my iax done wrong again |
19:04.52 | nicebub | Royk: i'm checking out the page ,now |
19:05.01 | nicebub | Royk: looks useful |
19:05.24 | Yog-home | is there a difference between Voicemail() and Voicemail2() ? |
19:05.38 | RoyK | IIRC Voicemail2 is 'beta' |
19:05.40 | denon | Yog-home: not anymore |
19:05.50 | denon | no ... Voicemail2 and Voicemail are both Voicemail2 now |
19:05.53 | Yog-home | yeh..it looks the same |
19:05.55 | Yog-home | in the code |
19:05.57 | denon | so just use Voicemail |
19:06.01 | Yog-home | well..from my first glance |
19:06.10 | denon | they are identical as of the other day |
19:06.16 | nicebub | are you leacing bindaddr alone? |
19:06.37 | Yog-home | does anyone have all the "show application XYZ" type documentation collected together in a doc file ? |
19:07.08 | Yog-home | <- new to * |
19:07.54 | zoa | you can find it on voip-info.org |
19:08.27 | Yog-home | o..the wiki |
19:08.55 | tmoertel_moz | grrr... kernel oops in wcfxo |
19:09.13 | UnixDawg | ok who has voicepulse in thier iax.conf that point otu what I missed |
19:09.36 | Yog-home | is the X100P pretty much the 'card to get' for home phone line type deal ? |
19:09.49 | jsharp | Yog-home Yes. |
19:09.54 | *** join/#asterisk Marlow (~marlow@2001:838:300:70:0:0:0:2) |
19:10.12 | Yog-home | yeh i guess it's good too since it supports * |
19:10.43 | jsharp | Yah. And if you buy an X100P, you support the company that wrote asterisk. |
19:10.55 | jsharp | freenode supports ipv6? Interestin. |
19:11.22 | UnixDawg | ? |
19:11.24 | Yog-home | yeh! i just noticed the V6 addy too! |
19:12.49 | jsharp | You'll probably have to get a tunnel through one of the tunnel brokers. |
19:13.04 | Yog-home | Is the X100P made by digium or Zaptel ? I notice the drivers for it are 'Zaptel'... |
19:13.10 | Yog-home | jsharp: yeh probably |
19:13.40 | bkw_ | Yog-home digium |
19:13.40 | jsharp | "zaptel" is the name of the class of hardware that Digium makes and asterisk supports. |
19:13.49 | Yog-home | V6 has nice built in tunneling capability..can be somewhat transparent in some cases |
19:14.00 | Yog-home | ah. I see |
19:14.34 | Yog-home | probabaly from the tunnel broker if your ISP doesn't support it |
19:14.41 | tholo | freenet6 |
19:14.42 | Yog-home | or ARIN maybe for a portable one |
19:14.45 | jsharp | dheckaman: You can go to any of the tunnel brokers. freenet...hurricane electric. |
19:15.07 | jsharp | Yog-home: But if you get one from ARIN, you still need to find an upstream that you can peer with. |
19:15.12 | Yog-home | yep |
19:15.15 | dheckaman | if I have a public ip now, can I just make myself a ipv6 address? |
19:15.19 | dheckaman | and I just show up? |
19:15.21 | jsharp | Nope. |
19:15.31 | dheckaman | oh |
19:15.32 | jsharp | You need to peer up with an ipv6 provider. |
19:15.38 | dheckaman | why? |
19:15.41 | jsharp | like freenet6.net or tunnelbroker.net |
19:15.49 | Yog-home | they probably don't give out those to anyone either..I'm sure they're keeping some sort of CIDR type routing in effect so they want everything aggrogated to keep the # of routes down |
19:15.52 | nicebub | Royk: im running 2.4.18-27.0.8 and don't see /proc/net/ip_conntrack |
19:16.12 | jsharp | For the same reason you need to peer up with someone to get IPV4 connectivity. |
19:16.14 | nicebub | hmm |
19:16.17 | dheckaman | I dont want a tunnel, I just want to connect to freenode on ipv6 |
19:16.19 | jsharp | YOu can't just magically appear on the network, unfortunately. |
19:16.19 | dheckaman | ah |
19:16.21 | RoyK | nicebub: insmod ip_conntrack |
19:16.40 | Yog-home | dheckaman: you need an IP address that's routable to your provider... |
19:16.44 | RoyK | nicebub: it's only useful for firewalls, really |
19:16.49 | dheckaman | ah |
19:16.52 | dheckaman | hmm |
19:17.02 | jsharp | dheckaman: Unless your ISP can give you native IPV6 addresses, you must build an IPV4 GRE tunnel to an IPV6 provider. |
19:17.02 | Yog-home | dheckaman: and I bet 90% of providers today (ISPs) arn't routing V6 yet |
19:17.11 | jsharp | They're not. |
19:17.13 | nicebub | Royk: got it |
19:17.19 | dheckaman | ah, ok |
19:17.33 | RoyK | nicebub: and upgrade your kernel. 2.4.18's OOOOLD |
19:17.51 | nicebub | can i do it from command line?... |
19:17.59 | RoyK | do what? |
19:18.00 | nicebub | the machine is in mexico..i'm in colorado |
19:18.05 | nicebub | upgrade the kernel |
19:18.15 | Yog-home | wonder how much addy space the avg broker is giving out...V6 addresses are so huge that I'dthink you'd get a at least 8 bits of address space easily |
19:18.30 | RoyK | you can, but you need to reboot and if something goes wrong, you're looking forward to a rather long trip |
19:18.31 | jsharp | I had a /48 from hurricane electric. |
19:18.39 | dheckaman | LOL |
19:18.42 | jsharp | Which is a metric buttload of addresses. |
19:18.58 | Yog-home | jsharp: haha..nice ... 16 bits ... "class B" |
19:19.04 | jsharp | Heh. Yeah. |
19:19.11 | nicebub | i do have someone down there in case something happens |
19:19.17 | dheckaman | does it cost money to get a ipv6 range? |
19:19.21 | jsharp | Not yet. |
19:19.24 | Yog-home | dheckaman: probably |
19:19.24 | tholo | Typically, when you sign up for IPv6, you will get a /48 prefix... |
19:19.27 | Yog-home | o really ? |
19:19.27 | Yog-home | hrm |
19:19.40 | dheckaman | for free? |
19:19.49 | dheckaman | then I can wait 20 years |
19:19.50 | Yog-home | the tunnel brokers would charge for the service |
19:19.52 | dheckaman | and sell it to MIT |
19:19.59 | tholo | Depends where you get it. freenet6 is free... |
19:20.06 | lecram | <PROTECTED> |
19:20.23 | jsharp | I wish I could figure out how to get IPV6 here at home. Damn NAT. |
19:20.55 | UnixDawg | grr my inbound is not working |
19:21.03 | UnixDawg | makes no sense |
19:21.10 | Yog-home | most stuff isn't doing v6 native anyway...so it'd wind up being xlated back into a V4 anyway |
19:21.48 | Yog-home | o hold it... /64...that'd be 64 bits of address space ? V6 is 128 bits if I remember |
19:21.53 | sxpert | Yog-home: postgresql 7.4 has native v6 support |
19:22.42 | lecram | usually from the beginning |
19:22.55 | Yog-home | yeh..that's how they do it w/ V4..i'd expect the same for V6 |
19:23.23 | dheckaman | do I have to have a public ip to use ipv6 on the internet? |
19:23.34 | Yog-home | dheckaman: probably not |
19:23.42 | Yog-home | dheckaman: just need to be able to tunnel |
19:23.54 | dheckaman | hmm |
19:23.55 | Yog-home | yeh... more than you'd ever need :) |
19:24.04 | jsharp | You need either a public IP, or be able to do "stupid routing tricks" with your NAT firewall. |
19:24.07 | dheckaman | what is the tunnel like? tcp/udp? |
19:24.16 | jsharp | GRE. |
19:24.27 | sxpert | jsharp: no nat necessary ;) |
19:24.31 | Yog-home | it's just IP in IP |
19:24.38 | dheckaman | aah |
19:24.39 | nicebub | i love conntrack |
19:24.52 | Yog-home | V6 packed gets put intisde a V4 packet and sent to the tunnel provider |
19:24.52 | Marlow | can anybody tell me, what dtmf fwd accepts ? |
19:25.05 | jsharp | Except I'm running NAT at home and that breaks the GRE encapsulation. |
19:25.24 | Yog-home | jsharp: hrm..GRE shoudl work thru NAT |
19:25.34 | Marlow | native ipv6 is neat, but not easy to get .. |
19:25.35 | dheckaman | my ipcop has the option for GRE |
19:25.41 | dheckaman | I allways wondered what that is |
19:25.47 | sxpert | RoyK: lol |
19:25.55 | sxpert | RoyK: lol |
19:26.07 | sxpert | RoyK: is definitely jealous ;) |
19:26.09 | *** topic/#asterisk by kram -> the tornado sounds like a freight train |
19:26.10 | Marlow | however .. they are down right now :( |
19:26.19 | denon | tornado? |
19:26.24 | kram | *nods* |
19:26.26 | sxpert | RoyK turned into a slapper-bot |
19:26.32 | Yog-home | <PROTECTED> |
19:26.35 | sxpert | kram: quick, hide in the basement |
19:26.37 | denon | tornados are kinda cool .. but ive been a little too close to em |
19:26.42 | denon | so I get nervous now |
19:26.50 | RoyK | kram: I think zoa has some serious problems |
19:26.57 | RoyK | kram: needs a new shrink or so |
19:27.03 | zoa | aha kraaaaam :) |
19:27.19 | denon | someone ... quick mirror CVS before digium goes to Kansas :) |
19:27.25 | Marlow | RoyK : roligt nu .. det holder han jo ikke til .. |
19:27.38 | denon | T400Ps falling from the sky .. |
19:27.40 | denon | Christmas! |
19:27.44 | sxpert | RoyK: same for you I guess |
19:27.48 | RoyK | ...svensker... |
19:27.49 | jsharp | kram's gettin the same weather system that blew through here yesterday. |
19:27.54 | Marlow | RoyK : niksen .. |
19:27.58 | Marlow | RoyK : dansker .. |
19:28.07 | RoyK | ah - ok - ikke så galt... |
19:28.14 | Marlow | RoyK : i Sverige :( |
19:28.21 | RoyK | :) |
19:28.23 | RoyK | jj |
19:28.23 | Marlow | RoyK : men ikke længe mere .. |
19:28.23 | Yog-home | o no..more swedes |
19:28.28 | Yog-home | they're everywhere :) |
19:28.37 | RoyK | Yog-home: he's danish - he just lives in .se |
19:28.45 | cypromis | shit happens |
19:28.45 | Marlow | Yog-home : nope . as i just said to RoyK ... i'm danish .. |
19:28.46 | Yog-home | ah |
19:29.03 | Marlow | Yog-home : and that i live in .se is very temporary .. |
19:29.03 | Yog-home | danish..swedish...it's liek a 10 mile drive :) |
19:29.37 | Marlow | Yog-home : nah .. more like 375 miles .. |
19:29.47 | Yog-home | heh :) |
19:29.49 | Simon_ca | i installed the festivel deb but tsstestasterisk isn't valid - archives seem to suggest a patch.. any pointers? |
19:29.53 | Marlow | Yog-home : or 60 swedish miles .. |
19:30.13 | sxpert | Marlow: what the *** is a swedish mile ? |
19:30.18 | RoyK | one 'scandinavian' mile is 10km |
19:30.36 | Marlow | yeah .. but only the swedish use that . |
19:30.37 | RoyK | we call it 'mil' - pronounced as in 'meal' |
19:30.45 | RoyK | Marlow: and norwegians |
19:30.50 | Marlow | ok så :) |
19:30.57 | RoyK | ikke dansker? |
19:30.57 | Marlow | in Denmark it's not used anymore .. |
19:31.04 | RoyK | dansker er teite |
19:31.04 | *** join/#asterisk costello (~costello@hoochie.digium.com) |
19:31.06 | Marlow | RoyK : nej ... faktisk ikke .. |
19:31.29 | Marlow | RoyK : eheh ... i don't think so .. |
19:31.29 | RoyK | :) |
19:31.44 | voidptr | sooo. |
19:31.51 | bobman | Simon_ca: I ended up getting the deb-src and patching it from the /usr/src/asterisk/festival-1.4.2.diff |
19:32.31 | Marlow | RoyK : stadig på jobbet ? |
19:32.33 | RoyK | ja |
19:32.41 | Marlow | RoyK : arghh .. |
19:32.48 | RoyK | korrekt |
19:32.56 | RoyK | men - stakk.. bbl |
19:33.09 | Marlow | RoyK : god tur |
19:35.49 | *** join/#asterisk nassy (~nassy@24-193-228-121.nyc.rr.com) |
19:36.18 | costello | Hi all. Does anyone knows how to configure SIP for using G.723? I understood for G.729 I need license if I want to use. |
19:36.43 | cypromis | yeah |
19:36.48 | cypromis | for g723.1 you need a lot of $$$$$$$$$$$ |
19:36.52 | bkw_ | JerJer wake up |
19:36.58 | bkw_ | I gots to show you something |
19:37.47 | jsharp | With SIP, do you need to send authentication information just if you're going to register or do you need to send it to place a SIP call as well? |
19:39.39 | costello | jsharp: I'm using SIP only to do a call from a registered user on another platform and I'm calling asterisk for playing an announcement. Each time only G.711 is used even I configured preferentially on the SIP user G.723 |
19:39.44 | Simon_ca | bobman: where's the patch from? |
19:39.57 | Yog-home | bah...voip-info.org DNS server isnt giving up an address for 'em |
19:40.27 | Yog-home | both DNS srevers down :( |
19:40.38 | bobman | Simon_ca: The patch comes from asterisk's source. At least, in the CVS checkout I got. |
19:41.06 | RoyK | <PROTECTED> |
19:41.38 | costello | cypromis: Can you help me to configure G.723.1??? |
19:41.52 | Yog-home | haha..voip-info.org's DNS servers are in Hawaii :) |
19:42.00 | Yog-home | someone must have snipped the line |
19:42.09 | Marlow | nope .. |
19:42.20 | Marlow | the guy that is running the site is located in Hawaii .. |
19:42.22 | *** join/#asterisk nocnoc (~cvenegas@atlas.ifxnw.cl) |
19:42.25 | Marlow | if i remember correctly |
19:42.43 | nicebub | tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 745/asterisk |
19:42.48 | nicebub | what is port 2000? |
19:42.51 | nocnoc | anybody done this? ATA ---> Asterisk ---> SIP Provider using g.729 ? |
19:42.56 | nicebub | this is from netstat -anp |
19:43.09 | Marlow | Yog-home: but it's allways slow |
19:43.19 | Yog-home | Marlow: yeh.. both name servers are the same IP too :-) |
19:43.21 | Marlow | Yog-home : :( |
19:43.22 | RoyK | PostgreSQL 7.4 Released :) |
19:43.24 | nocnoc | isnt port 2000/tcp *'s manager? |
19:43.29 | blitzrage | yo yo yo |
19:43.32 | nicebub | that's what i was thinking |
19:43.40 | nicebub | especially cause it's TCP based |
19:43.44 | Marlow | Yog-home : arghh ... i think i have to offer that guy some backup dns ... |
19:43.49 | nicebub | and I know the manager API is TCP based |
19:43.50 | bkw_ | no 2000 is chan_skinny |
19:43.54 | nicebub | ah |
19:43.54 | nicebub | ok |
19:43.56 | nicebub | tx |
19:44.00 | blitzrage | yah.. manager is like 3800 something isn't it? |
19:44.02 | bkw_ | manager.conf has the ports |
19:44.06 | nicebub | ok |
19:44.08 | Yog-home | Marlow: looks like lava net having some issues |
19:44.12 | bkw_ | port = 5038 |
19:44.13 | RoyK | 5038 is manager |
19:44.21 | blitzrage | ahh.. I knew I got 38 from somewhere ;) |
19:44.23 | Yog-home | Marlow: hehe..maybe volcano :) |
19:44.29 | nocnoc | forgive my mental diarrea :( |
19:44.32 | Marlow | Yog-home : good point :) |
19:44.34 | nicebub | what about 4569 for UDP? |
19:44.35 | blitzrage | I find voip-info.org too slow to be useable |
19:44.40 | blitzrage | that's IAX2 |
19:44.41 | bkw_ | iax2 |
19:44.43 | nicebub | udp 0 0 0.0.0.0:4569 0.0.0.0:* 745/asterisk |
19:44.44 | nicebub | ok |
19:44.46 | nicebub | thanks |
19:44.50 | pattieja | blitzrage: really? |
19:44.55 | nicebub | i've only been using sip, so tx |
19:45.08 | blitzrage | pattieja: yah.. I don't know why.. but I find it painfully slow |
19:45.08 | pattieja | blitzrage: I saw some useful information on it |
19:45.18 | blitzrage | pattieja: oh, I find it USEFUL, just very slow. |
19:45.19 | pattieja | blitzrage: well, it is slow, no doubt about that |
19:45.30 | blitzrage | pattieja: I wasn't speaking of the content at all :) |
19:45.37 | pattieja | ok |
19:45.48 | pattieja | blitzrage: did you see the latest entries about echo cancellation? |
19:45.59 | Yog-home | so anyone here using SER as a main SIP proxy and * as just a voice mail system ? |
19:46.03 | bkw_ | netstat -na is faster |
19:46.10 | blitzrage | pattieja: not yet. i haven't read much on the wiki in a bit. |
19:46.15 | pattieja | I'm wondering if our problems stem from the fact that we have an analog 2-line phone plugged into the line that goes into the X101P |
19:46.35 | bkw_ | pattieja could be |
19:46.37 | blitzrage | pattieja: unfortunatly, I work 45 hours a week at a place an hour away from me.. so my 40 hours a week turn into 65, and I don't have much time to play lately. |
19:46.38 | Yog-home | yen...the "-p" option is great tho..shows the program using the port |
19:47.17 | pattieja | that line is split at a box where it picks up another line to go to the 2-line phone and the other goes to the X101P |
19:47.39 | pattieja | bkw_: blitzrage: I don't think we've ever used the system without the phone being plugged in at the same time |
19:47.45 | *** join/#asterisk dobey (~dobey@gateway.ximian.com) |
19:47.47 | pattieja | the phone is not off-hook, mind you, either |
19:48.10 | pattieja | but it is in the circuit, and it's a 900MHz cordless phone, too |
19:48.33 | dobey | are there any API docs for libiax and possibly a specification for the protocol? |
19:48.37 | blitzrage | pattieja: couldn't you unplug the phone and see if you still have the problem? (I'm not totally sure what is going on, as I just walked in the door) |
19:48.58 | pattieja | which puts pulsing noises into the earpiece when both lines are in use simultaneously. It's apparently picking up noise from the LEDs that flash to indicate the lines are in use |
19:49.17 | voidptr | someone tell me what matt's astgui is really aimed for ;) |
19:49.28 | pattieja | blitzrage: well, currently, we've been using it to test the system. We call from the first line to the second line's number (which is connected to the X101P) |
19:49.42 | blitzrage | ahhh I see |
19:49.44 | pattieja | but I will attempt to try it without it |
19:49.52 | Yog-home | anyone here using * in an enterprise VOIP environment ? like 500 or more ppl ? |
19:50.00 | blitzrage | yah. I don't have an X101P, so I have very little knowledge of it :) |
19:50.08 | pattieja | blitzrage: and I've been going nuts trying to get the echo worked out of the system |
19:50.08 | blitzrage | just the TDM400P on loan from the school |
19:50.36 | pattieja | blitzrage: any POTS line cards? (the X101P is just a newer model than the X100P) |
19:50.53 | bkw_ | not really |
19:51.04 | blitzrage | pattieja: nope, I haven't used either the X100P or 101P. |
19:51.10 | pattieja | I can never remember stuff like this, but the X10[01]P is signalled with FXO, so I think that means it's an FXS device |
19:51.15 | pattieja | blitzrage: ok |
19:51.15 | blitzrage | pattieja: I only use Cell and VoIP, no landline stuff. |
19:51.22 | voidptr | whatever it is, doesn't look like its very usable... neither operator console, nor softphone |
19:51.35 | blitzrage | pattieja: the X100P is an FXO device with FXS signalling. |
19:51.37 | bkw_ | no its signaled with FXS |
19:51.42 | pattieja | blitzrage: we're trying to use only softphones (VoIP) in our scenario too, but connected to a single land line |
19:51.56 | blitzrage | pattieja: I also don't use softphones :) (but I have) |
19:52.03 | blitzrage | pattieja: gotcha |
19:52.05 | pattieja | ahh. ok. see I told you I can't remember these things |
19:52.17 | blitzrage | pattieja: yah... always the opposite signalling |
19:52.19 | pattieja | bkw_: thanks |
19:52.50 | blitzrage | pattieja: yah I know.. it's a bit confusing at first until you memorize it pretty much |
19:52.56 | pattieja | blitzrage: yeah, I've tried a smattering of X-Lite, DIAX, iaxcomm/client (going to try the latest version here soon), gnophone, linphone, kphone, etc.) |
19:53.25 | pattieja | I even was able to get X-Lite to run under Wine on Linux |
19:53.28 | blitzrage | pattieja: I've pretty much only used x-lite/pro. I like it, but find the DTMF support a bit lacking sometimes |
19:53.40 | blitzrage | pattieja: yah, I've heard it runs under Wine, but never tried so far. |
19:53.54 | pattieja | blitzrage: really? it seems that DTMF works the best for me on X-Lite |
19:54.54 | pattieja | blitzrage: you have to have a beefy Linux box, I think, to make it sound normal. If you have a lower powered machine, it sounds garbled and pulsating, but still understandable. I haven't tried it on anything faster than the laptop I use which is a 600MHz PIII |
19:55.12 | bkw_ | MAN the wind is blowing |
19:55.34 | blitzrage | pattieja: yah, I have it on an AMD 1800. I am probably using an older version now, and more than likely was just the way I had my * box configured :) |
19:55.50 | blitzrage | pattieja: but I can use the DTMF now with x-lite and my menu system |
19:55.53 | pattieja | but, if I get iaxclient/comm to work, that will be really sweet. The main reason is that X-Lite isn't a true Linux app, but it's the only app that works properly with the USB headset I had to get |
19:56.06 | pattieja | cool |
19:56.26 | blitzrage | pattieja: gotcha. Yah, I think I'm going to try an iax client one of these days. More than likely will use more of that stuff on my laptop once I get back to school |
19:56.40 | pattieja | bkw_: went to bed last night with it raining pretty badly. Woke up this morning to reports that the basement had at least 2 inches of water |
19:57.09 | jsharp | Surfs up. |
19:57.12 | pattieja | blitzrage: DIAX for Windows looks very promising and is already at a very useable stage of development |
19:57.14 | pattieja | jsharp: :) |
19:57.37 | blitzrage | pattieja: yah, I've been hearing good things about it. I'll give it a shot at some point I'm sure. |
19:58.02 | pattieja | although, it still crashes every time I exit the application |
19:58.02 | blitzrage | WORK! |
19:58.08 | pattieja | oh, sorry |
19:58.13 | pattieja | me too |
19:59.19 | dobey | hrmm |
20:01.26 | voidptr | i dont care what soft phone works, as long as it works and uses (iax|iax2) |
20:04.11 | *** join/#asterisk joe_satriani (~joe_satri@ppp-10-156.28-151.libero.it) |
20:04.14 | joe_satriani | hi all |
20:04.28 | *** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-153-89.NYCMNY83.dynamic.covad.net) |
20:09.16 | *** join/#asterisk jtodd (~jtodd@207.141.153.205) |
20:09.20 | *** part/#asterisk ajh (~ajh@c-24-0-23-89.client.comcast.net) |
20:10.13 | jtodd | I can't reach my test machines. Could someone please cut/paste the line(s) that show up in dmesg upon boot when a TE410P is present? |
20:10.32 | joe_satriani | Is anybody who used auto call ? |
20:10.42 | jtodd | (i.e.: I wouldn't have to ask this if I could see my normal boxes, but I'm working in an unexpectedly odd network environment at the moment) |
20:10.56 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
20:11.57 | jsharp | I get this when I modprobe t4xxp |
20:11.58 | jsharp | Zapata Telephony Interface Registered on major 196 |
20:11.58 | jsharp | Found TE410P at base address fc000000, remapped to e087a000 |
20:11.58 | jsharp | TE410P version c01a003a |
20:11.58 | jsharp | FALC version: 00000005, Board ID: 00 |
20:13.22 | jtodd | jsharp: Thanks. Do you see anything in dmesg during the PCI probes right at boot (before modprobes have loaded the zap drivers)? |
20:13.58 | jsharp | Nope. Nothing shows up about the card till you load the drivers for it. |
20:14.21 | Yog-home | SIGH -> Look for the condition syntax in examples or documentation. |
20:14.21 | Yog-home | {root@fs/pts/7}/usr/local/asterisk/etc# egrep -i gotoif * |
20:14.21 | Yog-home | {root@fs/pts/7}/usr/local/asterisk/etc# |
20:15.17 | jtodd | Hm... OK, I'm used to the T1 cards which I thought I remembered saying "Found a Wildcard" in the dmesg during boot, but apparently I'm confusing that with the modprobe output. |
20:16.44 | jsharp | "Found a Wildcard" shows up in dmesg when you modprobe it...be it upon boot or at any other time. |
20:18.06 | *** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-24-2.NYCMNY83.dynamic.covad.net) |
20:19.54 | pattieja | :( |
20:20.26 | pattieja | iaxcomm works as a client, even seems to have some very nice features, but it still doesn't work properly with my USB headset. It seems that only X-Lite under Wine works properly. |
20:20.39 | Connor- | what is expat ?? I'm trying to compile oh323 |
20:20.53 | pattieja | Connor-: sounds like something to do maybe with XML? |
20:21.32 | jsharp | Bleh. This PAM stuff is turning out to be much harder than I thought. |
20:21.48 | atacomm | does * have a problem with knowing when SIP phones are online? half the time it is saying busy when i dial into Asterisk..... |
20:21.59 | Connor- | yea, just found it.. dependancies.. |
20:24.44 | Connor- | Oh frill.. |
20:24.46 | Connor- | [chan_h323.so]WARNING[1074395872]: File loader.c, Line 232 (ast_load_resource): libpt_linux_x86_r.so.1: cannot open shared object file: No such file or directory |
20:24.46 | Connor- | WARNING[1074395872]: File loader.c, Line 400 (load_modules): Loading module chan_h323.so failed! |
20:24.46 | Connor- | Ouch ... error while writing audio data: : Broken pipe |
20:25.10 | jsharp | Make sure that library exists somwhere that ld.so.conf can pick it up. |
20:25.15 | jsharp | Then rerun ldconfig. |
20:26.52 | Connor- | where should that lib be? |
20:27.07 | jsharp | Mines in /usr/local/lib |
20:27.24 | jsharp | That's where I put it. |
20:29.36 | Connor- | what package does libpt_linux_x86_r.so.1 come from.. I can't find it.. |
20:30.09 | bkw_ | um |
20:30.14 | bkw_ | dude chan_h323? |
20:30.19 | Connor- | never mind.. openh323 |
20:30.21 | bkw_ | you must not be following direction? |
20:30.25 | jsharp | it comes from oh323. |
20:30.29 | jsharp | Er, openh323. |
20:30.31 | Connor- | I guess I have to manully move it. |
20:30.38 | bkw_ | no |
20:30.42 | bkw_ | you didn't read the readme :P |
20:30.45 | bkw_ | haha |
20:30.49 | Connor- | yes I did. |
20:30.53 | bkw_ | no you didn't |
20:31.01 | bkw_ | you don't move it.. you set LD_LIBRARY_PATH |
20:31.10 | bkw_ | granted you can move it.. but thats not what the README says |
20:32.00 | Connor- | well.. It's sitting in /usr/src/ |
20:32.09 | Connor- | well.. It's sitting in /usr/src/openh323/lib |
20:32.18 | Connor- | I don't run ANY lib from /usr/src |
20:32.20 | *** join/#asterisk cfo (~cfo@194.19.190.217) |
20:32.29 | bkw_ | I don't run h323 on my box anymore :P |
20:32.36 | bkw_ | One Number Find Me .. now thats an app I need to write |
20:32.40 | *** join/#asterisk Mystique (~bcook@mystique.poklib.org) |
20:33.23 | bkw_ | dobey what use? |
20:34.02 | bkw_ | what do you want the information for? |
20:34.08 | bkw_ | or how do you wanna use it |
20:34.40 | *** join/#asterisk raha (~raha@hoochie.digium.com) |
20:35.37 | anthm | echo /usr/src/openh323/lib >> /etc/ld.so.conf ; echo /usr/src/pwlib/lib >> /etc/ld.so.conf ; ln -s /usr/src/openh323 /usr/src/pwlib /root ; rm -fr /usr/src/openh323 /usr/src/pwlib |
20:35.57 | joe_satriani | Is anybody who used auto call ? |
20:36.05 | bkw_ | auto call? |
20:36.15 | bkw_ | oh god you people are about as responsive as a rock |
20:36.26 | anthm | 1 command is not part of the right way but more like what i want to do w h323 |
20:36.30 | *** join/#asterisk ktracho (~ktracho@hoochie.digium.com) |
20:36.49 | joe_satriani | yes, if I put a script in /var/spool/asterisk/outgoing it can call a number and if it answer forward to an internal number |
20:37.01 | bkw_ | easy sample.call |
20:37.21 | gadams666 | anyone have mysql integrated for cdr? |
20:37.26 | *** join/#asterisk ktracho (~ktracho@hoochie.digium.com) |
20:37.34 | dobey | bkw_: i want to notify the user of incoming calls (callerid), and voice mails, and maybe provide a simple app for playing the voicemails |
20:37.37 | bkw_ | gadams666 cdr_mysql works wondrs |
20:37.37 | joe_satriani | bkw_: where i can find that file |
20:37.46 | bkw_ | dobey it will email the thing |
20:37.56 | bkw_ | dobey the email can include all that.. and the message too |
20:37.59 | Mike | never sleepet so much |
20:38.06 | bkw_ | cvs checkout asterisk-addons |
20:38.13 | gadams666 | that's what i'm comnfiguring right now. have the db creatd, do I need to load a module in modules.conf? |
20:38.25 | bkw_ | yes |
20:38.26 | gadams666 | got it, compiled it, installed it. :) |
20:38.39 | dobey | bkw_: i don't want e-mail, i want to do this as a sort-of client |
20:38.58 | bkw_ | dobey write it.. thats about all you can do |
20:39.11 | bkw_ | whats so wrong with the MWI on the phone? |
20:39.21 | dobey | bkw_: i want to write it |
20:39.23 | gadams666 | basically a load=cdr_addon_mysql.so |
20:39.28 | bkw_ | gadams666 yes |
20:39.42 | dobey | bkw_: i'm trying to ask aobut the information i can get with the protocols available for contacting the server |
20:39.44 | Mystique | hey all, is anyone around that can answer a few basic questions for me? |
20:39.46 | joe_satriani | bkw_: wich is the cvs parameter to obtain the list of module-addon ? |
20:39.52 | dobey | looking in iax-client.h doesn't tell me much of anything |
20:39.59 | bkw_ | joe_satriani their are only two |
20:40.00 | *** join/#asterisk Shido6 (~shido@d57-81-103.home.cgocable.net) |
20:40.08 | bkw_ | they are in the same directory |
20:40.15 | joe_satriani | on is mysql_cdr and the other? |
20:40.15 | Mike | bkw_: care to help me a bit with iax? |
20:40.38 | gadams666 | getting closer, "failed connect" - betting mysqld.sock not located in /tmp :) |
20:40.52 | *** join/#asterisk bobman (~bobman@mube.psouth.net) |
20:41.30 | ktracho | anyone knows about a full SIP SOFTPHONE for Windoze with HOLD & CONFERENCE features? |
20:42.04 | wasim | DEBUG[131081]: File chan_iax.c, Line 3553 (socket_read): Ooh, voice format changed to GSM |
20:42.25 | wasim | <PROTECTED> |
20:42.29 | *** join/#asterisk lexluther (~admin@80.88.142.221) |
20:42.29 | darius | ktracho: just sjphone have those features? I don't recall |
20:42.30 | atacomm | bkw: are there any known problems with SIP and Asterisk where asterisk doesnt even try the phone and returns a busy? I'm just trying to figure out if it is Asterisk, or if its the X-Lite softphone....want to make sure there aren't any issues before launching |
20:42.48 | *** part/#asterisk lexluther (~admin@80.88.142.221) |
20:42.52 | bkw_ | atacomm yes.. when you seed from the astdb |
20:43.20 | atacomm | bkw_: what? |
20:43.30 | wasim | Asterlogy: -r |
20:44.47 | ktracho | I'm gonna checkout the SJPHONE! Thx! |
20:48.07 | Connor- | okay, that's waked.. I got h323 working.. half way.. |
20:48.25 | Connor- | I can call.. it connects, and works, but, I get disconnected in the middle of a call. |
20:48.29 | atacomm | bkw: what do you mean? |
20:51.14 | bkw_ | do this |
20:51.20 | bkw_ | database deltree SIP |
20:51.21 | bkw_ | restart * |
20:51.45 | bkw_ | the device hasn't re-registered since restart |
20:52.31 | atacomm | hmm....so its more a question of make the phone register more often? |
20:52.45 | bkw_ | yes |
20:52.57 | bkw_ | i have the same problem with my 7960 |
20:53.18 | bkw_ | when you seed the sip registrations from astdb on start.. and the phone hasn't re-registered yet.. it will come up busy |
20:53.37 | *** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net) |
20:53.45 | atacomm | what a pain |
20:55.09 | atacomm | is that were default ip comes in handy? |
20:55.29 | Yog-home | does * run chrooted ? |
20:55.49 | Yog-home | or does the System command perhaps ? |
20:56.28 | ManxPower | Yog-home: No and No, but I'm sure you can |
20:56.33 | Yog-home | I run a System() command like "System(/bin/echo -e a bunch of text > /tmp/blah) and /tmp/blah is never created |
20:56.48 | *** join/#asterisk WOB (~WOB@ndn-83-105.telkomadsl.co.za) |
20:57.18 | WOB | hey all |
20:57.32 | ManxPower | Yog-home: Does the *exact same command run at the command line? |
20:57.47 | ManxPower | For example quotes can be an issue |
20:57.52 | WOB | anyone know what would be causeing the problem if I try and compile the chan_capi 0.3.0 drivers and it brings up a [chan_capi.o] Error 1 |
20:58.02 | WOB | Any help would be great! |
20:58.15 | Yog-home | shoudln't matter in this case |
20:58.24 | rajo_home_ | jtodd: what's the current state of TRIP with asterisk? |
20:58.26 | Simon_ca | is it possible to set a var from a system command, ie something like exten => 1234,1,Setvar($DATE(System(date)) |
20:58.44 | n3wb33 | WOB: paste a bit of the error, and make sure you have i4l-devel installed ... or whatever that is called in your distro |
20:58.56 | Yog-home | ah..i put quotes around the text and it worked |
20:58.57 | Yog-home | :( |
20:59.28 | Yog-home | ugh..doesn't solve my problem though.. I'm looking for some variable I can access that references the SIP "To:" field |
20:59.34 | Yog-home | I guess that's not available..ohwell |
21:00.37 | Yog-home | letsee if anyone else has suggestions... |
21:00.46 | rajo_home_ | ~seen jtodd |
21:00.48 | | jtodd is currently on #asterisk (51m 32s). Has said a total of 4 messages. Is idling for 45m 31s |
21:00.59 | jtodd | what what? |
21:01.06 | rajo_home_ | :) |
21:01.11 | jtodd | rajo_home: Nothing. There is currently no progress. |
21:01.13 | WOB | n3wb33: Im running redhat 9 .. what is the package called for that? |
21:01.32 | *** join/#asterisk boropest (~boropest@mail.mkda.com) |
21:01.43 | rajo_home_ | jtodd: ah okay... pity, |
21:01.44 | jtodd | rajo_home: Wish there was. I have a spark of interest from the Austrian guys doing ENUM, which I think would be cool if they put some funding and manpower behind it. |
21:01.54 | n3wb33 | WOB: i dont know nothing about red hat. |
21:02.50 | rajo_home_ | jtodd: yep. we're planning doing enum with * too, but for an interconnection of several * boxes on the universities here TRIP would be handy |
21:03.51 | Yog-home | I have a customer who wants to use * for voicemail... unfortunately, all his phones are set up using SIP "user@domain" style addresses instead of "extension@sipdomain" ... he's using SER as his SIP router/proxy... the # extensions in his dialplan are simply aliases under SER which map to the "sip:user@domain" style addresses...when I shunt a call off to * for voicemail, asterisk doesn't see the dialed extension, but isntead of " |
21:03.51 | Yog-home | IP invite as the dialed extension...so I either have to include a bunch of usernames as extensions, or somehow get the dialed extension # to map to a * voicemail box...any suggestions ? |
21:03.53 | jtodd | rajo_home: Imagine interconnection between twelve different providers, each with daily (hourly!) changing rates and costs. Without TRIP, it sucks. |
21:04.23 | *** join/#asterisk bobman (~bobman@mube.psouth.net) [NETSPLIT VICTIM] |
21:04.31 | jtodd | rajo_home: ENUM is probably sufficient if you don't have a bunch of gateways. If you're just trying to find end users, ENUM is all you need. |
21:04.51 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) [NETSPLIT VICTIM] |
21:04.57 | *** join/#asterisk boropest (~boropest@mail.mkda.com) [NETSPLIT VICTIM] |
21:05.04 | *** join/#asterisk Shido6 (~shido@d57-81-103.home.cgocable.net) [NETSPLIT VICTIM] |
21:05.40 | rajo_home_ | jtodd: yes, we want to use enum for kind of LCR via voip from the pstn at the university, and having trip to do the connections to other universities here |
21:06.28 | Yog-home | guess not |
21:06.32 | jtodd | rajo_home: ENUM is not designed for LCR, and I'm not quite sure how you'd implement that. ENUM gives you 1 answer, and 1 answer only. TRIP can give you fifteen different answers, and then costs, and all kinds of other decision criteria, and lets you make the choice yourself. |
21:07.08 | *** join/#asterisk anachron (~fradooj@adsl-64-169-89-27.dsl.snfc21.pacbell.net) |
21:07.21 | n3wb33 | ~seen kapejod |
21:07.22 | | kapejod <~kapejod@pD9E82713.dip.t-dialin.net> was last seen on IRC in channel #asterisk, 19h 12m 1s ago, saying: 'a bit slow for a phone ;)'. |
21:07.49 | *** join/#asterisk lexluther (~admin@80.88.142.221) |
21:07.58 | Connor- | Why would h323 be terminating the call prematurely? |
21:08.02 | rajo_home_ | jtodd: well, currently there's kind of an ENUM-hype here in house... and as this is a good way to install some *-boxes, we've taken the chance |
21:08.05 | *** part/#asterisk lexluther (~admin@80.88.142.221) |
21:09.55 | rajo_home_ | jtodd: obviously ENUM is a bit limited to do so, but it is a beginning. So TRIP is not ready yet... hmm... |
21:10.17 | citats | jtodd: enum can do LCR assuming your DNS server handles it... quite simple actually... either build wildcard routes or do what i did and write a module for powerdns to generate records on the fly |
21:10.28 | jtodd | rajo_home: TRIP isn't merely not ready - it doesn't exist. There is a reference implementation from the Vocal project, but I've been unable to even get it to compile. |
21:10.47 | jtodd | citats: You're not using ENUM, you're using some back end engine that happens to answer via DNS queries. :-) |
21:11.09 | jtodd | citats: and wildcard routes aren't what I'd call LCR.... |
21:11.13 | citats | jtodd: i think thats what it says in websters when you lookup enum :) |
21:11.17 | rajo_home_ | jtodd: ...that doesn't surprise me... compiling vocal stuff is quite... well... challenging sometimes |
21:11.42 | Connor- | Who's doing h323 ? |
21:11.42 | citats | jtodd: if you have multiple gateways in different cities and you setup wildcard routes pointing appropriate prefixes to those gateways then that very much is LCR |
21:12.14 | jtodd | citats: Er... no, I think that it's not quite that simple. ENUM assumes that you can use the DNS, and a query from any location will get the same response, using the "standard" DNS caching mechanisms. Your system is just a back end that uses an ENUM answer for what normally is a static answer. |
21:12.58 | citats | jtodd: it doesn't assume that actually... i generate different responses based on the source, timeofday, current call load on gateways, etc. |
21:13.23 | jtodd | citats: Right. That's not "pure" ENUM; that's an LCR engine that happens to answer via ENUM, as I said. :-) |
21:14.44 | *** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-24-2.NYCMNY83.dynamic.covad.net) [NETSPLIT VICTIM] |
21:14.46 | citats | jtodd: ENUM is far from pure no matter how you look at it :) |
21:15.35 | citats | the LCR engine can be software, or a human looking at a phone book and typing in a few wildcard records |
21:16.39 | iGN | Are there any good newsgroups on CTI in general that you'd like to recommend? |
21:18.34 | tzanger | what is the difference between ANI and CID? |
21:19.31 | tholo | CID can be turned off. |
21:19.47 | tholo | ANI is not supposed to be provided to an end user. |
21:20.36 | *** join/#asterisk joe_satriani (~joe_satri@ppp-10-156.28-151.libero.it) [NETSPLIT VICTIM] |
21:20.58 | denon | tholo: aren't PRIs really end-users, by the telco's point of view? |
21:21.22 | tholo | Yeah, and not all telcos will provide ANI on a PRI... |
21:21.24 | Connor- | ~seen jerjer |
21:21.25 | | jerjer is currently on #asterisk |
21:21.42 | denon | tholo: but if some do .. seems like it defeats the point :) |
21:22.12 | tholo | In some cases (like when you run an 800 service) you can subscribe to it. |
21:22.22 | tholo | I.e. when you need it for billing purposes. |
21:22.29 | denon | nod |
21:23.08 | tholo | In other situations, if you receive it, you are still not supposed to put it up in the CID data on a desk phone. Of course, you *can* do it... ;-) |
21:24.11 | denon | buggy script :) |
21:24.19 | denon | sent it to caller id instead of billing .. will have to debug that later ;) |
21:24.24 | tholo | Honest offiser, it was just a bug! |
21:24.29 | tholo | Officer, too. |
21:24.44 | denon | ossifer? |
21:24.55 | tholo | Hm, I like that version better.... 8-) |
21:25.07 | denon | drunken assmaster |
21:25.14 | Yog-home | hrm... in * voicemail, is the 'greet.gsm' file ever played ? I've tried calling voicemail w/o the prefix "u" or "b" .. perhaps greet.gsm is just obsolete and the "unavail.gsm" file is used as the 'greeting' ? |
21:28.37 | Mike | someone can give me an example how to link with iax 2 * servers |
21:28.55 | Mike | using gsm between them |
21:33.07 | Yog-home | hrm..bizzarre .. I see code in app_voicemail to play greet file, but it never plays it :( |
21:33.25 | *** join/#asterisk stonefly (~trillian@toby.stoneflytech.com) |
21:37.25 | *** part/#asterisk rajo_home_ (~rainer@p508AEFB0.dip.t-dialin.net) |
21:38.06 | *** join/#asterisk terry__ (~terry@adsl-68-89-230-153.dsl.spfdmo.swbell.net) |
21:38.38 | Yog-home | hm...this was working before.... <sigh> |
21:39.26 | terry__ | Does anyone have a recommendation for a good (and relatively inexpensive) 4-port FXO gateway? |
21:39.49 | dheckaman | kmart has some good ones |
21:40.06 | dheckaman | they even have ones with cream filling |
21:40.19 | terry__ | so helpful. :) |
21:40.23 | dheckaman | :) |
21:40.54 | terry__ | Does the cream filling cost extra? I'm on a budget. |
21:41.03 | dheckaman | hmm |
21:41.20 | *** join/#asterisk nocnoc (~cvenegas@atlas.ifxnw.cl) |
21:41.22 | dheckaman | I think so, its the special, fluffy cream filling... |
21:41.31 | terry__ | I tried Digium, but their 4-port isn't ready yet and they don't have a release date for me... |
21:41.33 | nocnoc | is there such thing as a sccp channel? |
21:41.54 | tholo | chan_skinny |
21:44.15 | Simon_ca | how do you route based on callerid? |
21:45.04 | *** join/#asterisk bobman (~bobman@mube.psouth.net) |
21:45.12 | nocnoc | tholo - is that available on the latest CVS? |
21:45.28 | Mike | gui for asterisk? |
21:45.42 | tholo | Yes. It is not really production ready, however. Apparently the funding for it went away. |
21:45.52 | *** join/#asterisk jgaviria (~jgaviria@63.245.86.109) |
21:45.55 | tholo | Simon_ca: search for "anti-GF" |
21:46.19 | Connor- | what's the dial string for oh323 someone? |
21:46.31 | tholo | I believe it is even in the handbook... |
21:46.36 | carrar | it is |
21:46.38 | carrar | section 4.4.3 |
21:46.43 | carrar | 4.3.3 |
21:46.45 | carrar | sorry |
21:47.02 | anthm | Dial(RTFM/${EXTEN}) |
21:48.31 | Stealth_Man | is there any special settings in sip.conf for GS phone to be registered ? |
21:49.16 | tzanger | wow a lot's changed in CVS |
21:50.54 | bkw_ | blah blah blah |
21:51.29 | dheckaman | hmm |
21:51.35 | dheckaman | echo on the sip side |
21:51.45 | dheckaman | but no echo on the POTS side |
21:51.56 | *** join/#asterisk zwi (~chris@216.88.131.43) |
21:52.05 | Stealth_Man | wtf wrong with gs and * shit... it is not allowing to me to register GS into * |
21:52.07 | terry__ | simon_ca: just in case you actually wanted an answer... route by callerid ... exten => 100/5555551212,1,Congestion would route a call to extension 100 from 5555551212 to the congetstion app |
21:52.07 | Simon_ca | ok, this one isn;t in the handbook. :) What do you pass to Festival() to get it to readback the callerid, i.e. like an ani readback service. ${callerid} doesn't work. i guess another way of asking is how do you get the callerid into a var.... |
21:52.35 | citats | Simon_ca: try ${CALLERID} or ${CALLERIDNUM} special variables are case sensitive |
21:53.01 | anthm | hmm maybe wtfm |
21:53.02 | tholo | And check README.variables in the source directory |
21:53.03 | Connor- | I'm having oh323.. someone help? |
21:53.11 | anthm | write the f^&%^& manual |
21:53.16 | *** join/#asterisk GhostNr1 (~Ashmed@193.10.185.3) |
21:53.59 | voidptr | wafm, write a fine manual |
21:54.15 | *** join/#asterisk pattieja (~pattieja@63-252-5-1.ip.mcleodusa.net) |
21:54.19 | Mike | if my remote server has more bandwidth should i put the iax server over there? |
21:54.27 | n3wb33 | cat *.c |
21:54.51 | dheckaman | cat /dev/urandom > /etc/asterisk/extentions.conf |
21:54.57 | dheckaman | :) |
21:55.10 | sxpert | dheckaman: lol |
21:55.13 | terry__ | in case anyone new has logged in... Anyone have any experience with any particular 4-port fxo gateways for use with asterisk? |
21:55.29 | tholo | cat /dev/urandom > /dev/hda |
21:56.00 | dheckaman | yea |
21:57.15 | Simon_ca | citats: thanks. I tried ${CALLERID} and doesn;t work. just tried ${CALLERIDNUM} and it does. (as does ${CALLERIDNAME) but ${CALLERID} itself appears undefined... |
21:57.38 | Simon_ca | so calleridnum will work fine, but curious |
21:58.17 | tzanger | citats: you're there! |
21:58.37 | tzanger | malloc: dispose_cmd.c:249: assertion botched |
21:58.37 | tzanger | free: called with unallocated block argument |
21:58.41 | tzanger | that doesn't sound good :-) |
21:59.09 | tzanger | malloc: stringvec.c:73: assertion botched |
21:59.11 | tzanger | nor that |
21:59.27 | Stealth_Man | bkw: can you look into my sip.conf please ??? |
21:59.27 | Connor- | Can someone help me for just a minute? Trying to get oh323 working.. Please?? |
21:59.44 | sxpert | Connor-: good luck ;) |
22:00.56 | Connor- | sxpert: Got it compiled and loading... |
22:04.50 | UnixDawg | I dont get why my incoming is not working |
22:05.48 | UnixDawg | Host Username Perceived Refresh State |
22:05.48 | UnixDawg | 66.234.228.132:4569 in-PLy97oR 192.168.0.2:60621 60 Registered |
22:06.00 | UnixDawg | why is it not putting it on normal ports |
22:06.37 | Simon_ca | anyone know of a way to set a variable with the output of a system call? ie Setvar(TEMP,System(wget http://localhost/cgi-bin/weather)) - I want pass it to festival... |
22:07.32 | Connor- | Well.. I got it to call netmeeting.. |
22:07.43 | Connor- | How do I get it to work with CCM ? |
22:08.31 | UnixDawg | something is wrong here |
22:08.33 | bkw_ | is it me or has all hell broke loose today |
22:09.16 | bkw_ | Connor you want chan_h323 to work with CCM? |
22:09.29 | twilson | Is there any advantage to using skinny on a Cisco 7960 as opposed to SIP when using asterisk? |
22:09.31 | Connor- | no. chan_oh323 |
22:09.37 | tholo | No, no, he is using the*other* H.323 implementation! |
22:09.42 | bkw_ | twilson not yet |
22:09.48 | bkw_ | chan_skinny needs alot of love |
22:09.53 | Connor- | chan_h3h don't work with ccm |
22:09.56 | bkw_ | nope |
22:09.58 | bkw_ | was aobut to say that |
22:10.08 | bkw_ | the rtp stream changes once CCM sets the call up |
22:10.17 | Connor- | right, and JerJer won't fix it. : |
22:10.19 | Connor- | :) |
22:10.22 | twilson | so far I don't have a lot of love for chan_skinny... :-) |
22:10.29 | bkw_ | Connor well you could fix it |
22:10.40 | bkw_ | JerJer has bigger issues to deal with |
22:10.42 | Connor- | Don't know a thing about h323 stuff. |
22:11.48 | anthm | chan_?h323 + netmeeting has 1% chance of working if you manually set the codec in both to ulaw |
22:12.01 | Connor- | Well.. oh323 works with netmeeting.. wonder why it's not working with the CCM |
22:12.13 | anthm | h323 and netmeeting both belong together at the bottom of the ocean floor |
22:12.25 | twilson | I have four analog lines that I need to hook up to an asterisk box... I don't have room for 4 FXO cards, and a t1 card/channel bank seems a little excessive. Anyone have any recommendations? |
22:12.31 | anthm | consider them both voip roadblocks |
22:12.32 | Connor- | I don't care about netmeeting.. just trying to get it to work with CCM |
22:12.42 | bkw_ | CCM is shit |
22:12.45 | bkw_ | pure shit |
22:12.57 | bkw_ | poop |
22:13.14 | Connor- | come on guys.. I'm not here to debackle or argue of what's hot and what's not.. I just need to get the shit to work. |
22:13.37 | anthm | helping to get it to work prolongs its lifespan , no ? |
22:13.51 | bkw_ | HAHA |
22:13.54 | bkw_ | ~google bkw asterisk |
22:15.12 | Connor- | can't get debugging to work for oh323 |
22:15.51 | anthm | what does CCM stand for? |
22:15.59 | jets | cisco callmanager |
22:15.59 | Connor- | Cisco Call Manager |
22:16.41 | anthm | most h323 issues come from codec mismatch and nat |
22:18.00 | anthm | everyting seems to speek ulaw so you always get that working 1st |
22:18.11 | anthm | byt making it the only codec in the list |
22:20.59 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
22:22.21 | *** join/#asterisk RickColl (~RickColl@Toronto-HSE-ppp3832260.sympatico.ca) |
22:24.15 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
22:25.11 | bkw_ | http://www.tric.nl/~cg/asterisk.html |
22:25.18 | bkw_ | HOLY SHIT.. almost 30k lines |
22:25.19 | bkw_ | haha |
22:25.49 | tholo | Yeah -- you just keep on talking, and talking, and talking, and... |
22:26.21 | Mike | mike (1593) |
22:26.24 | Exomorph | He also splits his lines up alot. :) |
22:26.25 | Mike | didnt make it at the top |
22:26.25 | Mike | lol |
22:26.33 | tholo | Yeah, |
22:26.34 | tholo | that |
22:26.35 | tholo | is |
22:26.36 | tholo | one |
22:26.36 | Mike | im an amateur |
22:26.37 | tholo | way |
22:26.43 | tholo | of getting lots of |
22:26.44 | tholo | entries |
22:26.45 | Mike | i think i have to type even more |
22:26.46 | Mike | :P |
22:27.18 | Stealth_Man | ---- >kram is either insane or just a fair op, kicking a total of 3 people! |
22:27.20 | Stealth_Man | hehe |
22:27.21 | tholo | Me, I prefer to type sensible stuff elsewhere -- typically code... |
22:27.31 | Mike | bkw_: thats because i havent been here the 74 days next time ill be on high score |
22:27.42 | bkw_ | NEVER |
22:27.53 | pbxtech | mike who? |
22:27.55 | pbxtech | :) |
22:27.56 | bkw_ | Nobody will be able to top me.. I spent 18 hours + per day in front of this computer |
22:27.57 | bkw_ | JUST TRY ME! |
22:28.00 | Exomorph | LOL Take a look at this... (from the page bwk sent) |
22:28.00 | Exomorph | Poor bkw_, nobody likes him/her. He/She was attacked 27 times. |
22:28.15 | bkw_ | pbxtech how does it taste? |
22:28.17 | bkw_ | :P |
22:28.20 | pbxtech | :P |
22:28.44 | bkw_ | Stealth_Man what were you jumping up and down about ? |
22:28.47 | bkw_ | some sip.conf? |
22:28.56 | bkw_ | put it on the web.. msg me the URL if you want me to look at it |
22:28.59 | Stealth_Man | :)))))))) |
22:29.13 | Stealth_Man | bkw : yes i will |
22:29.22 | Stealth_Man | becausei have stupied erorrs i can;t figure out anything :( shitt |
22:29.30 | Mike | bkw_: they sell lifes for about 3.99 on any pharmacy:P on packages of 3 for use with girls |
22:29.31 | Mike | :P |
22:30.39 | Takapa | what time is it at digium's now? |
22:30.48 | bkw_ | its 4:30 pm @ digium |
22:30.49 | bkw_ | isn't it? |
22:30.54 | bkw_ | are they CST? |
22:30.54 | Takapa | hm.. k |
22:31.00 | bkw_ | or EST |
22:31.24 | *** join/#asterisk SirGaia (~ircap75@217.Red-80-37-235.pooles.rima-tde.net) |
22:31.32 | malcolmd | CST |
22:31.46 | SirGaia | Hi all |
22:31.57 | coolhp | Would anyone happen to know if Digium is still working on the 5v version of the TE410P? |
22:32.28 | n3wb33 | mark said a couple of days ago that he had one of those working |
22:32.31 | Stealth_Man | bkw: can i copy/paste sip.conf to you ? |
22:32.45 | coolhp | Thanks n3wb33. |
22:33.54 | Shido6 | any news on a Japanese version of the TE? |
22:34.45 | bkw_ | so help me Stealth_Man you paste me that config again.. i'm gonna reach thru IRC and smack them teeth out your head! |
22:34.50 | bkw_ | :> |
22:35.38 | bkw_ | put it on a website or use a paste bin |
22:35.43 | bkw_ | we need an * paste bin |
22:35.57 | Stealth_Man | ok .... hold on :) we will have it ;-) |
22:37.59 | Yog-home | hrm...when an AGI script prints to STDERR, where does the results go ?? |
22:38.23 | n3wb33 | Yog-home: to the *CLI |
22:38.23 | Stealth_Man | bkw: http://asterisk.xvoip.com/bin/ |
22:38.26 | bkw_ | should go to the cli |
22:38.36 | Yog-home | hrm..don't see it...do I have to have debug on ? |
22:39.09 | Stealth_Man | bkw: I will make open access to this, so people will be able to upload conf :) it iwll be Asterisk Paste Bin :)) |
22:39.11 | citats | Yog-home: they only go to the main * cli, not a remote |
22:39.20 | Yog-home | oh... |
22:39.29 | Yog-home | so asterisk -r dun work |
22:39.36 | citats | use the AGI VERBOSE command to send stuff everywhere |
22:40.07 | Yog-home | is there any way to get a 'main' CLI up w/o just running asterisk w/ -c ? |
22:40.19 | *** join/#asterisk Stealth_Man (~Stealth_M@h-67-101-152-122.NYCMNY83.dynamic.covad.net) |
22:40.22 | bkw_ | start asterisk with safe_asterisk then asterisk -r |
22:40.25 | levon | nite people |
22:40.39 | citats | Yog-home: nope |
22:40.47 | Yog-home | hrm...i'm in a "asterisk -r" already and the messages don't show |
22:40.49 | Yog-home | oh..ok... |
22:41.05 | citats | Yog-home: because STDERR goes to STDERR, and STDERR isn't duped to remote consoles |
22:41.10 | Yog-home | yep |
22:41.13 | Yog-home | makes sense |
22:41.15 | *** join/#asterisk daecon (~daecon@h-66-167-169-238.DNVTCO56.covad.net) |
22:41.18 | *** part/#asterisk daecon (~daecon@h-66-167-169-238.DNVTCO56.covad.net) |
22:41.24 | *** join/#asterisk daecon (~daecon@h-66-167-169-238.DNVTCO56.covad.net) |
22:41.31 | Yog-home | I could probably redirect STDERR to a file or something when I start * I guess |
22:42.06 | Yog-home | citats: what is the 'agi verbose' command..an AGI command or console command ? |
22:42.07 | *** join/#asterisk dnc (~duncan@213.240.59.220) |
22:42.43 | citats | Yog-home: its an AGI command... use 'show agi' in the * cli to see all the available AGI commands |
22:43.18 | UnixDawg | ok this is weird my outgoing is working fine |
22:43.27 | UnixDawg | but my iax incoming is not |
22:44.25 | anthm | I added a new section in musiconhold.conf [custom_exec] |
22:44.31 | UnixDawg | 66.234.228.132:4569 in-PLy97oR 192.168.0.2:61190 60 Registered |
22:44.31 | UnixDawg | *CLI> |
22:44.36 | anthm | default => /usr/local/bin/ogg123,-q,--audio-buffer,2,/tmp/1.ogg,-d,wav,-f,- |
22:44.45 | anthm | that example lets me play ogg file in moh |
22:44.46 | UnixDawg | its not putting me on the correct port |
22:45.01 | anthm | and does not disturb existing functionality whatsoever |
22:45.10 | UnixDawg | hmm |
22:47.02 | *** join/#asterisk anthm (~anthmct@CPE-24-167-193-161.wi.rr.com) |
22:49.01 | denon | citats: you around? |
22:52.09 | *** join/#asterisk joe_satriani (~joe_satri@ppp-210-148.28-151.libero.it) |
22:56.36 | joe_satriani | to activate cdr_mysql I need to make it only ? |
22:57.37 | *** join/#asterisk mortck (~test@p213.54.146.115.tisdip.tiscali.de) |
22:57.46 | UnixDawg | you need the add-ons I believe |
22:58.01 | bkw_ | its cdr_addon_mysql.so now |
22:58.15 | joe_satriani | I ve yet cvs it |
23:00.37 | *** join/#asterisk cypromis (~michael@217.11.142.161) |
23:04.41 | *** join/#asterisk Titoon (julien@deepblue.titoon.net) |
23:05.24 | marcus- | heheh |
23:05.40 | marcus- | so if you only have a credit of $0.50 left of your voicepulse connect account |
23:05.49 | marcus- | and you make a call that is longer than $0.50 |
23:05.52 | marcus- | it wont cut you off |
23:06.23 | bkw_ | marcus- good to know |
23:06.32 | Stealth_Man | heh |
23:06.37 | Stealth_Man | nice .. terrible |
23:07.00 | UnixDawg | grrr |
23:07.06 | Stealth_Man | bkw: so what migth be wrong in conf ? |
23:07.07 | UnixDawg | ok somethign is wrong here |
23:07.09 | bkw_ | Current: $6.05 |
23:07.10 | bkw_ | Pending: $0.00 |
23:07.17 | UnixDawg | and I think its on thier side now |
23:07.20 | bkw_ | I got 6.05 left |
23:07.45 | UnixDawg | wif who |
23:07.52 | bkw_ | voicepulse |
23:07.52 | marcus- | woah, thats amazing |
23:07.59 | marcus- | i just had a *good* customer service experience with att wireless |
23:08.03 | UnixDawg | I have 15 bucks left |
23:08.06 | Stealth_Man | bkw: I will give you free service for life ! |
23:08.10 | marcus- | they must be scared about the whole number portability thing |
23:08.10 | bkw_ | marcus- OMG really? |
23:08.14 | marcus- | my phone broke last night |
23:08.15 | bkw_ | haha |
23:08.21 | Stealth_Man | make me working :) |
23:08.23 | marcus- | i called them today to see if it was still under warrant, and they said it was out by a month |
23:08.36 | marcus- | but that i could get a new phone for free if i sign another 1yr contract |
23:08.44 | *** join/#asterisk Gazzas (Gazzas@c211-28-128-147.eburwd3.vic.optusnet.com.au) |
23:08.47 | marcus- | of course, i've had attws for over 5 years now, and i really like not being under contract |
23:09.05 | Stealth_Man | marcus: ask them for optional insurance for your cell. phone ... it gonna cost you 1$ i think amonth |
23:09.10 | UnixDawg | *CLI> NOTICE[135322624]: File chan_iax2.c, Line 4537 (socket_read): Host 66.234.228.132 failed to authenticate as voicepulse |
23:09.10 | bkw_ | anthm what did you do ? |
23:09.16 | Stealth_Man | after 1 month you can call and ask for new phone |
23:09.19 | UnixDawg | voicepulse is screwed again |
23:09.23 | Stealth_Man | they allow 3 phone replacements ... |
23:09.30 | anthm | I added a new sectinon |
23:09.31 | marcus- | so i said "well, if i'm going to have to sign a new contract, i'm going to research what your competitors are offering first" |
23:09.37 | anthm | [custom_exec] |
23:09.43 | anthm | that lets you set up your own pipe |
23:09.51 | marcus- | suddenly i hear some typing, and then "oh, well, you've been a customer for a long time and we appreciate that. i'm sure i can work something out with the warranty dept" |
23:10.10 | anthm | allowing you to use any app to play back |
23:10.25 | marcus- | ud; i just made a 10 minute call via vp |
23:10.33 | bkw_ | UnixDawg mine works fine |
23:10.34 | anthm | here is an example entry defining a default channel |
23:10.38 | anthm | default => /usr/bin/mpg123,-q,-s,--mono,-r,8000,-b,2048,http://64.236.34.97:80/stream/1014 |
23:10.54 | anthm | sticking a shoutcast into moh |
23:10.57 | bkw_ | anthm their is already a patch on the net to make that possible |
23:11.09 | anthm | and here is one doing ogg |
23:11.10 | *** join/#asterisk dmh (~dmh@ool-44c22549.dyn.optonline.net) |
23:11.10 | anthm | default => /usr/local/bin/ogg123,-q,--audio-buffer,2,/tmp/1.ogg,-d,wav,-f,- |
23:11.25 | dmh | digium come out with 4port fxo yet? |
23:11.25 | Stealth_Man | bkw : are taking free long distance from me or not ?:) |
23:11.29 | bkw_ | http://bugs.digium.com/bug_view_page.php?bug_id=0000413 |
23:11.30 | anthm | this makes everything possible not just 1 thing |
23:11.42 | bkw_ | anthm oh sure make it where stupid people can break it |
23:11.53 | bkw_ | haha |
23:12.12 | UnixDawg | bkw pvt for 1 min |
23:12.14 | anthm | you have a total front end to the exec call |
23:12.32 | bkw_ | UnixDawg not right now.. just keep it in the channel |
23:12.44 | anthm | so you never have to touch src again no matter what you may need any app that can deliver the right format can be loaded |
23:13.01 | anthm | <PROTECTED> |
23:13.13 | bkw_ | anthm better test and provide examples.. |
23:13.14 | bkw_ | do this |
23:13.18 | bkw_ | post it to bugs.digium.com |
23:13.19 | *** join/#asterisk zonyl (~newbie@CPE-65-31-131-186.wi.rr.com) |
23:13.20 | dmh | you cant compete with the traditional pbx until the 4port fxo becomes available at a lower cost |
23:13.23 | bkw_ | with documentation |
23:13.24 | dmh | i do say |
23:13.35 | bkw_ | dmh yes you can |
23:13.43 | dmh | state how |
23:14.01 | dmh | you would need a large backplane |
23:14.06 | bkw_ | most PBX's ie Partner ACS's have only 3 FXO's and 8 extensions |
23:14.09 | anthm | hehe i was gonan do that with my extensive app_queue patches till I realized it was more typing than making the hacks |
23:14.11 | dmh | and that architecture is not cost efective |
23:14.19 | anthm | this one is pretty easy to document tho |
23:14.35 | dmh | bkw, the smallest port CO card you can get for a traditional phone system is 4 |
23:14.47 | zonyl | is there a IAX lib for perl for establish receive calls? |
23:14.49 | dmh | and those are like $90 |
23:14.56 | bkw_ | dmh you can still compete just try... 4 lines per box.... or a T1 card and channel bank |
23:14.58 | bkw_ | still easy |
23:15.09 | dmh | channel banks are not cheap |
23:15.19 | bkw_ | not too expensive |
23:15.21 | dmh | the 4 port fxo is key |
23:15.24 | dmh | it must be made! |
23:15.28 | bkw_ | it will be |
23:15.29 | dmh | cheaply that is |
23:15.34 | dmh | i know |
23:15.35 | dmh | i cant wait |
23:15.36 | dmh | heh |
23:15.37 | UnixDawg | its still failing to auth in inbound |
23:15.48 | UnixDawg | yet when I register it works fine |
23:15.54 | Stealth_Man | unix: and i still have no luck with even outgoing ... |
23:16.25 | UnixDawg | stealth my out going works |
23:16.28 | dmh | i heard rumor its under aproval |
23:16.31 | UnixDawg | I have tested it |
23:16.33 | dmh | or somethin |
23:16.57 | dmh | it has been made, just not yet released |
23:20.07 | Stealth_Man | ehh |
23:21.25 | bkw_ | blah |
23:21.54 | UnixDawg | makes no sense that its not registering on inbound |
23:22.33 | bkw_ | inbound is diffrent |
23:22.39 | bkw_ | bet they need to fixor your stuff |
23:23.12 | Yog-home | ok..what are the arguments to the AGI app ? I expected AGI(script,arg0,arg1,arg2) to show up in $ARGV[0..N] ... but it doesn't..what gives ? |
23:23.25 | bkw_ | use Asterisk::AGI |
23:24.05 | Yog-home | ugh |
23:24.21 | Yog-home | I have to get another damn package :) |
23:24.57 | bkw_ | Asterisk::AGI is all you need baby |
23:25.06 | Yog-home | kinda funny that the agi-test.pl file that comes w/ * doesn't use it |
23:25.34 | bkw_ | Asterisk::AGI makes it simpler to work with it |
23:25.36 | Yog-home | I'm writing an AGI to 'reverse map' a SIP user name to an extension alias using the SER aliases table :-) |
23:25.55 | Yog-home | then I can map sip:user@domain -> extension :) |
23:26.16 | Yog-home | all this just to get VM working :) |
23:26.39 | Yog-home | too bad my customer had to use user names instead of just extension numbers for his users :-( |
23:27.02 | Yog-home | but that's kinda what SER assumes anyway... SIP addy doesn't have to be a phone # |
23:27.15 | bkw_ | doesn't have to be in * either |
23:27.41 | Yog-home | yeh..but I just don't feel like creating a bunch of extensions in the form of "username" |
23:27.47 | Yog-home | i could do that I guess |
23:28.03 | *** join/#asterisk TerraByte (~paul@as3.dm.egate.net) |
23:28.07 | Yog-home | but it kinda scares me..what if one of the users is named "s" or "i" or "h" :-) |
23:28.18 | TerraByte | Is there some reason when * emails VoiceMails the last 2s seems cut off every time? |
23:28.50 | Yog-home | phone is probably cutting off audio when it gets the BYE |
23:29.07 | TerraByte | Playing through VoiceMailMain is fine |
23:29.34 | Yog-home | hrm... |
23:29.43 | Yog-home | maybe it's bundling it up wrong |
23:30.50 | TerraByte | ? |
23:32.27 | Takapa | is it correct to assume that I would have to define ZAPATA_NET when compiling zaptel to be able to push "raw" data connections over a TE410P? |
23:32.57 | Stealth_Man | anthm: what this patch is doing ? |
23:33.30 | *** join/#asterisk ionix (~ionix@MTL-ppp-150366.qc.sympatico.ca) |
23:33.37 | doughecka | wow, 256 megs of DDR 2100 crucial memory for 37 bucks |
23:33.42 | Stealth_Man | Yog-home: are you using SER ?? |
23:33.46 | *** join/#asterisk scromp (~bnaylor@haybaler.sackheads.org) |
23:33.55 | anthm | the patch lets you pick the whole exec string to spawn the music player |
23:34.08 | anthm | meaning you can use any music player you can get to work |
23:34.28 | bkw_ | wav49 is gsm kinda with a riff header so windows will play it |
23:34.29 | doughecka | NOO, WHY do they do this!?!?! I order something, and they freaking mark it down 5 dollars for a one day sale the day AFTER I order it!!! |
23:34.38 | bkw_ | windows doesn't like it and cuts off the last two seconds |
23:34.42 | bkw_ | with a nasty error message |
23:34.50 | bkw_ | TerraByte send in gsm and use quicktime to play them |
23:36.02 | Yog-home | doughecka: your order pushed their profits over the to so they could do the mark down |
23:36.08 | doughecka | bah |
23:36.11 | doughecka | it was 100 dollars |
23:37.47 | marcus- | damn. i'd forgotten just how lame it is to be an oper on efnet |
23:38.30 | TerraByte | bkw: I switched to wav |
23:38.33 | TerraByte | it seems to work |
23:39.17 | drgalaxy | marcus-: what made you remember? |
23:39.53 | doughecka | drgalaxy: they just gave it back to him |
23:39.54 | doughecka | :) |
23:40.05 | marcus- | oh someone opered me again last night |
23:40.09 | marcus- | its been a few years |
23:40.56 | *** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net) |
23:42.02 | doughecka | BAH |
23:42.15 | doughecka | and they are selling my mp3 player for the same price I got it off ebay |
23:42.18 | doughecka | without shipping |
23:42.25 | doughecka | brandnew |
23:45.58 | marcus- | With few options left, ambulance crew members cut the telephone off at the base and took it and Fleming to St. Mary's Hospital. |
23:46.00 | marcus- | doh! |
23:46.17 | doughecka | ? |
23:46.50 | Marlow | ? |
23:46.52 | drgalaxy | doughecka: are you new to being a consumer? every time you buy a computer it gets cheaper later on |
23:46.55 | Marlow | sick phone ? |
23:46.55 | marcus- | http://www.cnn.com/2003/US/Midwest/11/18/offbeat.payphone.ap/index.html -- EAST ST. LOUIS, Illinois (AP) -- A man and a pay phone were rushed to a hospital after he got his finger stuck in the coin return slot while trying to retrieve his 50 cents. |
23:47.37 | *** join/#asterisk onixx (1000@CPE0040f47145d1-CM014320024117.cpe.net.cable.rogers.com) |
23:47.42 | doughecka | LOL |
23:47.46 | *** part/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net) |
23:48.17 | bkw_ | OMG |
23:48.20 | bkw_ | got his finger stuck |
23:48.40 | doughecka | what was he doing? reaching up inside the thing? |
23:48.49 | onixx | hi all... anyone can explain if possible to park a call with an ata-286 ? |
23:48.50 | *** join/#asterisk ManxPower (~werecat@ip68-107-251-114.no.no.cox.net) |
23:49.02 | bkw_ | onixx # transfers |
23:49.04 | bkw_ | are the only way to do it |
23:50.30 | marcus- | read the article :P |
23:50.49 | onixx | anything special to configure ? |
23:51.00 | doughecka | you need spoons |
23:51.56 | onixx | bkw_: when I press # nothing happens |
23:51.58 | bkw_ | anthm do you know how to make unified diff files? |
23:52.10 | bkw_ | show application dial optiosn T and t |
23:54.00 | onixx | bkw_: ok ... got it. thanks. do you know of any other features you can do with # or * ? |
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23:55.09 | Yog-home | Warning: Cannot install Asterisk::AGI, don't know what it is. |
23:55.10 | Yog-home | Try the command |
23:55.18 | Yog-home | grr...it's not in CPAN |
23:55.30 | marcus- | no, its not |
23:56.32 | Yog-home | ok..can anybody tell me where the argumetns that you pass in an AGI command go ? |
23:56.45 | *** part/#asterisk Chris_DE (~Chris_DE@p5083053C.dip0.t-ipconnect.de) |
23:57.23 | Yog-home | anthm: just cat them together, no ? |
23:57.44 | marcus- | uhm, they are put in *argv[] |
23:57.57 | n3wb33 | diff -u -r should do |
23:58.02 | Yog-home | yet, when I print @ARGV it doesn't show |
23:58.36 | marcus- | what does it show? |
23:58.43 | marcus- | blankness, or a number? |
23:58.44 | Yog-home | "" |
23:58.51 | Yog-home | blank |
23:59.01 | marcus- | oh, strange. what does your extensions.conf entry look like? |
23:59.19 | Yog-home | AGI(script.agi,arg1,arg2,arg3) |