| 00:00.11 | bjohnson | if you fixed it, a lot of people would be happy |
| 00:00.17 | *** join/#asterisk outsidefactor (barf@203-173-32-225.dyn.iinet.net.au) |
| 00:00.56 | phantam | yes |
| 00:00.58 | phantam | i know |
| 00:01.00 | phantam | but im not a programmer |
| 00:01.06 | phantam | and this doesnt look like a small issue |
| 00:02.04 | implicit | anyone here using g722? |
| 00:03.12 | CarlosMP_ | anyone know if the TE110P card will support a T1, non PRI? |
| 00:03.23 | fearnor | yes |
| 00:03.35 | implicit | fearnor: how do you like it? |
| 00:03.53 | ManxPower | phantam, As I said it does not sound logical. However, the pwlib and h323 libs are horribly buggy and chan_h323 works around those bugs. Of course openh323 keep changing things. So They just say use this and only this and statically link it just to be safe. |
| 00:03.57 | fearnor | sorry i was answering to carlosmp |
| 00:04.05 | CarlosMP_ | so will a standard channel bank like a adtran 750 work? |
| 00:04.09 | fearnor | carlos: yes |
| 00:04.19 | ManxPower | CarlosMP_, Yes. |
| 00:04.29 | phantam | ugggg |
| 00:04.52 | phantam | but why the hell does chan_h323 need pwlib sources and openh323 sources and this and that |
| 00:04.53 | ManxPower | phantam, There IS nother h323 channel driver. |
| 00:05.01 | phantam | ? |
| 00:05.04 | ManxPower | phantam, there's a reason for all of it. |
| 00:05.28 | fearnor | phantam: uhhh, because it needs to interoperate with it. and pwlib/openh323 don't have 'installable headers' package. |
| 00:05.31 | CarlosMP_ | ManxPower: Any other channel banks less $$ that would work? |
| 00:05.32 | ManxPower | Don't like it, write your own channel driver or fork one of the existing ones. |
| 00:05.34 | *** join/#asterisk florz (nobody@odnb-d9baa541.pool.mediaWays.net) |
| 00:05.47 | phantam | so how do i get this to work |
| 00:05.51 | fearnor | carlos: lots. i strongly suggest Adit 600 |
| 00:05.52 | phantam | i folled instructions but it errors |
| 00:05.58 | fearnor | phantam: hire a consultant. |
| 00:05.59 | ManxPower | CarlosMP_, Most Adtran channel banks. You can get them on eBay fairly cheap. |
| 00:06.01 | phantam | i tried doing just make opt on the old pwlib |
| 00:06.03 | phantam | and it failed |
| 00:06.06 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 00:06.13 | phantam | lol sure ill hire when i get cash |
| 00:06.17 | phantam | in a year or so |
| 00:06.17 | phantam | lol |
| 00:06.21 | fearnor | phantam: ask on openh323 mailing lists then. its not even asstricks related |
| 00:06.25 | phantam | open source used to be so fun |
| 00:06.43 | phantam | fearnor: thats like saying if the sip chan failed "oh its not asterisk related we wont help" |
| 00:06.47 | ManxPower | Heck, I spent $125 on consultants in the past 2 weeks. |
| 00:06.50 | phantam | or if all the chans stopped working |
| 00:06.53 | fearnor | phantam: open source is still fun. berating newbies for not doing homework > * |
| 00:06.57 | fearnor | :) |
| 00:07.04 | ManxPower | Not a lot of time, but very productive. |
| 00:07.14 | phantam | lol |
| 00:07.18 | phantam | well wouldnt have to do that |
| 00:07.22 | phantam | if programmers programmed properly |
| 00:07.29 | phantam | makes windows look stable |
| 00:07.45 | ManxPower | phantam, Get over it, Dearie. |
| 00:07.51 | fearnor | phantam: ask for a refund. |
| 00:07.51 | phantam | yaya |
| 00:07.54 | phantam | lol |
| 00:08.21 | JamesDotCom | phantam: now you're just being stupid and giving people less and less reason to help you |
| 00:08.27 | CarlosMP_ | fearnor: do you know what distributors carry adit? TechData, Ingram, Synnex don't carry |
| 00:08.28 | phantam | lol |
| 00:08.31 | phantam | not stupid but its true |
| 00:08.34 | fearnor | carlos: ebay does |
| 00:08.35 | JamesDotCom | i'm telling you that if you follow the instructions word for word, it will compile |
| 00:08.42 | phantam | i was told by 1 asterisk devel why all of a suddent h323 isnt important |
| 00:08.47 | ManxPower | phantam, Contribute in SOME way. I do a lot of support and ocasionally offer bounties for features I want. |
| 00:08.51 | fearnor | carlos: seriously speaking though |
| 00:08.54 | phantam | found it kinda poor reason |
| 00:09.01 | fearnor | carlos: you probably don't want to buy new. new = assrape. |
| 00:09.10 | JamesDotCom | if you're not capable of setting something up, dont keep whinging about it, pay someone to fix it if you're not capable |
| 00:09.14 | afe | any of you oldtimers that could take a moment and look at my first bug report (0003700) and then gently get back to me and tell me what I did wrong? :) |
| 00:09.15 | phantam | JamesDotCom: i tried compiling pwlib the way they say it errors out and says to contact gentoo admin |
| 00:09.18 | fearnor | buy grey-market equipment |
| 00:09.30 | ManxPower | I also write sample scripts, stuff like that. |
| 00:09.34 | fearnor | phantam: goddamnit, it SAYS TO CONTACT GENTOO ADMIN. |
| 00:09.39 | phantam | lol |
| 00:09.43 | fearnor | what do you do? come to #asterisk and bitch and moan? |
| 00:09.46 | phantam | it says that about every failed build |
| 00:09.55 | fearnor | well, not our problem. go away. |
| 00:10.01 | phantam | :P |
| 00:10.26 | nestAr | lol |
| 00:10.26 | bjohnson | CarlosMP_: adit 600's on ebay too usually |
| 00:10.37 | fearnor | adit 600s go for ~500$ or so |
| 00:10.44 | fearnor | and they are the best channel banks you can buy :) |
| 00:10.45 | fearnor | IMHO. |
| 00:10.52 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 00:10.54 | fearnor | at least the only channel bank with proper echocancel |
| 00:10.59 | CarlosMP_ | bjohnson: I saw a few, I just like getting new for customers... |
| 00:11.01 | fearnor | i had issues until i got adit. |
| 00:12.26 | bjohnson | phantam: get it to work and write a howto to make it easier for others. I'm told that the instructions are as good as possible |
| 00:12.53 | CarlosMP_ | fearnor: have you used it to convert incoming POTs lines to a TE110P? |
| 00:13.04 | fearnor | never done FXO |
| 00:13.14 | fearnor | with any bank |
| 00:13.26 | fearnor | read wiki tho |
| 00:13.33 | fearnor | there's much on channel banks and specific things you need to look at |
| 00:13.44 | bjohnson | I think tzanger uses adit600 with fxo ports |
| 00:13.57 | bjohnson | but I'm not sure why .. I thought he had PRIs |
| 00:14.02 | phantam | bbl |
| 00:14.03 | *** part/#asterisk phantam (~phantam@72.252.15.235) |
| 00:14.10 | bjohnson | can't wait |
| 00:15.20 | CarlosMP_ | bjohnson: I have a couple customers that have a bunch of POTS lines, rather than looking for a small 1U server with 4 PCI slots to support 13 POTS, I rather go to a TE110P, which handles a T1 and use a channel bank to convert fxo lines to T1 or T1/PRI |
| 00:15.35 | fearnor | smart choice. |
| 00:16.25 | CarlosMP_ | but I'm hoping that the channel bank doesn't cost more... |
| 00:16.40 | fearnor | it does if you buy new probably ;() |
| 00:17.39 | CarlosMP_ | I just want to get something that works well without requiring too much work |
| 00:18.00 | *** join/#asterisk WarchildX (user@gso26-198-064.triad.rr.com) |
| 00:18.02 | CarlosMP_ | And so I can come up with a "standard" deployment that can fit all of our customers |
| 00:18.31 | ManxPower | CarlosMP_, One size does not fit all. Do three standard configs. |
| 00:18.56 | CarlosMP_ | ManxPower: why 3? |
| 00:19.01 | ManxPower | 3x TDM400P for up to 12 ports, any combo. Careful of IRQ shareing. |
| 00:19.22 | CoaxD | god, i hate irq sharing motherboards. The ones you cant set irq priorities on |
| 00:19.30 | fearnor | coad: that's every motherboard. |
| 00:19.32 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 00:19.35 | _-Jon-_ | Hey everyone |
| 00:19.38 | CoaxD | fearnor: Nah |
| 00:19.40 | ManxPower | A TE110P and a channel Bank. |
| 00:19.40 | fearnor | yes. |
| 00:19.46 | CoaxD | fearnor: There are very good motherboards out there |
| 00:19.48 | ManxPower | And just a TE405P |
| 00:19.53 | fearnor | you will *always share irqs* |
| 00:19.54 | *** part/#asterisk WarchildX (user@gso26-198-064.triad.rr.com) |
| 00:19.54 | CoaxD | fearnor: They allow you to set irq based on pci slot |
| 00:19.58 | fearnor | get APIC-capable mobo |
| 00:19.59 | CarlosMP_ | Well, wouldn't a Channel bank still need the TE110P? |
| 00:20.07 | CoaxD | fearnor: Um |
| 00:20.14 | CoaxD | fearnor: Dude. :) |
| 00:20.51 | _-Jon-_ | I'm wondering if someone could help me out with a dialplan problem I'm having.. I'm wondering how I would go about this.. A caller calls in and gets a menu: press 1 for, 2 for something else, and so on. Once the caller presses 1 for example, it jumps to another menu, where it says again press 1 for something, 2 for somethign else. How do I make the 1 and 2 ont eh 2nd menu different extensions? or am I missing som |
| 00:21.03 | fearnor | you can always do irq assignment through ACPI if mobo is capable |
| 00:21.27 | CoaxD | fearnor: READ: If mobo is capable :) |
| 00:21.45 | CoaxD | nowadays, i bet you have a much better chance of getting that should you walk in and buy a mobo |
| 00:22.02 | CarlosMP_ | configuring irq's isnt the issue, it's the fact that coming across mobo's with more than 3 PCI slots are rare now, since everything is coming with PCI-Xpress, etc. |
| 00:22.03 | CoaxD | Jon: Jump to another extension context |
| 00:22.15 | CoaxD | Jon: Goto(blah-extension,1000,1) |
| 00:22.29 | CoaxD | Jon: Playback(new_menu) |
| 00:22.30 | _-Jon-_ | CoaxD, Thanks so much! |
| 00:22.43 | CoaxD | Jon: Yer welcome. :) |
| 00:22.58 | CoaxD | Carlos: WTF is PCI-Xpress? |
| 00:23.34 | CoaxD | Carlos: and btw, yeah, i've had that problem a lot. Mobo only comes with 2 frickin' pci slots, et |
| 00:23.35 | CoaxD | c |
| 00:24.01 | CoaxD | (There are still boards made that come with 5 pci slots. But where are the days of having a mobo loaded with like 8 of 'em?? I WANT THOSE DAYS BACK!) |
| 00:24.12 | CarlosMP_ | Intel 915GAV has 4..., but can't get that to fit in a 1U server |
| 00:24.18 | CoaxD | lame |
| 00:24.37 | CoaxD | well since just about everything works in either a firewire or usb port now, i suppose they figure nobody actually needs pci cards anymore |
| 00:24.44 | CarlosMP_ | Just thought of something - could I use a ADIT 600 with FXO module and connect to Asterisk via ethernet? |
| 00:24.56 | CarlosMP_ | rather than using a TE110P? |
| 00:24.58 | CoaxD | CarlosMP: No |
| 00:25.02 | CoaxD | er |
| 00:25.09 | CoaxD | sorry. I read ABIT. I was confused. *lol* |
| 00:26.01 | CarlosMP_ | Which would mean that the adit600 would be acting as a VOIP gateway as far as asterisk is concerned? |
| 00:26.07 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 00:26.29 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 00:26.36 | *** join/#asterisk Gronker (~Gronker2@adsl-220-89-19.ags.bellsouth.net) |
| 00:29.15 | fearnor | coaxd: go get passive-backplane boards |
| 00:29.25 | fearnor | carlos: pretty much |
| 00:29.32 | bjohnson | _-Jon-_: usually you use background instead of playback since it can be interuprted by the caller |
| 00:30.06 | bjohnson | got a weird problem now |
| 00:30.18 | CarlosMP_ | fearnor: so the abit CMG card would be the key... |
| 00:30.24 | bjohnson | was a working system . |
| 00:30.27 | bjohnson | crap have to go |
| 00:30.30 | fearnor | carlos: ADIT |
| 00:30.46 | CarlosMP_ | my mistake this time I mistyped... |
| 00:30.46 | fearnor | carlos: but if you are putting money into that...why do you need asterisk ;) |
| 00:30.47 | fearnor | heh |
| 00:30.55 | fearnor | CMG card = $$$ |
| 00:30.59 | CarlosMP_ | cheaper than Nortel... |
| 00:31.05 | CarlosMP_ | BCM400 |
| 00:33.24 | CarlosMP_ | On another note - what have you guys been using for CPU/RAM/HD? |
| 00:35.24 | bjohnson | ok back |
| 00:35.24 | Crad|Work | Quick question, I'm trying to override all channels for outbound caller id. I've set the zapata.conf with the following lines: http://rafb.net/paste/results/EFh0P973.html |
| 00:35.30 | Crad|Work | does anyone have an idea what I'm doing wrong? |
| 00:35.58 | bjohnson | with a SPA 3000 when I dial a local number, the cli shows a connection from the fxs to the fxo and that it is ringing .. but I just get dial tone |
| 00:36.07 | fearnor | crad: what knid of line do you have? |
| 00:36.10 | Crad|Work | t1 |
| 00:36.14 | fearnor | PRI? |
| 00:36.19 | Crad|Work | yeah |
| 00:36.25 | fearnor | signaling? NI2? |
| 00:36.32 | fearnor | it looks like it should work. |
| 00:36.37 | Crad|Work | hmm dont know that, what it's doing is sending our extension #'s as our caller id |
| 00:36.45 | fearnor | check with your telco if they allow you to set clid. |
| 00:37.00 | Crad|Work | they do, since it's setting the extension of our phones as the caller id |
| 00:37.00 | fearnor | wlel, check where you are setting extension ## |
| 00:37.17 | bjohnson | CarlosMP_: couple of pages about hardware sizing on the wiki |
| 00:37.18 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 00:37.18 | Crad|Work | I was under the impression I could override that at the zapata level |
| 00:37.26 | Katty | hmm, webcam is on |
| 00:37.38 | Crad|Work | on a side note, is there anything in 1.0.6 that would break 1.0.0 config files? |
| 00:37.47 | Crad|Work | (We're on 1.0.0 and my boss is fearful of upgrading) |
| 00:37.58 | angler | conif files wont break |
| 00:39.20 | PatrickDK | crad, you shouldn't have any problem |
| 00:39.50 | PatrickDK | my pre 1.0 configs worked fine on current cvs |
| 00:40.18 | afe | anyone knows if * on freeBSD has a different behaviour on SRV lookups in sip.conf |
| 00:40.39 | afe | compared to linux |
| 00:41.48 | bjohnson | Katty: caught you pucking your nose? |
| 00:41.54 | bjohnson | err picking even |
| 00:42.10 | Katty | no |
| 00:42.25 | Katty | www.brick.net/~izaa |
| 00:43.03 | Katty | www.brick.net/~izaah (= |
| 00:43.38 | mikeirc | I love os x... don't mind me..just thinking out loud... |
| 00:44.08 | tzanger | KDE3.4rc1 kicks ass |
| 00:44.24 | tzanger | Katty: is that you or your alter ego |
| 00:44.55 | tzanger | hmm angela... heh I knew a couple angelas |
| 00:44.57 | Katty | me |
| 00:44.57 | Katty | tell me |
| 00:44.57 | Katty | to d |
| 00:44.57 | Katty | o something! |
| 00:44.59 | Katty | i'll do it |
| 00:44.59 | tzanger | both were pretty awesome women |
| 00:45.01 | Katty | maybe |
| 00:45.05 | tzanger | pick your noser |
| 00:45.22 | Katty | that finger? |
| 00:45.37 | tzanger | the refresh on that could be beaten by a 300 baud modem |
| 00:45.44 | tzanger | yup that finger will work |
| 00:46.04 | Katty | i was flipping you off ;) |
| 00:46.14 | tzanger | doesn't matter you asked if the finger'll work and I said sure |
| 00:46.19 | bjohnson | work it baby |
| 00:46.28 | bjohnson | the camera loves you |
| 00:46.29 | Katty | tzanger: heh |
| 00:46.31 | tzanger | this is one of those PG13 webcams |
| 00:46.37 | Katty | bjohnson: also, shut up (= |
| 00:46.44 | Katty | bjohnson: i am not a slut, kthx |
| 00:47.03 | tzanger | ?? |
| 00:47.08 | tzanger | I work it and 'm no slut |
| 00:47.11 | Crad|Work | do I have to restart asterisk for changes to zapata.conf to take place, or will a reload in asterisk work fine for that? |
| 00:47.11 | tzanger | I'm a man-whore |
| 00:47.20 | tzanger | Crad|Work: restart is required for zapata.conf |
| 00:47.27 | tzanger | well any changes to echocan and the like |
| 00:47.35 | Katty | not my problem (= |
| 00:47.35 | tzanger | I think groupings and contexts will work with a reload but don't quote me on that |
| 00:47.43 | |Vulture| | oh great... not a webcam |
| 00:47.43 | tzanger | Katty: I dare you to smile |
| 00:47.45 | mikeirc | Katty: I like your sketch of the farie |
| 00:47.52 | tzanger | mikeirc: it's a pic of you :-p |
| 00:48.14 | Katty | i don't smile on camera (= |
| 00:48.17 | tzanger | if I had a webcam I'd scare y'all away |
| 00:48.23 | mikeirc | tzanger: Well...i do have long hair...hey!! |
| 00:48.27 | tzanger | hahahaha |
| 00:48.30 | tzanger | bah |
| 00:48.40 | tzanger | some webcam, she says she'll do requests but she won't pick her nose and she won't smile |
| 00:48.50 | tzanger | I think we got us a pseudo goth chick |
| 00:48.55 | Katty | tzanger: yup |
| 00:49.07 | Katty | i'll show my favorite cow! |
| 00:49.10 | tzanger | oh and she won't do private shows either... jeez |
| 00:49.15 | PatrickDK | heh |
| 00:49.27 | mikeirc | Katty: I got something for you to do on camera...you can build my asterisk server! :) |
| 00:49.56 | tzanger | Ann Rice and ATI... odd combo |
| 00:50.04 | Crad|Work | thx |
| 00:50.30 | bjohnson | does a different .gsm have to be recorded for each codec that might use it? |
| 00:50.46 | Katty | mikeirc: pffft |
| 00:50.47 | tzanger | bjohnson: uh .gsm is coded with teh GSM codec |
| 00:51.00 | tzanger | how the blue fuck do you manage 18 pages of questions |
| 00:51.01 | tzanger | wowza |
| 00:51.41 | bjohnson | so it plays ok with ulaw but not so great with 726-32 |
| 00:51.48 | mikeirc | Katty: hold up the squirrel you have in the gallery :) |
| 00:52.02 | Katty | k, moment |
| 00:52.18 | tzanger | holy shit she's got a pic with a smile in it |
| 00:52.26 | tzanger | and she's at a keyboard. what kind of music |
| 00:52.27 | *** join/#asterisk {zombie} (zombie@soulasylum.penguincare.com.au) |
| 00:53.10 | Katty | k, there is teh squirrely |
| 00:53.18 | tzanger | pics of shoes too... typical female. :-) |
| 00:53.25 | tzanger | have to say I admire a woman who can wear heels like that |
| 00:53.38 | tzanger | they ain't stillettos but they certainly ain't flat either |
| 00:53.38 | mikeirc | Katty: hehe.. too cool thx. ;) |
| 00:53.41 | Katty | those are little heels |
| 00:53.58 | *** join/#asterisk jbAU (~johnhewit@mail.lanskey.com.au) |
| 00:54.00 | *** join/#asterisk tuxinator_linux (~anonymous@m410e36d0.tmodns.net) |
| 00:54.09 | tzanger | the biggest heels you'll catch me wearing are I think 3/4" |
| 00:54.25 | PatrickDK | hmm, I have done 2" |
| 00:54.30 | Katty | any other requests? |
| 00:54.31 | tzanger | hahaha |
| 00:54.35 | tuxinator_linux | heels? |
| 00:54.35 | PatrickDK | traded with one of my girlfriends |
| 00:54.36 | tzanger | PICK YER DAMN NOSE |
| 00:54.41 | Katty | tzanger: no |
| 00:54.47 | PatrickDK | she found out she doesn't like heels, so she tool my shoes |
| 00:54.53 | PatrickDK | took |
| 00:54.53 | tuxinator_linux | am I in the wrong chat room |
| 00:55.02 | tzanger | you wear the same size as your gf?? |
| 00:55.21 | PatrickDK | tzanger, close enough, she was a 9, I was a 10 |
| 00:55.26 | tzanger | the woman I'm seeing now is 8" shorter than I am and I think she could just about fit both her feet in one of mine |
| 00:55.29 | tzanger | I'm a 12W |
| 00:55.29 | jbAU | tuxinator_linux : well i guess asterisk is all about communication |
| 00:55.33 | PatrickDK | I think 9 mans is 6 womans |
| 00:55.50 | tzanger | you know what they say about guys with big hands and big feet... |
| 00:55.55 | tzanger | ... big mitts and big shoes :-) |
| 00:55.56 | tuxinator_linux | jbAU: We are being very open |
| 00:56.09 | tuxinator_linux | yep |
| 00:56.31 | jbAU | tzanger : yeah, they're more clumsy with precision tools :P |
| 00:56.44 | tzanger | jbAU: nonsense |
| 00:56.48 | tuxinator_linux | I got my Meet * ticket today, and my hotel room. Now just getting my flight. |
| 00:57.08 | tzanger | I solder 204 pin PQFP chps with nothing more than a good handheld iron and a flux pen |
| 00:57.11 | jbAU | well i would quote my shoe size, but i have nfi what american shoe sizes are |
| 00:57.20 | tzanger | and I've done 0.5mm pitch TQFPs too |
| 00:57.41 | mikeirc | Katty: I've got a charles and a chigger cat too (Symphony and Calypso)! And another..sleeping on my lap right now.. :) Too many cats.... nah. ;) |
| 00:58.07 | bjohnson | grrrrr .. what is going on |
| 00:58.09 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 00:58.23 | jbAU | so - extension _NNX2 would be able to give a match for say.. 2202 ? |
| 00:58.32 | Katty | mikeirc: yay! (= |
| 00:58.32 | tuxinator_linux | You're in the Twilite Zone |
| 00:58.32 | tzanger | bjohnson: well I'm in my living room in my bathrobe with the laptop on the lap, showplow's are going by and the kids are sleeping... |
| 00:58.34 | bjohnson | yes |
| 00:58.48 | bjohnson | this SPA is acting up |
| 00:58.49 | Katty | ariel_: i turned the webcam on (= |
| 00:59.06 | ariel_ | really now. |
| 00:59.09 | mikeirc | tzanger: just another day eh? |
| 00:59.33 | tzanger | mikeirc: no, today I was up at a quarter after six shovelling this heavy shit snow since the snowblower was jammed from last week... |
| 00:59.51 | tzanger | after I was 1/2way done my 60 foot lane I said enough with this, went inside and fixed the snowblower |
| 00:59.53 | Katty | ariel_: yup. www.brick.net/~izaah |
| 00:59.57 | mikeirc | Man... I live in Florida... I don't know about that kind of stuff. ;) |
| 01:00.03 | jbAU | are there any recommendations to see if there are extension collisions? |
| 01:00.05 | tzanger | (I had it apart, the engine and wheels were on the porch and the impeller and blower were in my basement) |
| 01:00.15 | *** join/#asterisk joaovianna (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 01:00.22 | tzanger | got that fixed, blew out the driveway and the blower belt finally wore enough to be useless |
| 01:00.40 | tzanger | got the kids to school, did the work thing, grabbed my oldest for lunch (since I didn't have time ot make it this morning, we went out for lunch) |
| 01:00.48 | tuxinator_linux | No snow here in Phoenix, Arizona |
| 01:00.57 | tzanger | stopped by home to grab the old belt, went to TSC to grab a new one, got him back to school and worked the afternoon |
| 01:01.03 | Katty | snow is pretty |
| 01:01.06 | Katty | but cold )= |
| 01:01.09 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 01:01.24 | tzanger | picked up the kids, got home and fixed the blower, blew out the driveway again (snowstorm), made supper, bathed and played with the kids, got them to bed |
| 01:01.25 | jbAU | Katty - can't view your webcam, what's the deal! :) |
| 01:01.28 | tzanger | now i'm here :-) |
| 01:01.45 | tzanger | snow's only pretty in December |
| 01:01.54 | mikeirc | tzanger: And I thought I had a buzy day... hehe |
| 01:02.03 | tzanger | after about the third heavy snowfall and 40 below weather you get tired of it |
| 01:02.09 | tuxinator_linux | jbAU: http://www.onlineconversion.com/clothing.htm |
| 01:02.11 | tzanger | mikeirc: that's a typical day for me, minus the fixing the snowblower stuff |
| 01:02.13 | Katty | jbAU: try www.brick.net/~izaah/webcam.jpg? |
| 01:02.17 | *** join/#asterisk sneak (~sneak@64.220.234.21.ptr.us.xo.net) |
| 01:02.18 | ariel_ | it's kinda cold here tonight in florida. |
| 01:02.28 | ariel_ | ~weather ktmb |
| 01:02.41 | tzanger | oh. you. poor. bugger |
| 01:02.43 | tzanger | 13oC |
| 01:02.47 | mikeirc | ariel_: Yup... pretty chili here in lakeland FL |
| 01:02.50 | ariel_ | 55.0 f is cold for us. |
| 01:02.54 | hermie | ~weather kfnt |
| 01:02.57 | tzanger | mind you it's mild here, about 8 below |
| 01:03.03 | jbAU | tuxinator_linux ok - my shoe is a 10 in US standards |
| 01:03.11 | tuxinator_linux | ~weather kflg |
| 01:03.19 | jbAU | ~weather |
| 01:03.24 | jbAU | ~weather help |
| 01:03.39 | tuxinator_linux | I am in Flagstaff, Arizona right now |
| 01:03.50 | tuxinator_linux | A little snow on the ground |
| 01:03.52 | ariel_ | Katty, your going to brake lots of harts here. |
| 01:04.10 | Katty | ariel_: erm? |
| 01:04.21 | tuxinator_linux | Katty, how's my favorite Vegan? |
| 01:04.30 | Katty | uhh, fine (= |
| 01:04.33 | *** join/#asterisk Rick_Hunter (~rhunter@05-085.008.popsite.net) |
| 01:04.44 | *** join/#asterisk jskcr (~jskcr_@jskcr.user) |
| 01:04.56 | joaovianna | Hi guys ! what * command I use to play a message when one extension is answered ? This file will be only played to the callee (extension) ex. "you are answering a call from florida area !" |
| 01:05.10 | ariel_ | hermie, try spelling with a 23 month old trying to fight for the keyboard. |
| 01:05.11 | Katty | how are you tuxinator_linux? (= |
| 01:06.07 | jskcr | hy all |
| 01:06.11 | tuxinator_linux | Doing pretty well |
| 01:06.19 | Katty | excellent |
| 01:06.20 | jeofrey | where i can find g729 and g723 codec |
| 01:06.23 | jbAU | joaovianna so you want the person picking up the phone to answer it to play a message before it is directed to htem? |
| 01:06.24 | tzanger | ugh I hate this weather |
| 01:06.27 | tuxinator_linux | I'm excited I am going to MEET *. |
| 01:06.27 | tzanger | so dry |
| 01:06.32 | tzanger | this is like my eleventeenth nosebleed |
| 01:06.37 | tzanger | every single year around this time |
| 01:06.42 | Katty | omgwtf*lolz |
| 01:06.57 | tzanger | Katty: you forgot bbq |
| 01:06.57 | jbAU | tzanger nosebleed? damn man u need to move to a country without that type of weather |
| 01:07.08 | mikeirc | Katty: Convulsions? |
| 01:07.08 | tuxinator_linux | Tzanger: cut your fingernails |
| 01:07.10 | Katty | oh yeah |
| 01:07.12 | Katty | i mean |
| 01:07.13 | tzanger | jeofrey: www.digium.com |
| 01:07.16 | tzanger | tuxinator_linux: hahahaha |
| 01:07.18 | tzanger | I ain't pickin it |
| 01:07.27 | tzanger | I hear katty picks her nose on webcam though |
| 01:07.28 | Katty | omgwtfbbq*lolzklkthxbi |
| 01:07.32 | jeofrey | thanks |
| 01:07.40 | tuxinator_linux | I get them too in Phoneix |
| 01:07.55 | Katty | tuxinator_linux: don't spread lies aout me (= |
| 01:07.56 | mikeirc | Katty: you sound like an AOL user.. ;) |
| 01:08.00 | *** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-15-86.w81-248.abo.wanadoo.fr) |
| 01:08.03 | Katty | s/aout/about |
| 01:08.15 | Katty | mikeirc: no, i just mock them |
| 01:08.19 | tuxinator_linux | Katty: lies? |
| 01:08.29 | jbAU | tuxinator_linux i want to see proof! |
| 01:08.31 | *** join/#asterisk brownjava (~jeremy@alanoffice.microcerv.com) |
| 01:08.45 | Katty | tuxinator_linux: 19:30 < tzanger> I hear katty picks her nose on webcam though |
| 01:08.45 | brownjava | anyone have experience with the IAXy thingy? |
| 01:08.51 | tuxinator_linux | Katty lives in a cave... so dark |
| 01:09.03 | tuxinator_linux | oh, yep, sorry about that |
| 01:09.06 | Katty | tuxinator_linux: yup |
| 01:09.16 | tuxinator_linux | brownjava: nope, but I hear it works |
| 01:09.28 | CoaxD | fxs and fxo stuff. pots. *growl* |
| 01:09.29 | mikeirc | Yo Katty: What was up with that refresh rate of you cam? Why so slow? |
| 01:09.48 | Katty | mikeirc: i'm on dialup at home (= |
| 01:09.57 | brownjava | hmm...just bought one and having a monumental amount of trouble with it |
| 01:10.03 | tuxinator_linux | Katty: Ewww |
| 01:10.10 | Katty | mmhmm |
| 01:10.11 | tuxinator_linux | ~dialup |
| 01:10.13 | jbot | somebody said dialup was the worst type of internet connection available |
| 01:10.13 | mikeirc | Katty: No...I didn't know that existed anymore... |
| 01:10.13 | joaovianna | jbAU: Yes, I mean... Suppose you have calls comming from 2 places. One in english and one in portuguese. When the extension is answred I play the message. "please answer this call in portuguese" or "pleas answer this call in english". |
| 01:10.17 | tuxinator_linux | ~modem |
| 01:10.18 | jbot | [modem] (Modulator/Demodulator) A device to turn digital signals to analog ones and back again, so they can be transmitted and translated back to digital at another modem without loss. Used for communication through means of audio, telephone, CB, etc. Random disconnects? S10=255 sure to do the trick! |
| 01:10.19 | psywar | Anyone else having invalid UDP checksum problems with the SPA-2000? |
| 01:10.30 | jbAU | ~udp |
| 01:10.31 | jbot | [udp] only PlayerUpdate, ShotBegin, ShotEnd and GMUpdate |
| 01:10.34 | mikeirc | Katty: You in the boonies?? |
| 01:10.42 | Katty | mikeirc: yub yub |
| 01:10.44 | mikeirc | did I just say boonies? |
| 01:10.48 | mikeirc | :) |
| 01:10.55 | tuxinator_linux | Browjava: What kind of problems are you having? |
| 01:11.53 | tuxinator_linux | Katty is a cuttie, isn't she? |
| 01:11.56 | brownjava | tuxinator_linux: I've provisioned it, tried static and dynamic IPs, pointed it at our asterisk server...but whenever I pick up the phone, I don't hear anything |
| 01:12.08 | brownjava | tuxinator_linux: and the asterisk server has no record of an attempt to dial out |
| 01:12.28 | tuxinator_linux | brownjava: NAT? |
| 01:12.30 | Katty | tuxinator_linux: i'm not a piece of meat though |
| 01:12.57 | brownjava | tuxinator_linux: no NAT involved...device and asterisk are on two separate subnets, but they are 2 hops away from each other and no NAT involved |
| 01:12.57 | tuxinator_linux | Katty: I know, but it doesn't hurt to be a cute smarty. |
| 01:13.15 | tzanger | she's a cuttie? |
| 01:13.23 | tuxinator_linux | brownjava: Have you tried connectly on the same subnet as your * sever? |
| 01:13.29 | brownjava | tuxinator_linux: additionally, I can fire up "FireFly" and use it to connect to the Asterisk server using IAX, so I know IAX should work from here |
| 01:13.32 | psywar | I'm a piece of meat. |
| 01:13.36 | tzanger | she's cute, sure... young though wow |
| 01:13.43 | Katty | tuxinator_linux: i'll outsmart your cutie in a minute |
| 01:13.44 | brownjava | tuxinator_linux: unfortunately no...that's hard to try at the moment |
| 01:13.59 | tuxinator_linux | I'm only 24, so she is not that young to me. |
| 01:14.04 | psywar | I wish I wasn't. The meat is a prison. |
| 01:14.11 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 01:14.13 | tuxinator_linux | brownjava: so IAX is working... hmmm.... |
| 01:14.15 | tzanger | I'm 29 |
| 01:14.22 | ariel_ | psywar, udp means poor network connection. |
| 01:14.31 | tzanger | ariel_: huh? |
| 01:14.31 | psywar | eh? |
| 01:15.26 | tuxinator_linux | brownjava: I'm not sure... let me think about it some more. |
| 01:15.39 | mikeirc | Hmm..I think I'm 25..or 24..no pretty sure I'm 25.. |
| 01:15.39 | tuxinator_linux | man, got something in my eye |
| 01:16.14 | psywar | are you saying it tries to do tcp, then falls back to udp if there's packet loss? |
| 01:16.15 | Katty | tuxinator_linux: is it your finger? |
| 01:16.15 | brownjava | tuxinator_linux: thanks for tryin ;0 it's appreciated |
| 01:16.25 | psywar | or what? |
| 01:16.42 | Katty | gosh |
| 01:16.47 | Katty | that was a smirk |
| 01:16.59 | Katty | tux tab |
| 01:17.01 | Katty | tuxinator_linux: |
| 01:17.06 | Katty | etc |
| 01:17.23 | mikeirc | brownjava: Run ethereal on the server... set to show SIP raw text.. monitor traffic? |
| 01:17.40 | brownjava | mikeirc: ok...will take a few mins to set up |
| 01:17.54 | brownjava | mikeirc: it's not SIP either |
| 01:18.06 | tuxinator_linux | contact fell out, be back in a minute |
| 01:18.11 | mikeirc | brownjava: It might help you diagnose.. oh... hmm.. I'm lost then :) |
| 01:18.22 | tzanger | http://www.web-ee.com/Schematics/WirelessWeather/WirelessWeather.htm |
| 01:18.24 | tzanger | that is too cool |
| 01:18.33 | tzanger | he did a really good job of even explaining the hamming codes |
| 01:18.42 | tzanger | your checksum = the bit # that has the error |
| 01:19.12 | mikeirc | tzanger...thanks..my brain was already fried... I can't look at that.. hehe |
| 01:19.21 | tzanger | mikeirc: it's really nicely done |
| 01:19.33 | Katty | mmm, fried brains |
| 01:19.40 | tzanger | Katty: but you don't eat meat |
| 01:19.50 | Katty | sarcasm, goofball |
| 01:19.57 | tzanger | no shit eh :-) |
| 01:20.01 | mikeirc | Katty = Hanible lector?? |
| 01:20.05 | Katty | ;) |
| 01:20.28 | psywar | nothing like liver with fava beans and a nice chianti |
| 01:20.33 | tuxinator_linux | Ouch, that burns |
| 01:20.41 | xlyz | hi! I'm trying to use asterisk with a Welltech lan phone 302. At the moment it's the only phone connected. Aterisk show a 401 Unauthorized error. Any idea how to make it work? |
| 01:20.47 | Katty | mikeirc: hannibal lector, you mean? |
| 01:21.07 | mikeirc | Anyone seen constantine? I thought it was bad ass myself. |
| 01:21.07 | tzanger | xlyz: sounds like you don't have auth working, why don't you read the asterisk handbook online and see if you can't get that going |
| 01:21.07 | psywar | make sure you correct all spelling errors on IRC. |
| 01:21.09 | Katty | the ol let's cannibalize kids guy? |
| 01:21.47 | Katty | kids just don't do it for me. hehe |
| 01:21.51 | xlyz | tzanger: it's in the wiki? |
| 01:21.52 | mikeirc | Katty.. fried brains.. yep that's the guy.. ;) |
| 01:21.57 | tzanger | katty's not in to veal |
| 01:22.03 | tzanger | xlyz: oh god no, don't go there |
| 01:22.07 | tzanger | ~google asterisk handbook pdf |
| 01:22.29 | psywar | oddly, it's under the "support" menu item on asterisk.org |
| 01:22.33 | mikeirc | this jbot is pretty smart cookie.. |
| 01:22.35 | psywar | instead of under something like "documentation" |
| 01:22.54 | tuxinator_linux | fell out again, be bacl |
| 01:24.13 | mikeirc | Did any of ya'll see constantine yet? |
| 01:24.23 | Katty | i haven't |
| 01:24.24 | puzzled | yes |
| 01:24.29 | Katty | i was too busy updating my kernel |
| 01:24.36 | Katty | over DIALUP |
| 01:24.39 | Katty | :<<< |
| 01:24.40 | mikeirc | I thought it was sweet. ;) |
| 01:24.51 | *** join/#asterisk CarlosMP (~CPerez@64.40.137.60) |
| 01:25.01 | Katty | is it, umm, horror? |
| 01:25.12 | mikeirc | Katty: You need to get some more bandwith girl. :) |
| 01:25.30 | Katty | i know |
| 01:25.30 | *** join/#asterisk CarlosMP_ (~CPerez@64.40.137.60) |
| 01:25.42 | Katty | wanna reg up the asterisk service at work so i can ditch southewestern bell (= |
| 01:25.46 | mikeirc | Katty: Kinda...like the exorcist..but with a matrix twist.. |
| 01:25.52 | tuxinator_linux | Katty needs satalite at least |
| 01:26.07 | Katty | hmm, no |
| 01:26.09 | Katty | cookies |
| 01:26.12 | Katty | i need cookies at least |
| 01:26.19 | mikeirc | My buddy had satalite and it sucked.. never know though. :) |
| 01:26.34 | mikeirc | Katty: What's your favorite kind of cookie? |
| 01:26.38 | Katty | gingersnaps, me thinks |
| 01:26.39 | jbAU | satellite has major lag, especially if you're using pure satelite |
| 01:27.03 | mikeirc | hehe.. I gingesnaps?? that's not a cookie lol :) |
| 01:27.04 | tuxinator_linux | mikeirc: suck more than dialup? |
| 01:27.07 | Katty | dialup has major lag |
| 01:27.10 | Katty | ftp to website PLUS i screen irssi |
| 01:27.35 | jbAU | tuxinator_linux well you use dialup to send with satellite |
| 01:27.37 | mikeirc | Tuxinator_linux: his was pretty bad...but I think the guy installed it was a tard.... you make a pretty valid point though! |
| 01:27.44 | Katty | typing is /not/ fun right now |
| 01:28.04 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 01:28.25 | mikeirc | jbAU: Not with every provider..my bud had service from a company called pegasis.. both way sat tranmsision.. |
| 01:28.39 | jbAU | mikeirc ugh that's even worse |
| 01:28.59 | tuxinator_linux | Katty, what is wrong with typing? |
| 01:29.02 | mikeirc | jbAU: Maybee that's why his was so sorry. ;) |
| 01:29.03 | jbAU | mikeirc the typical lag of a satellite is 500ms+, so if you use it for both sending AND recieving it's painful |
| 01:29.33 | mikeirc | Good to know.. I'll still with old RoadRunner. :) |
| 01:29.39 | tuxinator_linux | it is better for large files |
| 01:29.58 | jbAU | yes the bandwidth is huge, but so is the latency |
| 01:30.01 | mikeirc | I downloaded something the other day... 725KB/sec!!! |
| 01:30.05 | Katty | tuxinator_linux: it's hella laggy |
| 01:30.18 | tuxinator_linux | use both |
| 01:30.20 | odie_flocon | hello all. |
| 01:30.25 | Katty | tuxinator_linux: pls note irssi reference |
| 01:30.27 | jbAU | hi |
| 01:30.33 | tuxinator_linux | woof |
| 01:30.48 | tuxinator_linux | irssi? |
| 01:30.58 | Katty | yes |
| 01:31.16 | Katty | irssi.org |
| 01:31.24 | tuxinator_linux | ~irssi |
| 01:31.25 | jbot | GTK+ based IRC client with GNOME panel support. URL: http://xlife.dhs.org/irssi/ |
| 01:31.25 | odie_flocon | even 1 way on sat is painfull. |
| 01:31.25 | ariel_ | ok back. Just got my baby sleeping. |
| 01:31.43 | Katty | ariel_: yay! |
| 01:32.07 | ariel_ | what did I miss... guess I need to scroll back and do some reading. |
| 01:32.08 | odie_flocon | and just how old is your baby? |
| 01:32.15 | tuxinator_linux | My connect is painfully slow right now. I am using the cullular network. Get about 40k. |
| 01:32.21 | ariel_ | she just turned 23 month old. |
| 01:32.40 | odie_flocon | cool, mine just turned 2 yrs. |
| 01:33.12 | mikeirc | tuxinator_linux: cellular huh? why's that? |
| 01:33.27 | ariel_ | it's just me and the baby tonight. Wife is working and my 17 year is in a sleepover... |
| 01:33.32 | psywar | satellite seems like an excellent way to distribute USENET articles |
| 01:33.40 | odie_flocon | ahhh |
| 01:33.42 | tuxinator_linux | Net anywhere I want it |
| 01:33.55 | ariel_ | psywar, your not getting those errors via sat? |
| 01:33.56 | odie_flocon | that's how most ISP's do it. |
| 01:33.58 | tuxinator_linux | Mikeirc: I only use it when I am out of the house, like now. |
| 01:34.13 | odie_flocon | they get a dedicated sat. downlink. |
| 01:34.17 | mikeirc | tuxinator_linux: Always connected... cool stuff. ;) |
| 01:34.27 | psywar | ariel_: no, on a regular 100Mbps network |
| 01:34.33 | psywar | ethernet |
| 01:34.47 | psywar | I replaced one of the cables with a pre-made, maybe that will fix it |
| 01:35.22 | ariel_ | psywar, like I started to say I have seen them. Mostly is due to bad packets |
| 01:35.26 | psywar | odie_flocon: can I build one out of parts from broken computer equipment? |
| 01:35.40 | tuxinator_linux | It will be nice when someone other than Verizon has wireless broadband in my area. |
| 01:35.44 | psywar | ariel_: I know, the "bad udp checksum" suggests that the packet is corrupted somehow. |
| 01:35.49 | tuxinator_linux | I use T-mobile |
| 01:36.07 | tuxinator_linux | unlimited for 29/month |
| 01:36.13 | jbAU | how would you be if someone stole your wallet whilst you were on the toilet |
| 01:36.16 | ariel_ | since I do some part time work with an voip provider they get them all the time from the Sat users. |
| 01:36.22 | psywar | hey did you folks know that you can use special AT commands to send SMS msgs via cell and other things? |
| 01:36.27 | tuxinator_linux | jbAU: ?? |
| 01:36.30 | odie_flocon | yeah |
| 01:36.30 | mikeirc | Anyone else here using BroadVoice? |
| 01:36.34 | jbAU | tuxinator_linux: http://www.ananova.com/news/story/sm_1301142.html?menu= |
| 01:36.43 | hermie | tuxinator_linux: how fast is the Verizon broadband? |
| 01:37.02 | psywar | I want a satellite downlink for usenet, that would rock. |
| 01:37.21 | odie_flocon | that's funny |
| 01:37.49 | odie_flocon | given the situation they couldn't run after the thief.. |
| 01:38.08 | tuxinator_linux | hermie: 300k |
| 01:38.09 | jbAU | odie_flocon damn straigh! |
| 01:38.16 | odie_flocon | heheh |
| 01:38.18 | hermie | tuxinator_linux: that's not too bad... |
| 01:38.23 | odie_flocon | that is too funny |
| 01:38.46 | hermie | "given the situation" |
| 01:39.15 | hermie | the qvc ladder video on ebaumsworld is hilarious |
| 01:39.38 | Goshen | Anyone here from the UK that can help me test my incoming IAX UK number? 08444846041 |
| 01:39.48 | Goshen | You have to dial from the UK |
| 01:40.07 | tuxinator_linux | jbAU: I lost about 5 pounds the other day in the men's room. |
| 01:40.16 | mikeirc | hermie: Hehe.. that was awsome.. I saw it a couple months ago...lol |
| 01:40.23 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 01:40.56 | mikeirc | What ever happened to SupaFly over at JoeCartoon.com? I haven't been over there in ages? |
| 01:41.13 | mikeirc | That was some funny shiot.. |
| 01:41.23 | Katty | cam is going off soon |
| 01:41.29 | tuxinator_linux | jbAU: Amrican pounds that is |
| 01:41.30 | mikeirc | Nooo.. :) |
| 01:41.45 | tuxinator_linux | Katty: why? That's not cool |
| 01:41.47 | Katty | yes |
| 01:42.01 | mikeirc | Actually..I'm not watching so it's ok... but only this time. :) hehe |
| 01:42.08 | Katty | k |
| 01:42.19 | tuxinator_linux | no problem Katty, just playing |
| 01:42.43 | Katty | heh |
| 01:43.12 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
| 01:43.52 | tuxinator_linux | My contact must be torn, it hurts. |
| 01:44.22 | mikeirc | man.. I love osx.. feel right at home in the console..yet still NATIVELY running photoshop and premier! Too perfect..smething's bound to go wrong. ;) |
| 01:44.37 | *** part/#asterisk CarlosMP (~CPerez@64.40.137.60) |
| 01:46.41 | rob- | goshen: I'm from the uk. do you want me to call you? |
| 01:47.22 | Goshen | yes please |
| 01:47.28 | Goshen | I want to test this IAX2 setup |
| 01:47.32 | Goshen | I don't know if I have it right |
| 01:47.48 | *** join/#asterisk _6Flamez_ (lklk@00045a809589.click-network.com) |
| 01:48.15 | tuxinator_linux | Where did the sun go? |
| 01:48.19 | hermie | if I were knighted, I think i'd move to Canada so I could be called 'sir' |
| 01:48.19 | Ash | Hmm |
| 01:48.26 | Ash | i'm getting a weird error |
| 01:49.01 | jbAU | mikeirc: you should check out iterm |
| 01:49.03 | Ash | 'No application 'Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}}' for extension (from-sip, 91801xxxxxxx, 1) |
| 01:49.07 | Ash | ' |
| 01:49.12 | Ash | where x's are real numbers |
| 01:49.17 | mikeirc | jbAU: iterm? |
| 01:49.20 | Ash | and then it returns a 403 Forbidden |
| 01:49.29 | Ash | has anybody seen this sort of error before? |
| 01:49.31 | jbAU | mikeirc yeah iterm - iterm.sourceforge.net |
| 01:49.43 | Ash | It's weird, because I can make 1800 calls just fine |
| 01:49.48 | Ash | but no local or long distance |
| 01:50.00 | Goshen | Ash: you are in Utah too? :) |
| 01:50.06 | *** part/#asterisk jeofrey (~jeofrey@espeed19-151.brunet.bn) |
| 01:50.19 | Ash | Goshen: haha, just calling a utah number (my cell phone) |
| 01:50.24 | Ash | I just moved from SLC to Las Vegas |
| 01:50.29 | Goshen | ahh |
| 01:50.32 | rob- | goshen: it doesn't work:-( |
| 01:50.35 | mikeirc | jbAU: oohh sweetness. :) gettin it now. Thanks ;) |
| 01:50.38 | Goshen | rob: what does it do? |
| 01:51.08 | Goshen | rob: oh, I got an error message...thank you for trying..I will poke around my configs |
| 01:51.21 | rob- | the service cannot be connected |
| 01:51.26 | Goshen | I got a rejected connect attempt... |
| 01:51.26 | tuxinator_linux | Ash: Vegas is growing like crazy |
| 01:51.47 | Ash | tuxinator_linux: yes it is! |
| 01:51.49 | Ash | it's nuts |
| 01:52.18 | rob- | goshen: what provider is it? |
| 01:52.19 | tuxinator_linux | Ash: Phoenix is right behind in growth |
| 01:52.25 | Ash | interesting |
| 01:53.22 | Ash | man.. this is so weird. |
| 01:53.27 | Goshen | voipuser.org |
| 01:53.32 | Ash | if I dial 9 then 7 digits, it outpulses it to the PRI at least |
| 01:53.39 | tuxinator_linux | You know what, I don't like being on hold |
| 01:53.56 | rob- | I have voipuser working here if you need any info |
| 01:53.56 | Ash | but with a long distance number, it just throws up |
| 01:54.09 | Goshen | you have IAX incoming working? |
| 01:54.16 | tuxinator_linux | How want's to make "on hold" games for * with me? |
| 01:54.43 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 01:54.53 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |
| 01:55.00 | Goshen | tux: like...* says a word backwords like How and you have to guess what it is saying? :) |
| 01:55.04 | rob- | goshen: sorry, I forgot I'm just using sip at the moment |
| 01:55.27 | stepcut | tuxinator_linux: does asterisk currently have the capability to do that ? |
| 01:55.58 | tuxinator_linux | Goshen: The easiest would be to do a triva game, where you key in your option. |
| 01:56.04 | Goshen | rob: ok, I will tweak these configs, thank you for trying, at least I know it is routing to me now |
| 01:56.06 | tuxinator_linux | stepcut: not sure |
| 01:56.08 | tuxinator_linux | stepcut: but we can make it. |
| 01:58.07 | stepcut | tuxinator_linux: I would very much like to do that (and was thinking about it this morning), but I do not currently have the time |
| 01:58.17 | tuxinator_linux | be back in a minute |
| 01:58.43 | rob- | goshen: one thing to look out for in iax.conf is that the name in square brackets is the username for incoming calls. |
| 01:59.56 | tuxinator_linux | Still on hold, grrr |
| 02:00.49 | Goshen | rob: I will look at that |
| 02:01.22 | Goshen | so have that be my number not [voipuser] ? |
| 02:02.49 | Goshen | like this? |
| 02:02.50 | Goshen | [08444846041] |
| 02:02.50 | Goshen | username=******* |
| 02:02.50 | Goshen | secret=******** |
| 02:02.50 | Goshen | type=user |
| 02:02.50 | Goshen | context=inbound-voipuser |
| 02:03.08 | rob- | yep, or make voipuser the username in the address to forward to |
| 02:03.18 | shido6 | where's the type?!?! |
| 02:03.41 | Goshen | what is my url to have it dial in to my box? |
| 02:03.45 | Goshen | type=user? |
| 02:03.52 | Goshen | shido6? |
| 02:05.23 | rob- | goshen: iax2/08444846041:password@hostname/exten |
| 02:05.45 | Goshen | ahh, thats the problem then ok |
| 02:06.00 | BrianR___ | got my first recording back from thevoice.digium.com |
| 02:06.14 | fearnor | brian: allison sounds sexy enough for you? |
| 02:06.33 | rob- | I think username= is only used for outgoing calls with type=peer or type=friend |
| 02:06.45 | BrianR___ | fearnor: Flawless on the first try.. Much better than our old recording, which sounds like a combination used car salesman and bad college radio disc jockey |
| 02:07.06 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 02:07.32 | Katty | bye now |
| 02:07.41 | ariel_ | Katty, good night |
| 02:07.57 | Katty | ariel_: sweet dreams when you get there (= |
| 02:07.58 | tuxinator_linux | Night Katty |
| 02:08.06 | Katty | you too tuxinator_linux! |
| 02:08.12 | *** part/#asterisk Katty (~angela@68.112.15.110) |
| 02:08.25 | tuxinator_linux | :-) |
| 02:08.41 | ACiDV | If I have a TE405 card that when the driver is loaded (wct4xxp) I cannot receive audio from asterisk, that all span alarm are OK but all led are off (and no cable connected), that zttest dont return any result, it's a sign that the card is crashed ? |
| 02:08.43 | Ash | weiiiird |
| 02:08.56 | Ash | I just moved the Dial() rules from my longdistance section into my from-sip section |
| 02:08.59 | Ash | and it magically workd |
| 02:09.00 | Ash | works |
| 02:10.19 | Mavvie | trolls < * |
| 02:11.09 | tuxinator_linux | ACiDV: Can't help you there |
| 02:11.54 | tuxinator_linux | ACiDV: sounds like no power |
| 02:12.02 | ariel_ | ACiDV, did you so make in the zapata and libpri first before you did the make in asterisk? |
| 02:12.07 | tuxinator_linux | ACiDV: Has it worked before? |
| 02:12.21 | ariel_ | ./so/do |
| 02:14.07 | ACiDV | Yes it's work before ... stop working today |
| 02:14.31 | ACiDV | and try with 1.0.6 and today cvs (update, make clean, install, etc...) |
| 02:14.39 | tuxinator_linux | The server hasn't moved at all causing the card to come loose? |
| 02:15.25 | ACiDV | No :( and the server is at 800km of my current location :| |
| 02:15.41 | tuxinator_linux | hmm |
| 02:15.58 | ACiDV | have both te405 and a tdm22b and if I load all without wct4xxp all work fine... |
| 02:16.28 | easydone | hmmm "attempting native bridge" when "canreinvite=no" |
| 02:16.37 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 02:16.42 | easydone | thought canreinvite would prevent that |
| 02:16.51 | Dr-Linux | tzanger |
| 02:17.23 | tuxinator_linux | Welcome Dr L |
| 02:17.47 | ACiDV | very weird :( |
| 02:18.22 | tuxinator_linux | I haven't had the opportunity to play with those drivers yet. Soon.... |
| 02:18.58 | tuxinator_linux | ACiDV: You can see 800km away? |
| 02:19.14 | freat | hello |
| 02:19.32 | ACiDV | about light ? :) no... :) cam in the same room :P |
| 02:19.39 | ACiDV | led oops |
| 02:19.47 | tuxinator_linux | You know what's worse than being on hold, when they repeat the hold music every 5 minutes. |
| 02:19.53 | freat | anyone here familiar w/ polycom phones? we've got a bunch of IP 500s... need to figure out if I can make a line appearance to monitor an extension... |
| 02:20.38 | *** join/#asterisk CarlosMP_ (~CPerez@64.40.137.60) |
| 02:22.42 | *** join/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl) |
| 02:25.30 | loko | in sip.conf, should type=friend or users ? |
| 02:26.58 | tuxinator_linux | ~weather KFLG |
| 02:28.18 | tuxinator_linux | ~weather KIWA |
| 02:28.21 | jbAU | loko loko - use friends |
| 02:36.21 | tuxinator_linux | ~weather KSJC |
| 02:36.44 | jbAU | ls -la |
| 02:37.58 | tuxinator_linux | ls -halt |
| 02:39.44 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 02:39.44 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.6 Released || Dev Conf 1PM CST MARCH 3rd -> IAX2/guest@66.250.68.194/996 || ClueCon Dev Conf June 8-10th more coming soon.... |
| 02:42.10 | *** join/#asterisk phantam (~phantam@72.252.15.235) |
| 02:42.11 | phantam | :) |
| 02:42.14 | phantam | im a happy camper |
| 02:42.26 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 02:42.31 | phantam | took a friendly gentoo-dev to get that darn h323 thang working |
| 02:42.46 | Mneumonic | anyone know how to enable call transferring in X-Lite? |
| 02:43.08 | phantam | had to mask out the newer pwlibs and oh323... and update to 0.6.5 ast-oh323 and it works |
| 02:43.12 | phantam | now just gotta find a way to test it |
| 02:43.13 | ariel_ | Mneumonic, only via the dial rules adding the Tt option. |
| 02:43.49 | Mneumonic | aliel_ . you mean in *'s dialplan or in the software |
| 02:43.58 | ariel_ | asterisk dial plan |
| 02:44.32 | Mneumonic | and then X-Lite picks up that code and allows it? or i have to dial a code when the call is on hold? |
| 02:45.00 | phantam | im still trying to figure out how to get fwd to work |
| 02:45.02 | ariel_ | Mneumonic, I don't like using it. but it use the # key to make the transfer. |
| 02:45.55 | Mneumonic | is there any easier way? I have 3 Sipura 841's on their way to me... so di i have to have the code in the dialplan for those? or just for the softphone? |
| 02:46.01 | ariel_ | phantam, fwd works fine. what is the problem there are many examples on the wiki and there site for it to work. |
| 02:46.08 | phantam | yes i know |
| 02:46.14 | phantam | i think its because the box is natted |
| 02:46.19 | phantam | and i know that it says there are issues with nat |
| 02:46.30 | phantam | im gonna setup a public ip on it to see if i can get it to work |
| 02:46.44 | ariel_ | Mneumonic, softphone like xlite only if you upgrade them to x-ten pro it has transfer on it. |
| 02:47.06 | ariel_ | phantam, go to there iax setup. |
| 02:47.08 | *** join/#asterisk CarlosMP_ (~CPerez@64.40.137.60) |
| 02:47.10 | Mneumonic | is there any other softphone's for windows that do it for free? |
| 02:47.12 | phantam | mmm |
| 02:47.17 | phantam | not sure if iax is compiled lol |
| 02:47.23 | phantam | i think thats one of the modules i had to noload |
| 02:47.47 | ariel_ | iax in my view is very important for asterisk. |
| 02:47.51 | phantam | hehe |
| 02:47.52 | phantam | i agree |
| 02:48.29 | ariel_ | phantam, belive me fwd via sip and nat works. I have it working on my system and I am behind a wrt54g firewall/nat router. |
| 02:48.31 | phantam | hmmm i didnt noload iax |
| 02:48.34 | phantam | maybe it is still there |
| 02:48.46 | phantam | hehe wrt54g sexy little routers eh |
| 02:48.54 | phantam | u running linksys firmware or hacked? |
| 02:49.05 | gambolputty | anyone compiled * on a wrt54gs? |
| 02:49.18 | ariel_ | well in 12 minutes it's the end of an error. NYPD Blue's last show.... bummer. |
| 02:49.23 | phantam | u dont compile on a wrt |
| 02:49.28 | phantam | u compile in a chroot on a real server |
| 02:49.32 | phantam | and then copy the bins over |
| 02:49.40 | gambolputty | have you done this yet then? |
| 02:49.48 | phantam | not yet no |
| 02:49.48 | phantam | lol |
| 02:49.54 | phantam | ariel_: how do i check if iax is laoded |
| 02:50.06 | phantam | nm |
| 02:50.07 | phantam | its there |
| 02:50.08 | phantam | hehe |
| 02:50.13 | phantam | iax2<tab> |
| 02:50.18 | phantam | that tab helps alot |
| 02:50.37 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4119108.sympatico.ca) |
| 02:50.48 | ariel_ | <PROTECTED> |
| 02:51.11 | ariel_ | <PROTECTED> |
| 02:55.14 | *** join/#asterisk jskcr (~jskcr_@jskcr.user) |
| 02:59.11 | *** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net) |
| 03:00.28 | *** join/#asterisk CarlosMP_ (~CarlosMP@64.40.137.60) |
| 03:03.09 | phantam | lol |
| 03:03.18 | phantam | guess hasnt been 10 minutes yet registration refused for iax |
| 03:03.19 | phantam | lol |
| 03:05.06 | modulus_ | HAHAHAHAAA |
| 03:06.37 | phantam | ? |
| 03:07.16 | Nugget | no more beer for modulus_. |
| 03:07.18 | xlyz | tzanger, ariel_ just to let you know that the phones now work :) |
| 03:08.14 | ariel_ | xlyz, great. |
| 03:08.26 | phantam | iax is fu*kin sick |
| 03:09.22 | padf00t | question: is it normal to see nothgin when you do "h.323. show codecs" in asterisk CLI |
| 03:10.38 | *** join/#asterisk CarlosMP_ (~CarlosMP@64.40.137.60) |
| 03:10.47 | ariel_ | h.323 is not a codec. codec is ulaw, alaw, g.729,GSM. |
| 03:10.50 | phantam | ill tell ya |
| 03:11.18 | phantam | oh323 is diff nevermind |
| 03:11.39 | padf00t | hmmm |
| 03:11.58 | roamer323 | phatam - my iax2 registration with fwd is solid for days and days |
| 03:12.20 | padf00t | i do "h.323 show codecs" command in asterisk CLI ... and i dont see any codecs displayed. so is this normal behavior? shouldnt i be seeing codecs anabled? |
| 03:12.24 | *** join/#asterisk riksta (~rick@client-82-13-19-160.brhm.adsl.virgin.net) |
| 03:12.37 | padf00t | h.323 show codecs Show enabled codecs |
| 03:12.50 | padf00t | thats what i see in the help h.323 |
| 03:14.05 | *** join/#asterisk blitztang (~blitzrage@d141-234-145.home.cgocable.net) |
| 03:14.18 | blitztang | Moc: you around? |
| 03:14.24 | ariel_ | show codecs |
| 03:15.27 | blitztang | if anyone has gotten the sip_notify stuff to work with a 7960, please find me in #asterisk-doc or any other #asterisk channel... thanks! |
| 03:15.31 | *** part/#asterisk blitztang (~blitzrage@d141-234-145.home.cgocable.net) |
| 03:16.11 | phantam | weird |
| 03:16.13 | phantam | this used to work |
| 03:16.27 | phantam | tried calling from sip phone to a extension (operator) and i get |
| 03:16.30 | phantam | <PROTECTED> |
| 03:17.26 | *** join/#asterisk harryvv (~none@S010600055d210201.vs.shawcable.net) |
| 03:22.23 | *** join/#asterisk jterrero (~jterrero@mcse-irc.isys-networks.com) |
| 03:22.47 | tuxinator_linux | Do you think Cisco will release a color SIP speaking phone? |
| 03:23.56 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 03:24.05 | jterrero | will they not release firmware for 7970s to support sip? |
| 03:24.06 | jterrero | its color |
| 03:24.22 | jterrero | and you can still use that phone with ast |
| 03:24.22 | _Vile | zeek, cool argument |
| 03:24.34 | _Vile | err, nm, was reading from days ago |
| 03:24.45 | tuxinator_linux | I think it only speeks SCCY |
| 03:24.53 | jterrero | sccp |
| 03:24.55 | jterrero | skinny |
| 03:25.03 | jterrero | you can still use it with ast |
| 03:25.04 | tuxinator_linux | ya, that's it |
| 03:25.12 | *** join/#asterisk DHuang (~DHuang@adsl-102-99.swiftdsl.com.au) |
| 03:25.12 | tuxinator_linux | he he ;-) |
| 03:25.14 | _Vile | no sip for 7970, u sure? |
| 03:25.17 | DHuang | hi! :-) |
| 03:25.20 | jterrero | pretty sure |
| 03:25.26 | *** join/#asterisk FuriousGeorge (~FuriousGe@ool-43516ebb.dyn.optonline.net) |
| 03:25.27 | _Vile | hm |
| 03:25.28 | tuxinator_linux | I haven't seen it |
| 03:25.31 | jterrero | they have not released the firmware for it yet |
| 03:25.39 | DHuang | anyone know the defaul AST_MAX_EXTENSION value? |
| 03:25.44 | _Vile | nice, delaying for call manager sales |
| 03:25.58 | tuxinator_linux | It looks fun to play with |
| 03:26.15 | _Vile | nothing much different except backlit, color screen, a few other cool features |
| 03:26.20 | Juggie | its the phone they use on 24 |
| 03:26.28 | Juggie | if anyone watches that. |
| 03:26.46 | _Vile | I've seen them, I use strictly 7960 and 7940 because of the price points |
| 03:26.47 | ariel_ | they use that phone in allot of shows. including Vegas |
| 03:26.56 | jterrero | yeah |
| 03:26.58 | DHuang | ya.. CISCO sponsor them |
| 03:27.01 | FuriousGeorge | i have read a little about asterisk but never used it or seen it work. whats the concensus on its viability for a small business. |
| 03:27.16 | FuriousGeorge | specifically one with ample bandwidth and ptentially 4 voip lines |
| 03:27.18 | jterrero | me too, have bout 8 7940s at work, the rest are 60's |
| 03:27.19 | tuxinator_linux | I was looking at the 7960 |
| 03:27.24 | jterrero | cant beat 240 a pop |
| 03:27.24 | jterrero | heh |
| 03:27.26 | phantam | argggggg |
| 03:27.27 | _Vile | Furious, works fine |
| 03:27.29 | phantam | is fwd down? |
| 03:27.31 | phantam | or it just me |
| 03:27.38 | _Vile | I have 8 7940s and 1 7960 for the operator |
| 03:27.43 | _Vile | managed at work |
| 03:27.51 | Ash | FuriousGeorge: I'm working at a subsidiary of another company that just started up, and asterisk is perfect for our small office |
| 03:27.53 | tuxinator_linux | I have to go, be back later tonight |
| 03:27.57 | FuriousGeorge | _vile_: low masintenence? |
| 03:27.59 | Ash | we bought a pack of 7960s and we're going |
| 03:27.59 | ariel_ | FuriousGeorge, great system for that if you want to try it out get an iso called Asterisk@home will setup you up quickly |
| 03:28.08 | tuxinator_linux | Easy to put SIP on the ciscos? |
| 03:28.11 | _Vile | Furious, except for a few firmware updates every couple of months... |
| 03:28.12 | Juggie | i dont like the cisco phones. |
| 03:28.13 | jterrero | yeah |
| 03:28.17 | Juggie | they suck pretty much. |
| 03:28.18 | Ash | the ciscos are easy to put sip on |
| 03:28.23 | phantam | if i dial 8,411 it ring a few times and then just hangs up and says service unavailable in sjphone |
| 03:28.26 | _Vile | easy management |
| 03:28.28 | jterrero | just use a tftp server to upload the firmware |
| 03:28.32 | _Vile | yup |
| 03:28.32 | jterrero | you can get it from cisco.com |
| 03:28.33 | tuxinator_linux | cool |
| 03:28.38 | tuxinator_linux | See ya |
| 03:28.43 | Juggie | except, theres no way to reset the phone settings without the phone booting |
| 03:28.43 | jterrero | can also get some example config for your SIPMACADD files |
| 03:28.50 | Juggie | thats one problem |
| 03:28.57 | phantam | neone? |
| 03:28.58 | Juggie | conferencing only allows 3 lines. |
| 03:29.14 | _Vile | Juggie, true... unless you dialin to a conference |
| 03:29.16 | ariel_ | phantam, what is your fwd number let me call you for a quick test. |
| 03:29.18 | _Vile | same with 3way calling |
| 03:29.26 | phantam | its |
| 03:29.35 | DHuang | AST_MAX_EXTENSION = 80 |
| 03:29.47 | harryvv | anyone know the laws and regs concerning of transfering a existing home phone number to a pstn voip termination wholesaler? I read that most in the states cannot. |
| 03:29.52 | phantam | 617495 |
| 03:30.14 | FuriousGeorge | i know enough about linux to install and run gentoo. its actually pretty easy cuz you never have to worry about dependencies. i work in IT but strictly windows pl;atform. can someone like me deploy this? |
| 03:30.29 | phantam | yes furius |
| 03:30.33 | _Vile | harryvv, you should be able to port your number as long as it's local |
| 03:30.34 | phantam | i do bout the same |
| 03:30.35 | Ash | FuriousGeorge: if you're just using VoIP providers it's a cinch |
| 03:30.37 | phantam | and im deployin it |
| 03:30.38 | _Vile | doesn't matter if it' |
| 03:30.39 | FuriousGeorge | my experience with linux is that it takes forever and a lot of time at forums to get anything working |
| 03:30.41 | ariel_ | strange got a ring then fast busy. |
| 03:30.41 | _Vile | s voip or not |
| 03:30.42 | phantam | and theres people here to help if anythin |
| 03:30.48 | phantam | same here |
| 03:30.51 | phantam | when i call anything |
| 03:31.03 | FuriousGeorge | a good community is important, and this is good to see |
| 03:31.08 | phantam | <PROTECTED> |
| 03:31.11 | harryvv | Furios, that was my cast but also partly it comes from knowhow and experaince. |
| 03:31.16 | ariel_ | phantam, mine is working give it a call. 65342 |
| 03:31.20 | harryvv | cast=case |
| 03:31.32 | phantam | ring... fast busy |
| 03:31.37 | phantam | it might be my setup |
| 03:31.45 | ariel_ | sounds like it |
| 03:31.54 | phantam | but i coppied it off there site |
| 03:32.04 | phantam | ariel can u compare u'res to mine for me |
| 03:32.05 | phantam | pls |
| 03:32.09 | NormAst | Hay, when I call the echo test with my cell phone I don't get any echo.. Cool eh? Echo Can. at work! |
| 03:32.13 | ariel_ | sure use pastebin.ca |
| 03:32.19 | phantam | k |
| 03:32.20 | phantam | hold |
| 03:32.29 | _Vile | I have random issues with Echo |
| 03:32.37 | _Vile | bitch of a problem to track down on 96 lines |
| 03:32.46 | FuriousGeorge | gow much cpu is required for 4 lines of concurrent conversation. my understanding is that VoIP is cpu intensive. |
| 03:32.48 | harryvv | arial, know of a did wholsale provider for bc canada? |
| 03:32.52 | FuriousGeorge | gow=how |
| 03:33.01 | *** join/#asterisk lancey (Shady@support.net1.cc) |
| 03:33.03 | lancey | hi guys |
| 03:33.04 | NormAst | yea... I know.. 115 lines. |
| 03:33.05 | ariel_ | harryvv, slepp |
| 03:33.12 | harryvv | slepp? |
| 03:33.16 | lancey | does anyone know if g729 works okay on freebsd 5.3? |
| 03:33.19 | _Vile | Furious, I'd use something around the 1.8ghz range and wouldn't lose sleep about it |
| 03:33.30 | _Vile | hell 1.4 would work for 4 lines, depending on what card you're using |
| 03:33.38 | lancey | as it is binary only... |
| 03:33.39 | lancey | ? |
| 03:33.39 | _Vile | something low budget |
| 03:33.58 | FuriousGeorge | _Vile_: is system memory speed important or will ddr266 do |
| 03:34.00 | ariel_ | hell a 600mhz celeron works for 10 voip calls and sip phones. |
| 03:34.06 | _Vile | 266 would work |
| 03:34.21 | _Vile | yup |
| 03:34.36 | harryvv | arial what do you mean by slepp |
| 03:34.41 | FuriousGeorge | ariel_> if i buy sip phones they take on a lot of the work? |
| 03:34.52 | ariel_ | harryvv, he is providing ca did's and termination. |
| 03:35.04 | _Vile | furious, no work |
| 03:35.13 | _Vile | use a minimal system |
| 03:35.15 | *** join/#asterisk fearnor (~alex@66.250.55.42) |
| 03:35.23 | _Vile | 4 calls is nothing |
| 03:35.25 | harryvv | thanks |
| 03:35.26 | NormAst | I can provide Toronto, Hamilton, and Barrie DID's |
| 03:35.54 | harryvv | arial do you know slepps email by chance? |
| 03:37.04 | FuriousGeorge | shhh |
| 03:37.17 | Mavvie | zttool doesn't like screen. |
| 03:37.21 | Mavvie | screen doesn't like zttool |
| 03:37.27 | Mavvie | what a bunch of children. |
| 03:37.40 | modulus_ | mavvie, asterisk cli doesn't like EOT |
| 03:38.15 | fearnor | i don't like screen...screen doesn't like me :( |
| 03:38.27 | modulus_ | screen likes EOT |
| 03:38.31 | FuriousGeorge | does asteriks in any way facilitate the use of softphones with providers who dont offer them? |
| 03:38.31 | Mavvie | modulus_: wasn't that when it was in "asterisk -r" mode? |
| 03:38.42 | Mneumonic | hey gues... would this be a extemely affordable way to do PoE for phones like the Polycom IP 300? http://www.nycwireless.net/poe/ |
| 03:38.43 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 03:39.03 | modulus_ | mavvie, yes. |
| 03:39.25 | modulus_ | cli does like SIGINT though |
| 03:39.27 | Mavvie | modulus_: I have never been able to reproduce it. I can ^D as much as I want, it stays up. |
| 03:39.37 | modulus_ | [01-Mar:19:37 modulus_] mavvie, asterisk cli doesn't like EOT |
| 03:39.38 | fearnor | mneu: no. go buy a proper PoE switch. |
| 03:39.40 | modulus_ | "doesn't" |
| 03:40.00 | Mavvie | oh. I interpret "doesn't like" as in "goes totally haywire" |
| 03:40.14 | Mavvie | instead of "handles it different" :-) |
| 03:40.28 | modulus_ | it just doesn't acknowledge it |
| 03:40.35 | ariel_ | harryvv, no but he is the one that runs pastebin.ca |
| 03:40.36 | modulus_ | SIGINT works though |
| 03:42.18 | Mavvie | yes, and it is mpg123 which segfaults :-) |
| 03:42.27 | *** join/#asterisk DavidFisherman (~davidfish@ip68-111-78-199.oc.oc.cox.net) |
| 03:43.34 | DavidFisherman | can someone please help, I can't figure out how to get my IAXPhone to dial out |
| 03:43.38 | DavidFisherman | please |
| 03:44.53 | FuriousGeorge | does anyone know if using asterisk would in any way facilitate the use of softphones with providers like vonage that dont support them? |
| 03:45.15 | FuriousGeorge | *that dont suppor them at flat pricing ;) |
| 03:45.35 | mgth | furiousgeorge: asterisk does not work with vonage |
| 03:45.45 | FuriousGeorge | ahhh |
| 03:45.55 | mgth | furiousgeorge: asterisk does work with broadvioce |
| 03:45.57 | FuriousGeorge | b/c of the locked hw |
| 03:45.58 | mgth | *voice |
| 03:46.08 | FuriousGeorge | but bv does not hgave local number portability |
| 03:46.21 | mgth | furiousgeorge: No it does not |
| 03:46.41 | FuriousGeorge | which is a big problem in and of itself |
| 03:46.59 | FuriousGeorge | vonage wont work with any "locked" voip provider, will it |
| 03:48.27 | Zaw | does anyone know of a vendor that sells the Linksys PAP2 voip adapter without any 3rd party firmware like vonage, etc. |
| 03:48.27 | Zaw | ? |
| 03:49.02 | FuriousGeorge | i meant to ask: asterisk wont work with locked voip providers, right? |
| 03:49.45 | FuriousGeorge | zaw, could you flash the firmware on one of those, or are you trying to avoid doing that? |
| 03:50.25 | Zaw | FuriousGeorge: i'm just trying to stay legal :) |
| 03:51.08 | FuriousGeorge | oh, i didnt realize its ilegal. who gonna tell on you? |
| 03:51.36 | Zaw | i dunno if it's illegal or not |
| 03:52.38 | FuriousGeorge | i dont know anything about this stuff, but i assume linksys sells the same model for non vonage use, and the vonage one must allow for firmware updates |
| 03:52.52 | _Vile | probably not legal, but noone would know |
| 03:53.01 | _Vile | unless you told someone |
| 03:53.06 | _Vile | like you just announced |
| 03:53.12 | FuriousGeorge | lol |
| 03:53.13 | Zaw | i'd rather just but the non-vonage one |
| 03:53.15 | *** join/#asterisk sleepy_one (~chatzilla@dhcp16632045.neo.rr.com) |
| 03:53.18 | _Vile | yeah |
| 03:53.23 | _Vile | I buy PAP2-NA's |
| 03:53.25 | sleepy_one | hey all :) |
| 03:53.25 | SexyKen | Hey guys -- anyone do custom programming for Asterisk? |
| 03:53.33 | _Vile | unlocked to providers |
| 03:53.35 | FuriousGeorge | i pay all my taxes, always |
| 03:53.48 | sivana | SexyKen: yes |
| 03:54.01 | SexyKen | •sivana• Do you represent a company. |
| 03:54.06 | _Vile | but I'm a vendor |
| 03:54.08 | sivana | SexyKen: yes :) |
| 03:54.16 | _Vile | as a customer, I'd suggest becoming a vendor |
| 03:54.17 | SexyKen | sivana - What company please? URL? |
| 03:54.27 | sivana | SexyKen: www.voctel.com |
| 03:54.38 | sleepy_one | pardon my ignorance I'm trying to use faxing on my X100p and when I call rxfax to RX a fax I get: "pbx.c:1280 pbx_extension_helper: No application 'rxfax' for extension (default, s, 3)" any ideas? |
| 03:54.40 | sivana | SexyKen: www.aspworld.com |
| 03:54.51 | _Vile | I like aspworld |
| 03:54.56 | _Vile | I used to use it |
| 03:55.03 | _Vile | until I moved to php and c |
| 03:55.06 | sivana | :P |
| 03:55.21 | _Vile | I used to own jsworld.com |
| 03:55.21 | sivana | cool |
| 03:55.21 | _Vile | and netpedia.com |
| 03:55.28 | mikegrb | sleepy_one: rxfax is an addon, it doesn't come with asterisk |
| 03:55.30 | _Vile | and pcgaming.com |
| 03:55.44 | DavidFisherman | Hi guys, I can't get asterisk to play the welcome msg, can someone help please? |
| 03:55.46 | _Vile | I don't own pcgaming.com anymore though, and jsw and netpedia are long gone |
| 03:56.00 | sivana | what was netpedia? |
| 03:56.00 | *** join/#asterisk Tray (~traytray@ip24-253-102-200.lv.lv.cox.net) |
| 03:56.09 | _Vile | everything in addition to javascript |
| 03:56.13 | sivana | DavidFisherman: post what you want to pastebin |
| 03:56.13 | _Vile | html, css etc |
| 03:56.19 | _Vile | when the fruit was ripe |
| 03:56.22 | sivana | i c |
| 03:56.28 | Tray | Hey does anyone know how to get a t100p to do hdlc in debian? |
| 03:56.29 | _Vile | to compete w/ webpedia |
| 03:56.34 | SexyKen | sivana -- that doesn't say much about custom programming for Asterisk. |
| 03:56.34 | _Vile | which was owned by mydesktop |
| 03:56.41 | sivana | SexyKen: what do you need? |
| 03:56.45 | _Vile | our competition, then bought out by internet.com |
| 03:56.50 | _Vile | for millions |
| 03:57.05 | SexyKen | sivana -- It isn't quite explainable in 3 lines. |
| 03:57.13 | DavidFisherman | sivana, pastebin?> |
| 03:57.14 | _Vile | not me though, I was offerred $140k by developer.com and didn't take it like I should've |
| 03:57.20 | _Vile | dot com bust |
| 03:57.22 | sivana | ~pastebin |
| 03:57.23 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 03:57.42 | sivana | SexyKen: we do programming for * in C, MySQL, PostgreSQL |
| 03:58.02 | SexyKen | MySQL isn't a programming language though. |
| 03:58.04 | sivana | SexyKen: unless you can give me your requirements, what can I say? :) |
| 03:58.25 | _Vile | Sexy, it really is, it's a programming language for variable retrieval and placement |
| 03:58.30 | _Vile | depending on its use |
| 03:58.50 | _Vile | SQL is a language |
| 03:58.50 | SexyKen | sivana -- AIM or e-Mail address? |
| 03:58.50 | SexyKen | Not a programming language. |
| 03:58.51 | _Vile | I beg to differ, we can argue if you want |
| 03:58.57 | sivana | fire it off in an email: info@voctel.com |
| 03:59.01 | sivana | hehe |
| 03:59.06 | sivana | I love SQL arguments |
| 03:59.17 | mamcinty | haha |
| 03:59.31 | Tray | Damn this t100p card and hdlc and linux to hell |
| 03:59.31 | Luhiwu | Anyone knows if the called number is set into any variable? i need to use goto based on the called id number and i can't find the right variable. ${DIALEDPEERNUMBER} doesn't seems to work. |
| 03:59.45 | sivana | most of any data manipulation code is done in SQL, so kinda is a progamming language |
| 03:59.49 | _Vile | It's a Structured Query Language, no different than storing information in an array |
| 03:59.55 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfk0r.dialup.mindspring.com) |
| 03:59.57 | *** join/#asterisk W1thdraw (~Withdraw@ip68-5-125-44.oc.oc.cox.net) |
| 04:00.17 | _Vile | except more powerful |
| 04:00.28 | sivana | and stored procs are compiled too |
| 04:00.33 | _Vile | yep |
| 04:00.52 | _Vile | depending on the db |
| 04:01.13 | hermie | arrays aren't relations... they don't have foreign keys or NOT NULLs |
| 04:01.18 | sleepy_one | gnite all |
| 04:01.23 | _Vile | they do if you create them |
| 04:01.28 | sleepy_one | I need more sleep |
| 04:01.31 | sivana | hehe |
| 04:02.13 | Tray | So sethdlc hdlc0 cisco returns Error No such device (19) |
| 04:02.28 | *** join/#asterisk Sky-Knight (~Sky-Knigh@ip-66-218-240-53.cableaz.com) |
| 04:03.45 | *** part/#asterisk Sky-Knight (~Sky-Knigh@ip-66-218-240-53.cableaz.com) |
| 04:03.50 | *** part/#asterisk DavidFisherman (~davidfish@ip68-111-78-199.oc.oc.cox.net) |
| 04:03.55 | _Vile | im going to smoke, noone's arguing me :( |
| 04:04.33 | *** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com) |
| 04:04.43 | sivana | :) |
| 04:05.25 | Tray | damn it I want to setup hdlc over my t100p but the damn interface doesn't exist and I can't figure it out |
| 04:06.57 | _Vile | programming by definition is giving a computer a set of instructions to follow, any language that tells a computer a set of instructions by that very definition, is a programming language. |
| 04:07.13 | _Vile | fight that, i'll be back in 5 |
| 04:07.18 | *** part/#asterisk CarlosMP_ (~CarlosMP@64.40.137.60) |
| 04:07.52 | sivana | discuss |
| 04:08.18 | Juggie | you know what would be sweet... |
| 04:08.21 | Juggie | patch the cli output |
| 04:08.34 | Juggie | which i believe runs through ast_cli(...) for the most part |
| 04:08.38 | Juggie | to output in csv format |
| 04:08.41 | Juggie | to make it easy to parse |
| 04:08.53 | Zaw | _Vile: 'shutdown' and 'halt' are an instruction for a computer to follow, but are by no means programming languages |
| 04:08.59 | Juggie | that would be very useful for me. |
| 04:10.02 | _Vile | Zaw, depends on the language.. sh programming/scripting can use these two functions and be considered a language |
| 04:10.12 | phantam | mmmm |
| 04:10.31 | Dr-Linux | hello |
| 04:10.42 | Zaw | _Vile: so when a user clicks Start - Shutdown in Microsoft Windows, he's writing in a programming language? |
| 04:11.29 | Dr-Linux | me and my friend are connected to softphones, using sip but have live ip, he can hear me good, but i can't hear him ? |
| 04:11.31 | _Vile | Zaw, no, but the user is instructing a language to perform those actions, mostly a modified C/ASM combination |
| 04:11.36 | Dr-Linux | what could be the problem ? |
| 04:11.53 | _Vile | that's an action and not a language, the user isn't writing a program to say > start, shut down |
| 04:11.59 | Zaw | _Vile: yes, he is giving a computer a set of instructions to follow. he isn't doing any programming, though. |
| 04:12.39 | _Vile | for that to be considered a language, he'd need to write a program to click start and shut down |
| 04:12.47 | *** join/#asterisk tessier_ (~treed@210.245.113.94) |
| 04:12.49 | _Vile | and that would be language |
| 04:12.55 | Dr-Linux | Zaw: can you answer me question ? |
| 04:13.10 | Zaw | _Vile: i'm just going by your definition, 'programming by definition is giving a computer a set of instructions to follow' |
| 04:13.19 | _Vile | Zaw, include the rest of it |
| 04:13.50 | FuriousGeorge | if i could find a Ma Bell phone provider with unlimited call forewarding, it could solve the local number portability problem, right? |
| 04:14.07 | sivana | I told my computer to jump off a bridge, but it didn't. Did I just do programming? :) |
| 04:14.12 | Dr-Linux | me and my friend are connected to softphones, using sip but have live ip, he can hear me good, but i can't hear him ? |
| 04:14.25 | _Vile | sivana, no, your computer is in err, reboot and try again |
| 04:14.33 | Dr-Linux | what could be the problem ? |
| 04:14.34 | sivana | hehe |
| 04:14.57 | Zaw | _Vile: i interpreted your sentence differently than you had articulated it, i suppose |
| 04:15.54 | FuriousGeorge | allow me to elaborate, if i could just foreward all the phone calls on the number i dont want to lose to a number on my asterisk server, i could have my cake and eat it too |
| 04:16.08 | FuriousGeorge | n'est pas? |
| 04:16.39 | _Vile | Zaw, simply put, if you are writing code, you are writing a language -- you are giving instructions to a computer to follow, within certain execution parameters.. that's what coding's all about.. my consensus is, that if you are writing code to instruct a computer to do something, you are programming, thus, you are programming in a language that a computer can understand, and therefore every language that instructs a computer to do something, is a progra |
| 04:17.13 | _Vile | be it interpreted or binary assembly |
| 04:18.32 | Dr-Linux | you pplz cant see my messages or what ? |
| 04:18.47 | FuriousGeorge | for instance: if computer crash = true then jump of bridge |
| 04:18.54 | FuriousGeorge | else, reboot |
| 04:19.02 | _Vile | need legs |
| 04:19.14 | FuriousGeorge | while no legs do: |
| 04:19.19 | _Vile | sit |
| 04:19.19 | FuriousGeorge | order legs online |
| 04:19.20 | FuriousGeorge | end |
| 04:19.22 | FuriousGeorge | lol |
| 04:19.33 | sivana | Dr-Linux: try the mailing list. Ppl here either don't know the answer, or don't feel like debugging |
| 04:19.39 | sivana | we do it all day long..hehe |
| 04:19.52 | FuriousGeorge | _vile: did you get what i said before about solving the local number prtabiolity problem? |
| 04:20.14 | _Vile | Furious, no, too caught up in explaining what programming was |
| 04:20.23 | _Vile | say again |
| 04:20.47 | FuriousGeorge | what if i could find somewhere to port my number which had unlimited call forewarding |
| 04:20.58 | FuriousGeorge | then foreward all my incomming calls to my asteriks server |
| 04:21.02 | _Vile | then you could forward to any number that you want |
| 04:21.07 | *** join/#asterisk nix000 (~nixman@66.11.190.225) |
| 04:21.14 | _Vile | but you incur the cost of the forwarding charges if it is LD |
| 04:21.26 | _Vile | make sure the # you forward to is in your local eas |
| 04:21.30 | FuriousGeorge | but it wouldnt be |
| 04:21.32 | nix000 | anyone ever dealt with freeradius in here ... i am about to pull my hair ! |
| 04:21.40 | _Vile | then you will incur long distance charges |
| 04:21.58 | FuriousGeorge | i forgot: does astersk work with pstn providers or is it stricly voip |
| 04:21.58 | _Vile | unless they don't charge you |
| 04:22.04 | Mavvie | [~] root@tim>pkg_version -vs radiu |
| 04:22.04 | Mavvie | freeradius-1.0.1 = up-to-date with port |
| 04:22.09 | _Vile | I have 4 PRI' |
| 04:22.10 | Mavvie | looks like I do.... |
| 04:22.17 | _Vile | s from local and LD providers |
| 04:22.20 | _Vile | plugged into one box |
| 04:22.30 | _Vile | and it works like a charm for VoIP to PSTN termination |
| 04:22.38 | FuriousGeorge | im new here |
| 04:22.39 | nix000 | Mavvie, have you ever done accnting using msql ? |
| 04:22.40 | FuriousGeorge | whats a pri |
| 04:22.43 | _Vile | that's just in one box |
| 04:22.45 | Mavvie | _Vile: no problems with clock selection? |
| 04:22.49 | _Vile | PRI is 23 channels w/ D channel signalling |
| 04:23.05 | _Vile | DSS is 24 channels straight, 24 phone lines, both are supported from * |
| 04:23.11 | FuriousGeorge | what i d/ channel signaling |
| 04:23.21 | _Vile | Mavvie, you know, I experienced a small problem with clocking |
| 04:23.26 | Mavvie | nix000: no, we have it just to a plain file because the earlier versions didn't reconnect to the database. |
| 04:23.35 | _Vile | and ended up selecting one as a primary |
| 04:23.42 | *** join/#asterisk newsham ({d64KtK7VP@malasada.lava.net) |
| 04:23.44 | _Vile | and not selecting a secondary |
| 04:23.45 | Mavvie | _Vile: we had that this week. Two providers, one box. In the end, splitted it into two boxes. |
| 04:23.46 | _Vile | and it fixed it |
| 04:23.54 | newsham | is it possible to use a linksys voip box with asterisk? |
| 04:24.32 | Mavvie | _Vile: aha.I'll remember that for next time. |
| 04:24.38 | FuriousGeorge | vile_: sorry i got confused. did you just tell me asterisk does work with regular ma bell phone lines? |
| 04:24.59 | Mavvie | FuriousGeorge: asterisk is software, you are talking hardware. |
| 04:25.01 | Mavvie | ~hardware |
| 04:25.03 | jbot | hardware is probably http://www.digium.com/index.php?menu=hardware_products. If you don't know what you need, start with an TDM400P and an FXS module. |
| 04:25.08 | _Vile | Furious, on ISDN PRI's, you'll lose one channel for signalling, so you'll have 23 voice channels available... using the 24th for signalling.. all 64k.. on DSS you'll have all 24 channels at 56k, with inband signalling.. doesn't make much difference unless you want to be able to set caller id |
| 04:25.09 | FuriousGeorge | i know |
| 04:25.11 | nix000 | Mavvie, i have the darn thing able to talk to the db. it inserts post_auth if i uncomment it but it just does not log the accounting to the db even tho i have uncommented it ! |
| 04:25.23 | `Sauron | Hum di dum. |
| 04:25.41 | _Vile | that's on T1, DS-3 PRI is different, you can lose fewer channels per DS-3 than you do w/ DS-1 |
| 04:25.56 | Mavvie | hi `Sauron. |
| 04:26.00 | _Vile | Furious, it does |
| 04:26.03 | `Sauron | what up |
| 04:26.14 | _Vile | Furious, you can go the TDM approach or the T-1 Voice Approach |
| 04:26.22 | `Sauron | heh |
| 04:26.24 | newsham | vil: what about if you want less than a pri.. like 2 lines or 1? |
| 04:26.36 | `Sauron | mav, it's funny.. everytime I see you, I have to think for a few seconds to remember who you are :) |
| 04:26.55 | FuriousGeorge | vile: i was saying in know to mavvie. im not good with the telephony abbreviations yet |
| 04:26.56 | FuriousGeorge | what is dss |
| 04:27.01 | _Vile | newsham, then you'll be looking at a BRI |
| 04:27.01 | `Sauron | someone want to stay up another 3 hours for me to bid on this ebay auction? |
| 04:27.03 | _Vile | and not a PRI |
| 04:27.09 | nix000 | anyone know a gateway that supports ss7,C5 interface that works with asterisk ? |
| 04:27.10 | FuriousGeorge | i can wire a modular phone jack and install linux. astersk is gonna be a jump |
| 04:27.20 | *** join/#asterisk phantam (~phantam@72.252.15.235) |
| 04:27.20 | FuriousGeorge | i got lost |
| 04:27.23 | FuriousGeorge | just now |
| 04:27.24 | newsham | so you typically get an isdn type link (2B+D or something) and hook it up? |
| 04:27.28 | phantam | uggg |
| 04:27.30 | phantam | what the hell |
| 04:27.34 | phantam | i cant get it to work |
| 04:27.44 | _Vile | newsham, yep |
| 04:27.54 | Mavvie | FuriousGeorge: read the documentation about asterisk first, then the documentation about the phone system and then ask again :-) |
| 04:28.00 | newsham | are there services that give you multiple ptsn lines over the net (like vonage but multiple lines)? |
| 04:28.05 | _Vile | Furious, i actually don't know the definition of DSS, I just know the abbreviation and know what it means |
| 04:28.20 | `Sauron | dss is fun |
| 04:28.22 | _Vile | Digital Signalling Services? dunno |
| 04:28.37 | newsham | direct spread spectrum? :) |
| 04:28.39 | `Sauron | digital spread spectrum |
| 04:28.48 | _Vile | there's your answer :) |
| 04:28.56 | phantam | newsham: get a few vonage accounts... and put them in a trunk |
| 04:28.57 | `Sauron | does frequency hopping so you can use lower power and still get the same bandwidth |
| 04:29.01 | FuriousGeorge | lol, i read about asterisk, but i dont remember reading if the special hw for it only worked with ethernet or if it could route regular analog calls |
| 04:29.02 | `Sauron | makes it less intrusive |
| 04:29.03 | _Vile | I just know that dss is inband signalling |
| 04:29.29 | _Vile | taking space per channel for signalling |
| 04:29.42 | newsham | i was looking at the linksys boxes they're selling cheap now and how they're locked to vonage and I see some hints online that it might be possible to use them with asterisk, but i havent seen any definitive info |
| 04:29.44 | _Vile | PRI is cleaner, to me anyway |
| 04:29.52 | newsham | (just on resetting the boxes to factory state, not to setting them up afterwards) |
| 04:29.58 | phantam | huh |
| 04:29.59 | phantam | dude |
| 04:30.02 | phantam | u dont need the boxes |
| 04:30.10 | iceyp | Heres a question... I have 2 servers running asterisk in 2 locations in New Zealand, I'll be using 1 mysql database with the users located within the db. When a user connects with their extension number (they can use either server) how does the interconnect work? Like dialing 2202 (my next door neighbour) I wouldnt know which PBX he's connected to |
| 04:30.11 | newsham | does anyone know if its possible and if so how? |
| 04:30.14 | phantam | u can connect to vonage via asterisk directly |
| 04:30.20 | newsham | I dont want vonage. |
| 04:30.34 | phantam | oh u mean |
| 04:30.36 | _Vile | newsham, you need PAP2-NA's |
| 04:30.41 | _Vile | non locked |
| 04:30.45 | phantam | linksys -> asterisk -> linksys |
| 04:30.48 | newsham | I heard that they dont sell pap2-na's anymore |
| 04:30.54 | _Vile | they do |
| 04:30.56 | newsham | phantam: yessir. |
| 04:31.11 | newsham | vile: any idea where? |
| 04:31.16 | phantam | why not just bet cheap sip phones |
| 04:31.18 | _Vile | www.froogle.com |
| 04:31.32 | _Vile | you might have to buy some minutes from someone |
| 04:31.36 | _Vile | but that's the deal |
| 04:31.42 | _Vile | unless you're a vendor |
| 04:31.45 | _Vile | then talk to linksys |
| 04:31.48 | _Vile | and they'll hook you up |
| 04:31.52 | newsham | nah, i'm not going to buy minutes from someone to get hardware |
| 04:32.02 | phantam | just get sipphones |
| 04:32.04 | phantam | there cheap enough |
| 04:32.08 | iceyp | [asterisk - 202.7.6.33 - Wellington ] ---peer--- [asterisk - 202.7.6.33 - Auckland] (Users exist on both machines (anycast ip range) and connect to the closest server. How do the PABX's know which server to deliver the call to? |
| 04:32.12 | _Vile | or become a vendor |
| 04:32.59 | jbAU | iceyp: you have to do the work in your dial plan |
| 04:33.25 | iceyp | jbau so if the user is connected, deliver the call locally, if not, deliver it to the other server; if they not there, leave voice mail? |
| 04:33.31 | *** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com) |
| 04:33.33 | phantam | why |
| 04:33.35 | phantam | does iax hate me |
| 04:33.40 | newsham | buying bundled service with hardware or buying locked hardware is against my religeon |
| 04:33.41 | phantam | i dial 8,411 |
| 04:33.43 | jbAU | iceyp: are the servers in different states? |
| 04:33.43 | phantam | and i get |
| 04:33.44 | phantam | Mar 1 23:33:37 WARNING[13824]: chan_iax2.c:5546 socket_read: Call rejected by 65.39.205.121: No such context/extension |
| 04:33.58 | iceyp | jbau you could say that |
| 04:34.05 | JamesDotCom | phantam: and what does that error suggest? |
| 04:34.07 | _Vile | newsham, then buy ATA's from ebay, and pray to your religion that they're unlocked. |
| 04:34.09 | phantam | well |
| 04:34.10 | iceyp | 2 diff citys, on major peering exchanges |
| 04:34.11 | _Vile | Have a nice day. |
| 04:34.13 | phantam | i checked the contexts already |
| 04:34.23 | phantam | there all set |
| 04:35.22 | par | hi newsh |
| 04:35.26 | newsham | hey par |
| 04:35.29 | newsham | ltns |
| 04:35.36 | jbAU | iceyp: ok - well i meant different call ratings, then you'd say on PBX1 have a dial rule so that all numbers with say 02XXXXXXXX get routed to PBX2 for cheaper calls |
| 04:35.39 | par | yeah, in tx now |
| 04:35.48 | newsham | i'm in HI now |
| 04:35.50 | jbAU | iceyp: if 02XXXXXXX is an STD call for PBX1 |
| 04:35.58 | par | well aloooha! |
| 04:36.02 | jbAU | you are so new you even spelt new wrong |
| 04:36.09 | FuriousGeorge | vile_: i got my wires crossed and thought you were answering me when you were answering others. allow me to reask. asterisk can route calls from regular analog phone lines right? iow, does it work strictly via ehernet as a voip router, or is it an all purpose telephony pbx |
| 04:36.12 | iceyp | jbAU any idea how the config would look for that dialplan? |
| 04:36.39 | par | newsh: you have a PRI there? |
| 04:36.44 | newsham | no |
| 04:36.50 | newsham | i have pots and dsl |
| 04:36.57 | jbAU | iceyp simple, you just say dial 02XXXXXXXX,1,dial(IAX2/pbx2) |
| 04:37.17 | jbAU | and then 02XXXXXXXX,2,dial(zapta/pri) |
| 04:37.18 | _Vile | Furious, think of Asterisk as a Switch... you can have 2 4 port cards routing 4 ports of T1 from one card to the other 4 ports of another card, or routing 8 ports of T1 to SIP or IAX or H232, etc. |
| 04:37.31 | par | newsh: switching over to voip? |
| 04:37.32 | _Vile | it'll do anything that you want it to do. |
| 04:37.35 | jbAU | so that if for whatever reason the link to pbx2 is down you can still get out on POTs |
| 04:37.50 | ariel_ | phantam, back. the context is no place to too. But the error from fwd suggest posible incorrect password or account. |
| 04:37.51 | iceyp | jbAU nah its all pc to pc, so something like this ..... 02XXXXXXXX,1,dial(IAX2/pbx1) , 02XXXXXXXX,2,dial(dial(IAX2/pbx2) |
| 04:37.51 | newsham | par: nah, just goofing around with it.. maybe in the future.. playing with iax2, asaterisk and fwdnet |
| 04:37.53 | _Vile | if you're looking at a SIP system, look into SER as well |
| 04:37.54 | iceyp | would that work? |
| 04:38.04 | *** join/#asterisk miguellinux (~miguellin@200.47.223.190) |
| 04:38.08 | jbAU | sure it would, KISS |
| 04:38.22 | iceyp | KISS - keep it simple stupid? |
| 04:38.43 | jbAU | yepo |
| 04:38.44 | _Vile | kiss means that, yes. |
| 04:38.47 | iceyp | :) |
| 04:38.53 | phantam | huh |
| 04:38.56 | newsham | gotta goo, too much scroll in this window |
| 04:38.58 | phantam | how could that be if it registers |
| 04:39.00 | *** part/#asterisk newsham ({d64KtK7VP@malasada.lava.net) |
| 04:39.23 | jbAU | basically there isn't any magic to it, have aread up on least cost diailing/routing on voip-info.org |
| 04:39.30 | ariel_ | phantam, it's telling you it's rejected. |
| 04:39.44 | phantam | no |
| 04:39.47 | phantam | cause u can call me |
| 04:40.47 | ariel_ | calls inbound is different then when your calling them. |
| 04:40.56 | iceyp | jbAU how would i deliver all voice mail to one specific host? i.e. dial(IAX2/pbx2/voicemail) |
| 04:40.57 | ariel_ | register is only telling them where your at. |
| 04:41.01 | iceyp | so all mail is stored on one pbx |
| 04:41.26 | _Vile | Furious, please review voip-info.org extensively and return with questions. |
| 04:41.36 | jbAU | iceyp: well yes you could do that |
| 04:41.50 | _Vile | it'll answer most of your questions. |
| 04:41.52 | phantam | hmm |
| 04:41.53 | jbAU | iceyp: if it were me i would treat both sites seperately tho |
| 04:42.11 | nix000 | anyone know a gateway that supports ss7,C5 interface that works with asterisk ? |
| 04:42.16 | jbAU | iceyp: cause internet links go down |
| 04:42.17 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:42.26 | _Vile | it's where I learned from........ |
| 04:42.33 | iceyp | jbAU problem is if a user moves from wellington to auckland and they check their voice mail, they wont get it because its in wellington |
| 04:42.42 | ariel_ | nix000, Lucent TNT |
| 04:42.57 | _Vile | I don't think a TNT can do ss7 |
| 04:43.02 | _Vile | can it? |
| 04:43.16 | ariel_ | _Vile, there is an addon for it. But it's expensive. |
| 04:43.33 | _Vile | ariel, I'd be looking at excel solutions at that point |
| 04:43.47 | ariel_ | _Vile, yes your correct |
| 04:43.51 | _Vile | EXS-2000 w/ SS7 cards, software etc |
| 04:43.52 | ariel_ | but they asked. |
| 04:44.18 | jbAU | iceyp ok - there's a few ways to get around that one, the simpliest would be to of course store voicemail on the one system. |
| 04:44.28 | _Vile | TNT SS7 doesn't sound like a good idea, feels better as a PSTN <-> SIP exchanger |
| 04:44.42 | ariel_ | nix000, I think there is an t400p card that does ss7 from openss7.org |
| 04:45.02 | iceyp | jbAU thats what im wanting to do i guess |
| 04:45.17 | nix000 | _Vile, i have to interface to legacy hw |
| 04:45.20 | iceyp | let me finnish what i have and get a test system running before i do anything else |
| 04:45.21 | phantam | hehe |
| 04:45.27 | iceyp | i'll come back to voice mail later |
| 04:45.27 | phantam | ariel_: it was a flaw in the dialing on my client |
| 04:45.27 | phantam | lol |
| 04:45.38 | _Vile | nix, excel should be OK |
| 04:45.53 | nix000 | ariel_, i am googling for these cards now |
| 04:46.04 | _Vile | for setup and tear down messages, sigtran and all of that, I know that it handles it... but |
| 04:46.17 | _Vile | I also know that openss7 is in early stages of creation |
| 04:46.17 | nix000 | ariel_, if you have dealt with them that will be nice |
| 04:46.44 | _Vile | but they have finished parts that do work w/ the digium cards |
| 04:46.52 | _Vile | but not fully implemented |
| 04:46.56 | _Vile | and definitely not certified |
| 04:47.00 | phantam | whats a 1800 number i can try |
| 04:47.01 | phantam | lol |
| 04:48.04 | _Vile | nix, check ebay for ss7 and excel, dunno if anything is on there at the moment |
| 04:48.21 | _Vile | it's your best best, pctelecom.com can help you with software |
| 04:48.36 | _Vile | there are also a variety of other software vendors out there |
| 04:48.42 | _Vile | but you'll have to search for them |
| 04:49.17 | _Vile | check ebay first for "Excel" "EXS" |
| 04:49.22 | _Vile | and get your feeling about pricing |
| 04:49.27 | _Vile | before you continue. |
| 04:50.09 | phantam | wow thats choppy as hell |
| 04:50.37 | par | i have a cisco ubr924 with voip functionality.. |
| 04:50.48 | par | its my cable modem |
| 04:51.00 | FuriousGeorge | _Vile_: i think i get it now, its just a matter of what port is on the asterisk HW which dictates what type of connection i may use it with |
| 04:51.05 | par | it also has two FXS ports for pluggin in a phone and fax |
| 04:51.19 | par | how can i hook an asterisk box up to it? |
| 04:51.27 | FuriousGeorge | i know there is a whole slew of HW made for asterisk servers but i never really looked into it |
| 04:51.32 | _Vile | furious, good... continue learning. :) im going to smoke |
| 04:51.44 | FuriousGeorge | burn one down brother |
| 04:52.28 | par | i want to route some voip over the cable modem router's built-in and some over the normal internet to other gateways / devices |
| 04:53.01 | Mavvie | FuriousGeorge: try the ~hardware command on this channel for a good first pointer. |
| 04:53.16 | FuriousGeorge | ~hardware |
| 04:53.17 | jbot | hardware is, like, http://www.digium.com/index.php?menu=hardware_products. If you don't know what you need, start with an TDM400P and an FXS module. |
| 04:53.32 | FuriousGeorge | im weeks from buying anything |
| 04:54.18 | Mavvie | FuriousGeorge: so? You still need to know what is there before you can make a descission on what you need or what it is actually what you are going to do. |
| 04:54.43 | FuriousGeorge | im just trying to better my general undetrstanding of the software. i think its a great project, its just that asterisk.org makes my eyes bleed for a variety of reasons so id rather interact here than poke around tgher |
| 04:54.56 | phantam | is it just me or fwd choppy |
| 04:55.05 | FuriousGeorge | maviee: i prmise i will do hours of research before i buy anything |
| 04:55.27 | Mavvie | FuriousGeorge: not necessary if you read what they have for sale, because it explains what is possible and what you need. |
| 04:55.57 | phantam | is there a way to favor quality over compression? |
| 04:56.18 | Mavvie | phantam: disallow=all, allow=ulaw,alaw |
| 04:56.26 | Mavvie | (for example) |
| 04:56.32 | phantam | i thought gsm was supposed to be good |
| 04:56.44 | FuriousGeorge | im looking at it now, very informative |
| 04:57.16 | Mavvie | phantam: I thought we were talking about quality :-) |
| 04:57.21 | phantam | lol |
| 04:57.27 | phantam | gsm's pretty clear when i talk on it lol |
| 04:57.42 | phantam | but when i call over asterisk using gsm to fwd its pretty choppy |
| 04:57.44 | nix000 | ariel_, according to http://www.openss7.org/asterix.html openss7 does not love asterisk yet ... darn that was so close |
| 04:57.57 | Mavvie | for what it is worth, ulaw/alaw is 64Kbps raw uncompressed, the rest is all lesser quality. |
| 04:59.12 | phantam | where do i have to specify that |
| 04:59.17 | phantam | in sip.conf ? |
| 04:59.20 | Mavvie | yes |
| 04:59.23 | Mavvie | or/and iax.conf |
| 05:00.17 | FuriousGeorge | digium the only people who make asterisk hw or are there others? |
| 05:00.40 | Mavvie | not sure |
| 05:02.47 | FuriousGeorge | well this has been informative and motivating, im gonna install gentoo tomarrow, it'll take a week to get that and everything else working right, a month to get the hardware, and then ill be back |
| 05:02.53 | ariel_ | phantam, great to hear you got it working. |
| 05:02.59 | phantam | hehe yep |
| 05:03.07 | phantam | hmm should jitter buffer be enabled |
| 05:03.30 | ariel_ | nix000, look for there product they have a card they sell not free with ss7 |
| 05:03.36 | FuriousGeorge | thanks for all the info everyone, thank vile for me when hes done smokeing. ill see you again soon enough |
| 05:03.38 | *** part/#asterisk FuriousGeorge (~FuriousGe@ool-43516ebb.dyn.optonline.net) |
| 05:04.44 | phantam | ariel_: should i turn on jitterbuffer on in iax.conf |
| 05:04.54 | *** join/#asterisk jskcr (~jskcr_@jskcr.user) |
| 05:06.19 | ariel_ | phantam, jitterbuffer=yes |
| 05:06.20 | ariel_ | dropcount=1 |
| 05:06.20 | ariel_ | <PROTECTED> |
| 05:07.05 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 05:07.05 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:09.28 | phantam | what about tos |
| 05:09.29 | phantam | ? |
| 05:09.34 | phantam | for least skips |
| 05:09.52 | Juggie | bah, i know so little about unix c developement |
| 05:10.31 | *** join/#asterisk sbarrius (~sbarrius@c-24-15-201-23.client.comcast.net) |
| 05:11.25 | nix000 | ariel_, price is not an issue (for now) but it has to integrate with asterisk somehow and i dont see it. |
| 05:11.44 | sbarrius | Does anyone know if you can 'Request confirmation of answering by waiting for a #' with non Zap Channels? |
| 05:12.57 | PyroSteve | does ChanSpy work ? |
| 05:13.30 | phantam | ariel_: can u try to call me |
| 05:13.32 | BoRiS | everyone wants that :-p |
| 05:13.50 | phantam | tell me what u hear |
| 05:13.56 | sbarrius | wants # confirmation? |
| 05:14.23 | *** join/#asterisk JerJer (~jj@65.173.197.109) |
| 05:15.28 | mishehu | moible? |
| 05:15.29 | mishehu | heh |
| 05:15.53 | sbarrius | Is anyone doing follow-me scripts with asterisks? |
| 05:16.54 | shido6 | yep |
| 05:16.58 | shido6 | it works, ok |
| 05:17.03 | JerJer[moible] | mobile |
| 05:17.11 | JerJer[moible] | gprs baby |
| 05:17.27 | sbarrius | are you zap for follow me? |
| 05:17.29 | Mavvie | sbarrius: I use enum for that. |
| 05:17.57 | Mavvie | (mostly because I can, not because it's easier or more difficult than doing it in extensions.conf :-) |
| 05:18.52 | sbarrius | can you use enum for 'Request confirmation' |
| 05:19.15 | Mavvie | no. enum is for my follow me implementation |
| 05:19.41 | sbarrius | like you need to hit # to accept the call otherwise it times out into your asterisk voicemail |
| 05:20.29 | Damin | JerJer[moible]: I just got a Samsung i700 via Verizon... |
| 05:20.43 | Damin | JerJer[moible]: It should be here on Thursday.. |
| 05:21.00 | Damin | JerJer[moible]: First thing that is getting installed is Putty. :) |
| 05:21.21 | sbarrius | you can installl putty on it? |
| 05:21.39 | Damin | sbarrius: Yep.. There is a Putty for PocketPC |
| 05:21.51 | sbarrius | nice... |
| 05:22.13 | Damin | http://pocketputty.duxy.net/\ |
| 05:23.14 | Damin | Unless there is a better SSH client for the PocketPC? Even if it is a couple of bucks? |
| 05:23.36 | sbarrius | goto go, have fun with you new samsung! |
| 05:23.41 | sbarrius | quit |
| 05:23.49 | *** part/#asterisk sbarrius (~sbarrius@c-24-15-201-23.client.comcast.net) |
| 05:27.32 | *** join/#asterisk iceyp (~icepick@max.unix.co.nz) |
| 05:27.54 | iceyp | hey guys, How do i take digits a user has inputted and dial that number? i.e. exten => 010,6,Dial(ENTERED) |
| 05:28.22 | iceyp | my current diapl plan looks like this http://www.pastebin.com/248170 |
| 05:28.56 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-86-63.sentechsa.net) |
| 05:29.10 | Mavvie | ${EXTEN} |
| 05:29.25 | Mavvie | oh wait. |
| 05:30.23 | iceyp | ? |
| 05:30.32 | Mavvie | what does responsetimeout do according to you? |
| 05:30.35 | JerJer[moible] | mooooo |
| 05:30.56 | iceyp | it disconnects the call if there is no input from the user |
| 05:31.30 | JerJer[moible] | anyone ever figure out how to get the info digits out of libpri/zap ? |
| 05:31.34 | Mavvie | no. |
| 05:31.37 | Mavvie | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ResponseTimeout |
| 05:31.43 | Mavvie | Set maximum timeout awaiting response |
| 05:31.47 | *** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk) |
| 05:31.47 | iceyp | ok i'll remove it then |
| 05:31.56 | iceyp | thats what i said? |
| 05:32.07 | iceyp | if there is no response then disconnect? |
| 05:32.14 | Mavvie | no, it sets the timeout, it doesn't do anything with the user. |
| 05:32.15 | iceyp | which it does do |
| 05:32.38 | iceyp | well it currently disconnects them i think, even if i removed that how do i get asterisk to accept entered numbers? |
| 05:32.42 | iceyp | and dial an extension |
| 05:32.55 | Mavvie | go to the URL I pasted, and read at the See Also |
| 05:33.05 | iceyp | try it... iax2/010@voip.fast.co.nz |
| 05:33.46 | iceyp | it's getting to dial and doing this now : Mar 2 18:30:40 WARNING[69726]: app_dial.c:698 dial_exec: Dial argument takes format (technology1/[device:]number1&technology2/[device:]number2...|optional timeout) |
| 05:34.04 | JerJer[moible] | IAX2 |
| 05:34.07 | JerJer[moible] | caps |
| 05:34.31 | JerJer[moible] | and use a type=peer |
| 05:34.49 | iceyp | jerjer? who you talking to? |
| 05:34.57 | iceyp | is it not accepting my call? |
| 05:35.02 | fearnor | vile - are you using ESXs? |
| 05:35.13 | Mavvie | iceyp: did you read e See Also there? |
| 05:35.49 | iceyp | yeah |
| 05:35.51 | iceyp | http://www.voip-info.org/wiki-Asterisk+cmd+WaitExten |
| 05:36.51 | Mavvie | so, where do you read the number? |
| 05:37.06 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 05:37.52 | iceyp | http://www.pastebin.com/248177 that look right? |
| 05:38.20 | Mavvie | same question, where do you read the users input? |
| 05:39.14 | iceyp | Predigits=${EXTEN} or that would apear pre |
| 05:39.22 | Mavvie | that is a SetVar command. |
| 05:39.27 | Mavvie | SetVar doesn't read users input. |
| 05:39.46 | iceyp | exten => s-gathermoredigits,3,WaitExten(8) ; and give the caller 8 seconds overall to do their thing |
| 05:39.47 | iceyp | <PROTECTED> |
| 05:39.58 | *** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net) |
| 05:41.01 | *** join/#asterisk clive-- (~pirch@myw-stp-66-18-85-251.sentechsa.net) |
| 05:43.13 | *** join/#asterisk matjing (~Miranda@62.8.64.33) |
| 05:43.21 | iceyp | argh |
| 05:43.45 | mikeirc | Ya know something... I've been having a hell of a time getting * working (first time installer here).. and I just realized...if I hadn't gone the lazy route of installing AMP (which of course I did)... I would have gotten this thing up and going a long time ago... the documentation explains everything in plain english... Trying to be lazy and just using AMP actually wound up being more time consuming because I had no idea what was goi |
| 05:43.45 | mikeirc | ng on the back end... I'm half tempted to do a complete reinstall and just go plain jane... Just thought I'd share my new user experiences... |
| 05:44.27 | iceyp | "2206","","010","home","""2206""","SIP/2206-851a","","WaitExten","8","2005-03-02 18:43:58","2005-03-02 18:43:58","2005-03-02 18:44:18",20,20,"ANSWERED","BILLING" |
| 05:44.32 | mikeirc | ...had no idea what was going on on the back end... |
| 05:45.05 | mikeirc | oh...nevermind that last statement..it wsa already put in I see.. ;) |
| 05:45.32 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:51.22 | harryvv | yea i saved my cdr and read it in excel for some reason it did not pass the dates and times other then minitues used to excel |
| 05:51.39 | BoRiS | How did you put it into excel? |
| 05:52.32 | harryvv | well I didnt. As soon as i did a a: on the run bar it was listed as readable Microsoft Excel Comma Separated Values File |
| 05:52.55 | Juggie | mikegrb, amp sucks. |
| 05:53.12 | harryvv | Everything looks okay just the dates do not show I do see alot of # signs. perhaps those are in its place in the colums. |
| 05:53.27 | Juggie | harryvv, sigh unexpirenced user i see :) |
| 05:53.36 | Juggie | #### means your column isnt wide enough |
| 05:54.02 | Juggie | but you should be putting your csv into a database |
| 05:54.27 | harryvv | i se |
| 05:54.29 | harryvv | see |
| 05:54.43 | Nugget | oh god. here comes the misguided soul advocating mysql. |
| 05:55.04 | Ash | MYSQL IS AWESAM |
| 05:55.12 | Ash | WHY DO U HAET PERFROMANCE!@#%!@1121? |
| 05:55.18 | Nugget | heh |
| 05:55.25 | implicit | Nugget: seriously man |
| 05:55.34 | Nugget | +2, Funny, but -1 for missing "h8" instead of "haet" |
| 05:55.35 | implicit | mysql 5 is pretty damn good |
| 05:55.54 | Ash | no way, mysql 6 is where it's at |
| 05:55.56 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 05:55.59 | implicit | Ash: mysql 7 |
| 05:56.05 | Juggie | i didnt say mysql |
| 05:56.05 | Koshatul | mysql 1 billion is teh bomb |
| 05:56.08 | Ash | implicit is right |
| 05:56.09 | Juggie | i just said a database |
| 05:56.11 | harryvv | implicit somone told me to avoid it and go postgresql |
| 05:56.19 | implicit | harryvv: did you ever try either of them? |
| 05:56.22 | implicit | harryvv: :-P |
| 05:56.31 | harryvv | I fidiled with mysql once |
| 05:56.33 | implicit | harryvv: or just use someone elses programs that interfaced with them |
| 05:56.40 | Koshatul | go ms sql, it is the absolute best sql server on the market .... |
| 05:56.50 | Juggie | hahaha |
| 05:56.51 | implicit | if you rae just doing select * from table; they are pretty much the same, lol |
| 05:56.52 | Koshatul | straight face the whole time, i swear |
| 05:56.54 | Juggie | yah :P |
| 05:56.59 | implicit | hehehehe |
| 05:57.05 | Ash | if you want mssql on unix, just buy sybase |
| 05:57.07 | Ash | duh |
| 05:57.08 | Koshatul | ok i lied, i was giggling like a school girl |
| 05:57.12 | Juggie | freetds exists if you must use mssql :) |
| 05:57.21 | Juggie | since they just stole the protocol from someone else |
| 05:57.23 | Nugget | don't even have to buy it. older sybase is free on linux or freebsd. |
| 05:57.23 | harryvv | is mysql used for billing of long distance toll? |
| 05:57.24 | iceyp | argh, how do i take input data and insert it into dial()? http://www.pastebin.com/248183 >>>??? |
| 05:57.26 | implicit | or if you use sybshit |
| 05:57.35 | implicit | harryvv: are you against radius? |
| 05:57.40 | Koshatul | postgres |
| 05:57.43 | harryvv | no why |
| 05:57.50 | implicit | harryvv: then use it for AAA |
| 05:57.56 | harryvv | ? |
| 05:58.10 | *** join/#asterisk jerryh ([U2FsdGVkX@67.141.135.121) |
| 05:58.20 | harryvv | brb what is aaa |
| 05:58.21 | Juggie | implicit, remember some of us were talking ldap last night |
| 05:58.41 | Juggie | curious, i installed a ldap browser on my pc at work and hit up our active directory server |
| 05:58.43 | implicit | Juggie: i was pretty tired |
| 05:58.44 | implicit | i dont remember |
| 05:58.46 | Juggie | ahh |
| 05:58.49 | implicit | you don't like LDAP? |
| 05:58.51 | Juggie | well this will be intreasting none the less |
| 05:58.54 | implicit | ldap is good |
| 05:59.00 | Juggie | so i install a ldap browser on my pc at the office |
| 05:59.06 | implicit | thats the problem |
| 05:59.10 | implicit | never install an ldap browser |
| 05:59.15 | Juggie | and login to our active directory server |
| 05:59.21 | Juggie | (which also runs ldap) |
| 05:59.22 | Koshatul | you must learn to write your own queries in binary |
| 05:59.34 | Koshatul | except you will wear out 0 and 1 on your keyboard |
| 05:59.37 | Juggie | yah i just wanted to see what kinda information was available :) |
| 05:59.39 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 05:59.53 | Juggie | anywho, the kicker is we run mitel's new unified messaging platform |
| 06:00.00 | Juggie | which is tied directally into activedirectory |
| 06:00.05 | Nugget | keyboard?!?! when I was growing up we had to bang rocks together to make 1s. |
| 06:00.11 | implicit | Koshatul: forget keyboards |
| 06:00.17 | implicit | harryvv: radius is good |
| 06:00.18 | jerryh | bah, when i was a kid we had to carve our 1's and 0's out of rock |
| 06:00.20 | Koshatul | implicit: neural headgear ? |
| 06:00.21 | Juggie | which inherentally is fine however, mitel has chosen to store your voice mail pin in active directory! |
| 06:00.32 | Juggie | thus, browsing everyons profile with ldap, you could see everyones voicemail pin |
| 06:01.16 | Juggie | seems to me, active directory isnt the place for this kind of information |
| 06:01.19 | Koshatul | Nugget: *clack* *clack* 1, *clack* 0, *silence* call 000 they've died from repetitve use of input device to the cranium |
| 06:01.40 | Koshatul | Juggie: not even some kind of hashed pin ? |
| 06:01.44 | Juggie | nope |
| 06:01.46 | Juggie | plain text pin |
| 06:01.55 | Juggie | i could have logged into anyones voicemail |
| 06:01.58 | jerryh | welcome to software development, corporate style |
| 06:02.03 | Koshatul | can anyone without admin rights see it ? |
| 06:02.07 | Juggie | yes |
| 06:02.12 | Juggie | i am just a network user |
| 06:02.32 | Juggie | you should never store that kind of information in there. |
| 06:02.33 | Koshatul | Welcome to Software Development .. VB.Net Style :) |
| 06:02.46 | Koshatul | unless hashed, similar to a ad password |
| 06:02.48 | jerryh | Koshatul: same difference :D |
| 06:02.54 | Juggie | the rest of the ldap was just like, last login, number of logins, email, username, etc... random windows information |
| 06:02.58 | Koshatul | jerryh: :) |
| 06:02.58 | Juggie | nothing important |
| 06:03.16 | Juggie | and mitel decides to stick all my voicemail settings in there, and cap it off with a clear text pin. |
| 06:03.40 | Juggie | i work/live like 20 mins from their offices |
| 06:03.44 | Juggie | i should drive down for a visit |
| 06:03.48 | Juggie | and ask them wtf they are thinking |
| 06:04.04 | Koshatul | in their defense ... which i hate to sit on, so i won't stay long ... the only users that should be able to access that are "trusted" emplyoees, and to use a ldap browser, they |
| 06:04.04 | jerryh | Are you folks #asterisk regulars? |
| 06:04.11 | Koshatul | re misusing company time ... |
| 06:04.20 | implicit | jerryh: yeh |
| 06:04.24 | Juggie | Koshatul, ldap is supposed to be an open directroy |
| 06:04.26 | Koshatul | so you should get back to work slacker :) |
| 06:04.27 | Juggie | *directory |
| 06:04.36 | shido6 | to some of us , this is work, KoolHercz |
| 06:04.38 | shido6 | errr |
| 06:04.40 | shido6 | Koshatul |
| 06:04.40 | Juggie | its not intended to store secure information. |
| 06:04.42 | Koshatul | keyword: supposed :P |
| 06:04.43 | Juggie | yes, this is work |
| 06:04.44 | shido6 | tab completion :) |
| 06:04.52 | *** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 06:04.53 | Juggie | i was researching the possiblity of tieing asterisk/php to ldap |
| 06:05.02 | Koshatul | shido6: it was one line, i just hit enter while typing |
| 06:05.13 | Inv_arp | Juggie: not hard |
| 06:05.28 | Juggie | Inv_arp, i know |
| 06:05.33 | harryvv | juggie what do you use php for |
| 06:05.38 | Himeko | some places would fire you for that if you told them you installed unauth software on your computer |
| 06:05.44 | Koshatul | i so want your work, and i agree, storing passwords anywhere without a good reason in plain text is just ... plain stupid :) |
| 06:05.54 | implicit | ~seen cmc |
| 06:05.55 | jbot | i haven't seen 'cmc', implicit |
| 06:06.04 | Juggie | harryvv, alot of things, interfacing with the asterisk manager api, and other stuff non related to asterisk |
| 06:06.05 | Koshatul | Himeko: some places would fire you if you pointed out a problem in their software :\ |
| 06:06.23 | Inv_arp | sh*t i use sqlite/php/asterisk |
| 06:06.27 | harryvv | juggie is that premade or something you made. |
| 06:06.41 | Juggie | harryvv, all written inhouse. |
| 06:06.47 | harryvv | cool |
| 06:07.23 | Koshatul | Juggie: nice |
| 06:08.07 | Juggie | right now, we have click to talk, web enabled conferencing (to a point not finished) and an asterisk control pannel which can view all the information, and send commands you could send via the cli... but no editing sip/iax/extensions/voicemail via the web... (yet) |
| 06:08.35 | Juggie | i am trying to get permission to GPL all our work, but dont hold your breath. |
| 06:08.37 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 06:08.57 | pimpwell | was wondering if there was a tutorial geared in depth at the .call file |
| 06:09.28 | Juggie | pimpwell, sample.call exists in the asterisk source dir, i think in contrib? and there are docs on the wiki. |
| 06:09.53 | Juggie | also pimpwell, unless you need to schedule calls, i reccomend the manager api. |
| 06:10.01 | harryvv | seen ~slepp |
| 06:10.11 | Juggie | you dont have to worry about file permissions then for a web server writing into that dir or whatever. |
| 06:10.51 | pimpwell | let's say all I need to worry about is the construction of the .call file. Everything else is done elsehwere by someone else. |
| 06:10.52 | Himeko | i've seen him |
| 06:10.56 | Himeko | skinny guy |
| 06:11.01 | harryvv | who is that |
| 06:11.03 | pimpwell | I create the .call file and drop it in someone's outbound. |
| 06:11.08 | Himeko | slepp |
| 06:11.16 | Juggie | pimpwell, then look at the wiki |
| 06:11.24 | Juggie | however, you need to create the file in another directory |
| 06:11.31 | Juggie | and when you finish writing it, move it to outbound |
| 06:11.35 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 06:11.43 | Juggie | (move, not copy) |
| 06:11.44 | pimpwell | k |
| 06:11.45 | pimpwell | ty |
| 06:11.58 | pimpwell | www.voip-info.com |
| 06:12.08 | harryvv | Himeko is here local to vancouver? |
| 06:12.20 | pimpwell | www.voip-info.org |
| 06:13.35 | harryvv | thanks |
| 06:13.38 | harryvv | see some of his post |
| 06:14.36 | Juggie | pimpwell, theres a sample.call in the asterisk source root dir. |
| 06:16.47 | pimpwell | mine gets real tricky |
| 06:17.02 | pimpwell | need some sort of reference anyway |
| 06:18.42 | Himeko | here? |
| 06:18.54 | Himeko | he? |
| 06:18.58 | Himeko | edmonton |
| 06:19.57 | harryvv | Other then call waiting is there a way to break into a call in progress for emergency reasons? |
| 06:20.45 | slePP | harryvv: ? |
| 06:21.14 | harryvv | ohh slepp, somone said you provided vancouver did's have a web page of this info? |
| 06:21.29 | slePP | uuuuuhm |
| 06:21.35 | slePP | http://www.thinktel.ca/ |
| 06:21.37 | slePP | maybe :> |
| 06:21.43 | slePP | the web guy has been working on it for a while |
| 06:21.53 | harryvv | where are your termination points located at |
| 06:21.55 | harryvv | okay |
| 06:22.22 | Himeko | hey slePP |
| 06:22.27 | slePP | 'lo himeko |
| 06:22.42 | slePP | harryvv: off the top of my head, we have 604-678-xxxx |
| 06:22.53 | harryvv | good deal |
| 06:23.19 | harryvv | is there a issue transfering our telus number to that did? I know most dont. |
| 06:23.29 | slePP | should be workable, yeh. |
| 06:23.36 | slePP | we haven't yet ported a local vancouver, but we don't see a problem with it |
| 06:24.05 | harryvv | i see |
| 06:25.19 | slePP | when did you want service by? |
| 06:25.56 | *** join/#asterisk riksta (~rick@host217-42-22-145.range217-42.btcentralplus.com) |
| 06:31.48 | *** join/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 06:32.17 | par | so, all you really need is an asterisk server and a digium tdm400p to do voip.. no need for any fxs or fxo card |
| 06:32.19 | par | ? |
| 06:33.57 | Juggie | not if u have a t1/e1 no |
| 06:34.10 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 06:34.22 | SexyKen | Hey guys -- anyone know of some major bugs with Asterisk? |
| 06:34.43 | SexyKen | Bugs I should say..with the latest release...that weren't previously there. |
| 06:34.52 | Juggie | SexyKen. check the bug list |
| 06:35.01 | Juggie | http://bugs.digium.com |
| 06:35.04 | SexyKen | Okay. |
| 06:35.21 | Juggie | par, i'll answer again |
| 06:35.24 | Juggie | i misread your question |
| 06:35.32 | Juggie | you dont need any board to do voip. |
| 06:35.38 | par | juggie: haha i mean for people who don't have an integrated T1 |
| 06:35.50 | Juggie | its only when u want to access the pots network that u need a digium board |
| 06:36.20 | SexyKen | Anyone here use a Polycom Soundpoint IP 600 |
| 06:36.34 | Juggie | nope, whats your problem with it |
| 06:36.51 | par | ok so you just connect an ip phone to a switch |
| 06:37.01 | SexyKen | Not really a problem -- just curious how to make proper use of it. I want to be able to do 3 digit dialing with it...which doesnt currently work. |
| 06:37.51 | Juggie | par, yes ip phone to the same network your asterisk box is on |
| 06:38.14 | Juggie | SexyKen, you have to write 3digit dialing into asterisk and set it up on the polycom |
| 06:41.48 | *** join/#asterisk Essobi (kstone@75.137.26.216.host.teledvance.com) |
| 06:41.57 | Essobi | MMM. |
| 06:43.20 | iceyp | argh, how do i take input data and insert it into dial()? http://www.pastebin.com/248183 >>>??? |
| 06:46.40 | SexyKen | Juggie - Yea I dont know how to set it up within the Polycom. |
| 06:48.43 | Koshatul | anyone know where to get a cheap analog phone -> sip unit in australia ? |
| 06:50.07 | Juggie | SexyKen, wiki likely. |
| 06:50.23 | Juggie | does the poly com run a web server? check there... |
| 06:50.50 | SexyKen | Yea it does but nothing in the web iface about it...I'll keep lookin. |
| 06:51.56 | Juggie | SexyKen, anything in the phone gui about it |
| 06:53.31 | Juggie | also, the way i know how to do it would be to point the phone @ a ftp server. |
| 06:53.55 | Juggie | and have it download its config files and in the sip.cfg you have your dialplan |
| 06:58.02 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 06:58.31 | *** join/#asterisk brettnem (~brettnem@user-0ccsr2l.cable.mindspring.com) |
| 06:58.36 | brettnem | good evening all |
| 07:02.14 | *** part/#asterisk marc32344 (~marc32344@69-28-224-214.dsl.teksavvy.com) |
| 07:02.27 | JerJer[moible] | morning |
| 07:03.17 | joaovianna | Hi guys... I'm trying to call out using IAX2 (/var/spool/asterisk/outgoing) but my application start before the callee answer the phone, any clue ? |
| 07:05.53 | joaovianna | Voicepulse ??? |
| 07:06.22 | JerJer[moible] | something is causing the call to go to a connected state then |
| 07:07.38 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 07:08.00 | harryvv | Thats kind of funny that vonage does not have a termination point here in vancouver but are advertising there services here :) |
| 07:08.00 | joaovianna | JerJer, since I'm using a third part IAX2, the problem is in my asterisk or theirs ? |
| 07:08.26 | CoaxD | Okay, i'm convinced |
| 07:08.30 | CoaxD | Coyote Ugly was a DAMN good movie |
| 07:09.45 | h3x0r | harryvv: yeah well they sell vonage retail boxes in all Best Buy stores here in the US, and most of them are in areas they dont have local DIDs here |
| 07:10.21 | harryvv | The problem with that is it creates more risk more points of failure |
| 07:10.51 | h3x0r | not really, its just the underlying carriers they use for DIDs have shitty coverage |
| 07:11.04 | harryvv | okay so thay outsouce to those then |
| 07:11.07 | JerJer[moible] | third part? |
| 07:11.09 | h3x0r | packet8 and voiceglo use level3 and a wireless carrier respectively |
| 07:11.28 | af_ | anyone tried mozphone? (the tel url specially?) |
| 07:11.30 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 07:13.07 | *** join/#asterisk zignig (~simon@203.217.15.10) |
| 07:14.06 | riksta | what's this mozphone? |
| 07:14.37 | joaovianna | JerJer: My IAX2 is from Voicepulse. |
| 07:16.13 | af_ | it's a xpi extension for mozilla/firefix (iax2 client) |
| 07:17.40 | zignig | af_: not a full sip phone then ? |
| 07:17.49 | JerJer[moible] | joaovianna: ok and this is a problem how? |
| 07:18.21 | af_ | iax2 zigman |
| 07:18.25 | af_ | iax2 zignig |
| 07:18.33 | zignig | af_: zigman hehe |
| 07:18.39 | zignig | :) |
| 07:18.40 | af_ | tab stuff |
| 07:18.51 | par | http://www.esato.com/news/article.php/id=98 |
| 07:18.54 | par | seen that phone? |
| 07:18.57 | par | neat |
| 07:19.35 | par | wish fujitsu just made "seamlesslink" open source |
| 07:19.39 | par | or somone rev engs it and puts it in linux themselves |
| 07:19.41 | joaovianna | JerJer: Sorry. |
| 07:20.01 | harryvv | dosvidonia ! |
| 07:20.08 | par | an 802.11b phone that can do cellular public networks as well |
| 07:20.24 | harryvv | par, what brands do you know of? |
| 07:22.13 | par | an 802.11b ip phone that can do cellular public networks as well. |
| 07:22.38 | par | harryvv: only the net2com |
| 07:23.17 | par | does anyone know of an others? |
| 07:23.36 | *** join/#asterisk jtodd (~jtodd@ti.fox-den.com) |
| 07:24.09 | par | prefferrably for gsm (instead of this ghetto foreign dump to a compactflash card method) |
| 07:24.31 | Koshatul | ack, is there a quick and easy way to just busy out a pstn zap interface ? |
| 07:24.48 | Koshatul | (without going allthe way there and plugging in a analog phone ...) |
| 07:25.04 | JerJer[moible] | Koshatul: not really |
| 07:25.16 | JerJer[moible] | or at least i haven't found one |
| 07:25.21 | Koshatul | dang |
| 07:25.28 | Koshatul | actually ... |
| 07:25.44 | Koshatul | i'll create a extention with perpetual hold music, and then originate a call from the zap interface |
| 07:30.05 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 07:32.01 | Koshatul | ok, quick question numero dos: how di i originate a call from an extension, isn't there somewhere you can drop a file and it will follow the commands in that file ? |
| 07:35.01 | par | i wish pulver would come out with a WiSIP mobile ip phone that can do cellular also and automatically switch betweent hem |
| 07:35.13 | Koshatul | got it |
| 07:35.16 | Koshatul | and it worked |
| 07:35.25 | Koshatul | not sure how permanent it is though |
| 07:38.04 | *** join/#asterisk pranav (pranav@202-149-48-205.broadband.isp.exatt.net) |
| 07:38.53 | pranav | hello everyone |
| 07:39.22 | pranav | i want to record calls which i make |
| 07:39.36 | pranav | what is to be done on cdr |
| 07:40.15 | pranav | any other site for detailed application of call monitorng |
| 07:40.25 | *** part/#asterisk pranav (pranav@202-149-48-205.broadband.isp.exatt.net) |
| 07:40.38 | par | NEC and Motorola are teaming up i guess to come out with a roaming WLAN/Cellular |
| 07:41.15 | *** join/#asterisk fitzel (~flint@p3EE39BD3.dip0.t-ipconnect.de) |
| 07:41.25 | fitzel | Good morning |
| 07:41.44 | joaovianna | JerJer: My application still playing prompts before the call is answered. I put wait(2), answer, etc... I checked the all the manuals. It seens like voicepulse is answer my call and then forwarding to my called #... Is it make any sense ? |
| 07:42.52 | *** part/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 07:44.59 | fitzel | Has anybody sucessfully compiled and installed any late CSV together with chan_capi? I only get errors and crashes :| |
| 07:46.15 | Inv_arp | fitzel: what type of errors and crashes? |
| 07:46.23 | Inv_arp | pastebin.ca if needed |
| 07:47.44 | fitzel | During compile of chan_capi I get: chan_capi.c: In function `capi_read': |
| 07:47.44 | fitzel | chan_capi.c:826: structure has no member named `delivery' |
| 07:48.31 | SexyKen | Does anyone else have problems with Asterisk crashing with Extension Routing while using RealTime? |
| 07:48.35 | fitzel | But it compiles and I can install the modules. Asterisk can load them but when I try an to call any msn, there is just "nothing" happening. |
| 07:48.42 | par | oh yay mortorola cn620 |
| 07:49.14 | fitzel | When I do "show channels" after that, I get only some strange output, seems like it does show some random characters. |
| 07:50.11 | Inv_arp | fitzel: u should try mv all the old sources and re cvs/compile fresh |
| 07:50.47 | Inv_arp | SexyKen: extension routing? |
| 07:51.07 | fitzel | Ah, ok. You think there could hang around some old stuff? At least its worth a try. |
| 07:51.22 | SexyKen | Someone who is developing some Asterisk stuff for me told me this: extension routing crashes asterisk with realtime |
| 07:52.04 | fitzel | Here is the pastebin: chan_capi.c: In function `capi_read': |
| 07:52.05 | fitzel | chan_capi.c:826: structure has no member named `delivery' |
| 07:52.05 | fitzel | chan_capi.c:827: structure has no member named `delivery' |
| 07:52.05 | fitzel | chan_capi.c: In function `capi_new': |
| 07:52.05 | fitzel | chan_capi.c:1022: structure has no member named `delivery' |
| 07:52.07 | fitzel | chan_capi.c:1023: structure has no member named `delivery' |
| 07:52.08 | fitzel | chan_capi.c:1073: structure has no member named `callerid' |
| 07:52.10 | TheEmperor | hello, what's the latest stable version of *? |
| 07:52.11 | fitzel | chan_capi.c:1074: structure has no member named `dnid' |
| 07:52.12 | fitzel | chan_capi.c: In function `pipe_msg': |
| 07:52.15 | fitzel | chan_capi.c:1499: structure has no member named `delivery' |
| 07:52.16 | fitzel | chan_capi.c:1500: structure has no member named `delivery' |
| 07:52.17 | Inv_arp | pastebin.ca if needed |
| 07:52.19 | fitzel | chan_capi.c:1724: structure has no member named `dnid' |
| 07:52.24 | riksta | don't suppose anyone in the UK has an old 1U server they would like to sell? doesn't have to be good specification |
| 07:52.25 | fitzel | chan_capi.c:1724: structure has no member named `dnid' |
| 07:52.26 | fitzel | chan_capi.c:1724: structure has no member named `dnid' |
| 07:52.30 | fitzel | chan_capi.c:1724: structure has no member named `dnid' |
| 07:52.33 | fitzel | chan_capi.c:1724: structure has no member named `dnid' |
| 07:52.34 | fitzel | chan_capi.c: In function `load_module': |
| 07:52.36 | fitzel | chan_capi.c:2793: warning: passing arg 4 of `ast_channel_register' from incompatible pointer type |
| 07:52.39 | fitzel | ARGLLL |
| 07:52.42 | fitzel | Here is the pastebin: http://pastebin.ca/6710 |
| 07:52.45 | fitzel | Grmbl. |
| 07:52.49 | TheEmperor | hello, what's the latest stable version of *? |
| 07:52.55 | fitzel | 1.0.6 |
| 07:52.56 | riksta | TheEmperor: topic |
| 07:53.03 | riksta | fitzel: is that really neccessary? |
| 07:53.04 | fitzel | Ztopic |
| 07:53.08 | pashah | morning |
| 07:53.11 | TheEmperor | ? |
| 07:53.24 | TheEmperor | riksta: what do you mean.. |
| 07:53.33 | riksta | TheEmperor: read the bloody topic |
| 07:53.41 | TheEmperor | o |
| 07:53.43 | TheEmperor | sorry |
| 07:55.47 | Inv_arp | hm wonder how i can get a telco "blah number is disconnected" operator recording |
| 07:55.54 | *** join/#asterisk dg1nsw (~schulte@212.34.175.147) |
| 07:55.55 | fitzel | riksta, it was not my intention. |
| 07:56.11 | fitzel | I am bit of trigger happy this morning. |
| 07:58.03 | fitzel | Is anybody using a digi datafire card here on debian? |
| 08:04.18 | *** join/#asterisk afrosheen (~afro@c-24-0-139-118.client.comcast.net) |
| 08:05.27 | afrosheen | yeup. |
| 08:08.07 | izo | <PROTECTED> |
| 08:09.12 | par | This is not the first device from the company that support both GSM and Wi-Fi, but it is the first one that can hand-over an active VoIP call to GSM via Wi-Fi (WLAN), when you travel beyond the WLAN coverage. Vice-versa, only .PBX anchored. calls will continue when moving from GSM to WLAN. |
| 08:10.06 | par | how do you pbx anchor a gsm 3g call |
| 08:10.27 | par | must mean that you need an actual circuit |
| 08:23.16 | par | i guess we'll wait for support for that one :/ |
| 08:23.44 | wildcard0 | hey. what's the general opinion on asterisk@home? |
| 08:24.18 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:24.35 | *** join/#asterisk flewid (~flewid@24.42.244.169) |
| 08:24.39 | flewid | sup |
| 08:25.13 | flewid | question - i have two tdm cards in a box, and for some reason it seems if i softboot, it doesn't detect both cards, just the first one. I have to power down before powering up for it to recognize both cards properly |
| 08:25.15 | flewid | this common?> |
| 08:25.17 | flewid | :) |
| 08:25.32 | flewid | sory, tdm04/40 |
| 08:27.41 | *** join/#asterisk matjing (~Miranda@62.8.64.33) |
| 08:28.22 | RestLessGemini | I installed asterisk@home last nite.. everything is great,though i dont know the login password for maintainance section in AMP, also it didnt setup my x100p, well it did first but as soon as i plugged phone line into x100p, linux got hang and after reboot, it gave a handfull errors on modprobing wcfxo |
| 08:28.24 | ManxPower | flewid, I've seen it before. |
| 08:28.26 | *** join/#asterisk TrevMeister (~thammonds@ip68-4-223-70.oc.oc.cox.net) |
| 08:29.17 | wildcard0 | RestLessGemini, well since i was planning on using all SIP/IAX stuff, i prolly won't run into those problems :) |
| 08:29.29 | wildcard0 | i'll have to find the login/pass tho |
| 08:30.57 | RestLessGemini | well yeah .. its working great with SIP/IAX stuff |
| 08:31.22 | RestLessGemini | well i can tell you amp admin password |
| 08:31.33 | wildcard0 | oh what is it? :) |
| 08:31.33 | wildcard0 | hehe |
| 08:31.41 | RestLessGemini | user= wwwadmin, password = password |
| 08:31.52 | RestLessGemini | It took me 30 min just to find the passwrod :) |
| 08:31.55 | RestLessGemini | password* |
| 08:31.59 | wildcard0 | haha |
| 08:32.16 | RestLessGemini | but i am still looking for maintainance section password |
| 08:32.16 | wildcard0 | well im glad i dun have to go through that |
| 08:32.26 | RestLessGemini | memo me if you managed to find it |
| 08:32.33 | wildcard0 | when it's done downloading, i'll look for that too |
| 08:32.58 | RestLessGemini | tahnks |
| 08:33.13 | RestLessGemini | thanks |
| 08:35.41 | flewid | sorry, went to get a cofffe :) sec |
| 08:36.04 | flewid | ManxPower: so it's not just me, any way you know of to fix it? |
| 08:36.28 | modulus_ | hi |
| 08:36.28 | flewid | i mean, it's not like it'll be rebooting often, but right now i'm rebooting it frequently to make sure that it comes back up as planned upon a reboot |
| 08:36.33 | flewid | so it's annoying :) |
| 08:37.48 | *** join/#asterisk Tray (~traytray@ip24-253-102-200.lv.lv.cox.net) |
| 08:38.23 | Tray | Does anyone know how to setup a t100p with hdlc? |
| 08:38.29 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 08:39.34 | flewid | so anyone here going to VON? |
| 08:40.19 | *** join/#asterisk Inv_arp (junya@adsl-3-247-135.mia.bellsouth.net) |
| 08:42.27 | ta[i]nted | is there a way to mimic calls in asterisk? |
| 08:42.37 | ta[i]nted | i want to see how many concurrent calls my box can handle |
| 08:43.53 | Inv_arp | ta[i]nted: think there's sip call generators out there |
| 08:45.08 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:46.34 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 08:46.45 | teemu-x | If I'm instaling asterisk 1.0.2, what version of libpri/zaptel should I use? |
| 08:48.12 | Inv_arp | teemu-x: samne as the one u installing |
| 08:48.57 | teemu-x | ok now I just have to figure out where to get those |
| 08:49.53 | Inv_arp | teemu-x: why not get 1.06? |
| 08:50.23 | teemu-x | H323 doesnt seem to work |
| 08:50.24 | *** join/#asterisk FryGuy- (fryguy@c-24-23-19-33.client.comcast.net) |
| 08:51.37 | ManxPower | teemu-x, You should not install Asterisk 1.0.2 |
| 08:51.47 | teemu-x | why? |
| 08:52.03 | ManxPower | because there are massive numbers of bug fixes after 1.0.2 |
| 08:52.14 | teemu-x | it worked fine, and now that someone updated asterisk to 1.0.6, h323 got broken |
| 08:52.18 | ManxPower | use 1.0.3 is you have to. |
| 08:53.09 | ManxPower | teemu-x, you rebuilt the chan_h323 for the new version of Astersisk, right? And read the updated /path/to/src/asterisk/channels/h323/README, right? |
| 08:54.01 | ManxPower | If 1.0.6 H323 was REALLY broken I'm pretty sure we would have heard about it before now. |
| 08:54.59 | teemu-x | I've done everything like it's supposed to, I think |
| 08:55.26 | ManxPower | teemu-x, file a report on bugs.digium.com then. |
| 08:55.53 | teemu-x | when called party answers (= picks up the phone), call gets hanged up immediately (if either end is H323) |
| 08:56.23 | *** join/#asterisk tzafrir_home (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 08:56.24 | teemu-x | like I'm going to report a bug, I lack the haxor skillz to produce valid debug information |
| 08:56.37 | wildcard0 | well, start with an strace |
| 08:56.47 | wildcard0 | then you'll know where it's hanging at least |
| 08:57.21 | wildcard0 | if you can narrow it down to a particular area, you'll know where to being |
| 08:57.24 | teemu-x | what do you mean by strace? |
| 08:57.25 | wildcard0 | *begin |
| 08:57.30 | wildcard0 | man strace |
| 08:57.37 | ManxPower | teemu-x, Sounds like a codec problem to me. |
| 08:57.42 | ManxPower | disallow=all and allow=ulaw |
| 08:57.46 | wildcard0 | it shows you ... basically...what function a program is in |
| 08:57.47 | *** join/#asterisk syslinux (syslinux@203-173-148-209.bliink.ihug.co.nz) |
| 08:57.56 | wildcard0 | that seems reasonable |
| 08:58.25 | teemu-x | no manual entry for strace, and these codecs have worked fine for few months with asterisk 0.8 something and 1.0.2 |
| 08:58.35 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 08:58.57 | rvhi | hi, when I park, i keep getting this error and there is no moh |
| 08:59.06 | wildcard0 | rvhi, is there? :) |
| 08:59.07 | teemu-x | I have only gsm allowed now |
| 08:59.10 | rvhi | res_musiconhold.c:340 monmp3thread: Only wrote -1 of 640 bytes to pipe |
| 08:59.25 | tzafrir | strace command line with parameters |
| 08:59.26 | ManxPower | teemu-x, You don't have a bandwidth= line, do you? |
| 08:59.33 | teemu-x | no |
| 08:59.35 | Inv_arp | teemu-x: if everything was fine why they upgrad |
| 08:59.39 | teemu-x | in h323.conf? |
| 08:59.39 | ManxPower | teemu-x, so you have disallow=all and allow=gsm? |
| 08:59.44 | teemu-x | yes manx |
| 08:59.57 | ManxPower | teemu-x, Well, h323.conf is where h323 is configured for Astersisk. |
| 09:00.03 | rvhi | if i put someone on hold, i got moh |
| 09:00.04 | ManxPower | teemu-x, post to the mailing list first, it may be a config problem. |
| 09:00.06 | teemu-x | there were some features missing/not working on 1.0.2 |
| 09:00.30 | ManxPower | rvhi, I seem to recall a cvs update that fixed that problem |
| 09:00.43 | teemu-x | but I'd rather take 1.0.2 with missing features than 1.0.6 thats not working at all (all except couple of users have h323 hardware phones) |
| 09:00.46 | rvhi | i am using stable version |
| 09:00.57 | ManxPower | teemu-x, 1.0.x will never have new features added. |
| 09:01.32 | tzafrir | teemu-x, is the PBX exposed to the internet? if so, you should know that there was a security advisory after 1.0.3 |
| 09:02.00 | teemu-x | :( |
| 09:02.40 | wildcard0 | tzafrir, url? |
| 09:02.48 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:03.07 | tzafrir | As for strace, you probably don't have that package installed. Which means you should install it, as no system is complete without strace ;-) |
| 09:04.33 | wildcard0 | hehe |
| 09:05.03 | teemu-x | administrating a linux box is something i really wish not to do, as I'd probably wipe the box clean if I'd try to do something like that |
| 09:05.27 | teemu-x | upgrading asterisk is way too much for me already |
| 09:06.29 | teemu-x | or should I say that everything would work fine (working asterisk in few minutes), except that getting h323 to work takes always solid week of banging my head to wall |
| 09:06.29 | rvhi | teemu-x, is 1.0.6 not working? I am thinking about upgrade. |
| 09:06.58 | tzafrir | wildcard0, http://www.sineapps.com/print.php?rssid=430 |
| 09:07.00 | Inv_arp | rvhi: your current setup works? why upgrade |
| 09:07.06 | wildcard0 | thanks |
| 09:07.25 | tzafrir | teemu-x, what distro? |
| 09:07.28 | teemu-x | upgrade 102->106 broke h323 calls and I do know whre to problem is |
| 09:09.07 | wildcard0 | tzafrir, there were taken care of in >1.0.3 ? |
| 09:09.22 | tzafrir | fixed in 1.0.4 |
| 09:09.47 | cjk | hi, is there an application which does just print some output on the cli |
| 09:09.51 | teemu-x | tza: I guess this is some debian |
| 09:10.44 | ManxPower | cjk, "show application noop" |
| 09:15.14 | cjk | ManxPower, thanks |
| 09:16.30 | *** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 09:17.01 | kamran | any one using latest CVS |
| 09:17.26 | kamran | and ever used stable |
| 09:17.28 | moonwick | Concurrent Versions System (CVS) 1.11.5-FreeBSD (client/server) |
| 09:17.34 | moonwick | I dunno, how recent is that? |
| 09:18.11 | wildcard0 | Concurrent Versions System (CVS) 1.11.17 (client/server) |
| 09:18.15 | wildcard0 | im a bit ahead :) |
| 09:18.29 | kamran | i am talking about asterisk CVS version |
| 09:18.34 | darkskiez | hahahaha |
| 09:19.15 | Inv_arp | jajaja |
| 09:19.56 | kamran | any one using latest asterisk from CVS |
| 09:19.56 | darkskiez | Concurrent Versions System (CVS) 1.12.9 (client/server) |
| 09:20.12 | darkskiez | wow, debian more recent than another distro. |
| 09:20.24 | wildcard0 | you win |
| 09:20.37 | wildcard0 | ya. im amazed. debian usually has software from the late 50s |
| 09:20.45 | moonwick | nah, that's FreeBSD |
| 09:20.59 | wildcard0 | i think last time i did an apt-get from stable, a relay came in the mail |
| 09:23.14 | *** join/#asterisk jarod0820 (~xian-lian@61.173.22.140) |
| 09:24.21 | jarod0820 | Hello,everyone.I am a newbie. |
| 09:24.28 | wildcard0 | just born? |
| 09:24.43 | jarod0820 | A newbie for VOIP |
| 09:25.02 | jarod0820 | Exactly now? |
| 09:25.13 | RestLessGemini | lol |
| 09:26.41 | modulus_ | i'm a newbie too |
| 09:26.45 | modulus_ | jarod let's help each other! |
| 09:26.50 | modulus_ | jbot hug jarod0820 |
| 09:26.52 | jbot | ACTION hugs jarod0820 |
| 09:26.58 | modulus_ | jbot hug me |
| 09:27.00 | jbot | ACTION hugs modulus_ |
| 09:27.09 | modulus_ | jbot drink a beer |
| 09:27.11 | jbot | I don't want to drink a beer |
| 09:27.16 | jarod0820 | wait |
| 09:27.18 | wildcard0 | jbot get me a beer |
| 09:27.19 | jbot | what do I look like?! |
| 09:27.22 | wildcard0 | damn |
| 09:27.25 | wildcard0 | heh |
| 09:28.15 | jarod0820 | How can I help you!!! |
| 09:28.48 | modulus_ | update asterisk so the cli (asterisk -r) captures SIGEOT |
| 09:28.59 | *** join/#asterisk Jas_Williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 09:29.00 | modulus_ | and (correctly) processes it |
| 09:29.21 | *** join/#asterisk cereal_ (~nico@gifu.newel.net) |
| 09:29.24 | cereal_ | Hi all |
| 09:29.30 | modulus_ | hello world! |
| 09:29.36 | jarod0820 | !! |
| 09:29.55 | cereal_ | I have a problem with B-channels allocating on ISDN PRI can someone helmp me ? |
| 09:31.03 | modulus_ | hi jarod0820 |
| 09:31.17 | modulus_ | i sure am |
| 09:31.40 | *** join/#asterisk jas_williams (~Jason@host81-155-66-178.range81-155.btcentralplus.com) |
| 09:31.44 | jarod0820 | !!! |
| 09:31.51 | modulus_ | ### |
| 09:31.55 | jarod0820 | ??? |
| 09:31.59 | jas_williams | ... |
| 09:32.00 | modulus_ | SHE BANG |
| 09:32.03 | modulus_ | SHE MOVE |
| 09:32.27 | jarod0820 | What's this!! |
| 09:32.38 | modulus_ | i dunno!! |
| 09:32.44 | jarod0820 | ~~~~~~~~~ |
| 09:32.49 | modulus_ | i'm out of cigs |
| 09:32.50 | modulus_ | grrr |
| 09:33.23 | jarod0820 | i have |
| 09:33.25 | modulus_ | i guess i'll just settle for wine and some tetrahydrocannabinol |
| 09:33.39 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 09:33.50 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 09:34.12 | cereal_ | Does someone know how to fix the max number of channels for a certain service (agiscript) I t must be in zapata.conf right ? |
| 09:35.18 | jas_williams | cereal_: Max channels there is no max channels ? |
| 09:36.34 | jas_williams | cereal_: Do you whsh to limit the number calls to a service ? |
| 09:36.56 | cereal_ | jas_williams : yep only 10 concurent calls |
| 09:37.03 | cereal_ | 10 as max limits |
| 09:37.06 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:37.22 | ManxPower | cereal_, See "show applications" Pay special attention to SetGroup and CheckGroup options |
| 09:37.54 | cereal_ | ManxPower : ok show applications of wich documentation ? |
| 09:38.04 | ManxPower | cereal_, in the Asterisk CLI. |
| 09:38.13 | ManxPower | You are new, aren't you? |
| 09:38.21 | Zeeek | Manx is up late |
| 09:38.32 | modulus_ | Zeeek is up early |
| 09:38.34 | cereal_ | yep very new |
| 09:38.40 | jas_williams | cereal_: you need to use something like setgroup and check group before calling your agi script |
| 09:38.40 | *** part/#asterisk ScaredyCat (~ScaredyCa@j25065.upc-j.chello.nl) |
| 09:38.43 | ManxPower | Zeeek, I took a nap earlier so I could do some over night upgrades. |
| 09:39.19 | ManxPower | I managed to make the mail server boot, get 12 polycom phones upgraded, upgraded our largest Asterisk box |
| 09:39.30 | ManxPower | ..er..managed to make the mail server NOT boot. |
| 09:39.43 | modulus_ | manx what kinda call volume you got on largest? |
| 09:40.24 | cereal_ | ManxPower : the only problem is i never know on wich B channel the incoming call arrives .. can be the 1 the 10 the 30 etc .. so i cant just put channel => 1-x in zapata.conf |
| 09:40.24 | ManxPower | modulus_, Total of 96 channels, but a bunch of those are DACSd. I figure about 12 channels into Astrisk and 12 channels to the channel bank |
| 09:40.43 | ManxPower | cereal_, Do NOT do that stuff in zapata.conf. |
| 09:40.51 | Zeeek | Manx ah the joys of night work! |
| 09:41.03 | modulus_ | manx, calls per day? |
| 09:41.17 | modulus_ | thousands? tens of thousands? hundreds of thousands? |
| 09:41.40 | jas_williams | cereal_: look at this page in wiki http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup |
| 09:41.47 | wildcard0 | one...BILLION calls |
| 09:41.56 | modulus_ | DUN DUN!! |
| 09:42.18 | modulus_ | jbot moocow? |
| 09:42.18 | jbot | hmm... moocow is the moo cow |
| 09:42.24 | modulus_ | jbot moo cow? |
| 09:42.26 | jbot | ACTION moos at cow |
| 09:42.35 | modulus_ | jbot dogcow? |
| 09:42.36 | jbot | MOOOFF!! |
| 09:43.04 | ManxPower | modulus_, grep 2005-03-01 /var/log/asterisk/cdr-csv/Master.csv | wc -l |
| 09:43.12 | ManxPower | <PROTECTED> |
| 09:43.34 | cereal_ | jas_williams : thx for help will read that |
| 09:43.46 | *** part/#asterisk syslinux (syslinux@203-173-148-209.bliink.ihug.co.nz) |
| 09:44.13 | modulus_ | manx, sounds office environment-ish |
| 09:44.20 | *** join/#asterisk hanseatic (~konversat@80.171.227.2) |
| 09:44.21 | ManxPower | modulus_, Well, yes. |
| 09:44.36 | hanseatic | hello. |
| 09:45.36 | hanseatic | do i have to compile a new kernel to install capi2.0 on debian sarge? |
| 09:46.48 | jas_williams | hanseatic: Depends if Capi2 with the correct options is compiled in your existing kernel, I would do it anyway then you know it is right |
| 09:48.05 | Zeeek | News on tftp front. The server claims that it gets no ACK |
| 09:48.30 | Zeeek | I think it may have to do with the file legth of 129 bytes |
| 09:48.32 | hanseatic | debian's default is hisax... |
| 09:48.51 | hanseatic | thnx @jas |
| 09:49.54 | jas_williams | hanseatic: Then you need to remove the hisax and build in the capi 2 I had to do the same for my slackware |
| 09:50.11 | teemu-x | sigh... now that I finally got good version of pwlib (1.8.1), it fails to compile: "../../ptclib/httpclnt.cxx:385: ambiguous overload for `BOOL ? const char[2] : const PString &'" |
| 09:53.04 | ManxPower | teemu-x, What part of this do you NOT understand?? |
| 09:53.05 | ManxPower | This code runs on Open H.323 v1.12.2 and PWLib v1.5.2. If you use different |
| 09:53.05 | ManxPower | versions, you are on your own. See the Makefile for more details. |
| 09:53.22 | ManxPower | That's direct from the README for chan_h323 for CVS 1.0.x stable. |
| 09:53.31 | tuxinator_linux | Night guys |
| 09:53.43 | teemu-x | channels/h323/readme of *cvs head* says openh323 v1.15.1 and pwlib v1.8.1 |
| 09:53.47 | hanseatic | @jas thnx... it will be my first kernel compilation... is there anything i should be beware of? |
| 09:53.51 | teemu-x | I'm trying cvs head now |
| 09:54.05 | ManxPower | teemu-x, Maybe so, but last I heard you were using 1.0.x stable |
| 09:54.28 | teemu-x | yeah, but since I had no luck I'm now trying cvs head |
| 09:54.59 | teemu-x | tried yesterday that cvs head too, but couldnt find those specific version of openh323/pwlib anywhere |
| 09:55.17 | teemu-x | now that I found them, I tried cvs head again, but no luck since pwlib doesnt compile |
| 09:56.00 | *** part/#asterisk jarod0820 (~xian-lian@61.173.22.140) |
| 09:57.34 | ManxPower | teemu-x, You realize that CVS-HEAD isn't even guarnteed to compile, right? |
| 09:57.56 | teemu-x | yes |
| 09:58.42 | teemu-x | but if I cant get stable to work, I'd better try that |
| 10:00.30 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 10:00.42 | *** join/#asterisk afe ([1QVx5P+zZ@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 10:08.22 | jas_williams | teemu-x: What is wrong with stable ? |
| 10:09.06 | ManxPower | jas_williams, he thinks chan_h323 is broken in 1.0.x stable |
| 10:09.36 | jas_williams | it is not it works fine with the correct libs and configuration |
| 10:10.49 | ManxPower | jas_williams, He's getting dropped h323 calls |
| 10:13.21 | Zeeek | gentlement: in IAX2 what could make a phone unreachable? http://pastebin.ca/6711 |
| 10:13.51 | Zeeek | <PROTECTED> |
| 10:14.15 | Zeeek | the phone is on the same router as asterisk |
| 10:15.16 | *** join/#asterisk strace (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 10:15.19 | strace | hey gang |
| 10:15.19 | TheEmperor | hello |
| 10:15.30 | strace | where can I download mp3 free for commercial usage? |
| 10:15.51 | Zeeek | mp3.com ? |
| 10:15.52 | JamesDotCom | there's a site somewhere around with royalty free mp3s |
| 10:15.58 | TheEmperor | anyone know how to configure zaptel.conf and zapata.conf with a te410p digium card? I've read the docs but am still confused :( |
| 10:15.59 | JamesDotCom | try using google before asking questions like that |
| 10:16.09 | strace | JamesDotCom: done that |
| 10:16.53 | JamesDotCom | keep looking |
| 10:16.57 | JamesDotCom | theyre definitely around |
| 10:17.37 | strace | tell me where then |
| 10:17.54 | Zeeek | Manxpower ? |
| 10:18.19 | ManxPower | Zeeek? |
| 10:18.41 | JamesDotCom | http://directory.google.com/Top/Arts/Music/Sound_Files/Samples_and_Loops/Production_Music_Libraries/Royalty_Free/ |
| 10:18.59 | ManxPower | Zeeek, The only Emperor I'd help this late is Emperor Norton and he's dead. |
| 10:19.14 | Zeeek | Emporer? |
| 10:19.19 | TheEmperor | darn :) |
| 10:19.55 | strace | JamesDotCom: most of those aren't _free_, you have to pay a license for them to be free |
| 10:20.02 | strace | JamesDotCom: but thanks anyway |
| 10:20.09 | Zeeek | trying to figure out why the iaxphone is UNREACHABLE when it is registered and can make calls |
| 10:20.38 | Zeeek | The phone talk to asterisk, they're both on the same subnet |
| 10:21.43 | wildcard0 | firewall? |
| 10:22.01 | Zeeek | IAX on the same subnet |
| 10:22.22 | Zeeek | 192.168.1.205 <=====> 192.168.1.5 |
| 10:22.53 | sambal | do you need to pay licenses for MOH music? or is buying a cd enough? |
| 10:23.46 | Zeeek | license |
| 10:23.46 | ManxPower | Zeeek, iptables running on the asterisk box? |
| 10:23.54 | *** join/#asterisk meppl (~mephisto@pD95428E8.dip.t-dialin.net) |
| 10:23.57 | Zeeek | I don't think so |
| 10:24.00 | ManxPower | sambal, You always need a license. |
| 10:24.16 | sambal | ok :( what kind of? / where to get it? |
| 10:24.22 | ManxPower | MoH would be considered a "public performance" which is prohibited for most music. |
| 10:24.24 | Zeeek | but do you always need iptables ? |
| 10:24.27 | wildcard0 | get some classical. it's out of copyright |
| 10:24.29 | sambal | hmm |
| 10:24.43 | ManxPower | Zeeek, No. many distos install IP tables and run them. |
| 10:24.46 | Zeeek | not out of copyright if the performance has mechanical rights |
| 10:25.00 | ManxPower | wildcard0, Yeah, but there's not a lot of classical where the PERFORMANCE is out of copyright. |
| 10:25.06 | Zeeek | Manx at any rate the IAXy worked fine in this "slot" |
| 10:25.07 | wildcard0 | true |
| 10:25.09 | wildcard0 | but there is some |
| 10:25.21 | sambal | where to get it? :) |
| 10:25.24 | Zeeek | there are a few specific recordings around |
| 10:25.30 | ManxPower | The MoH incouded with Asterisk has a licence to use it with Asteris,. |
| 10:25.52 | wildcard0 | i think there was a gutenburg like project for music that was archiving a bunch of stuff just like that |
| 10:26.02 | sambal | hmm and putting radio as MOH? |
| 10:26.09 | ManxPower | sambal, totally illegal |
| 10:26.20 | sambal | i knkow some company's doing it |
| 10:26.21 | ManxPower | Lots of people do it, but it's still illegal. |
| 10:26.23 | sambal | :) |
| 10:26.24 | wildcard0 | you can usually get a license for that pretty easily if you don't remove the commercials |
| 10:26.43 | ManxPower | wildcard0, contact the radio station, right? |
| 10:26.46 | wildcard0 | yes |
| 10:27.23 | wildcard0 | they usually have a dept for that. or at least a guy |
| 10:27.25 | sambal | what are the regular prices for using a cd? |
| 10:27.26 | Zeeek | Manx I'm trying to figure out which parameters in iax.conf would disturb reachability? I know some stuff must not use qualify for example, but this isn't the problem |
| 10:27.54 | ManxPower | Zeeek, none of them. Unreachable means "got a icmp port unreachable or no response" |
| 10:28.18 | ManxPower | Zeeek, Does ipchains -L give anything other than empty tables? |
| 10:28.22 | Zeeek | let's then - what could be blocking the packets? |
| 10:28.30 | Zeeek | I'll check |
| 10:28.33 | wildcard0 | prolly iptables |
| 10:28.42 | ManxPower | yeah, iptables |
| 10:28.45 | sambal | iptables doesn't list all with -L |
| 10:29.07 | Zeeek | doesn't find any ipchains |
| 10:29.12 | Zeeek | in Slackware 9.1 |
| 10:29.22 | Zeeek | I mean the command |
| 10:29.37 | wildcard0 | iptables -L -v |
| 10:29.38 | ManxPower | Zeeek, I meant "iptables -L" |
| 10:29.49 | ManxPower | prolly -V would help too |
| 10:29.58 | Zeeek | Empty |
| 10:30.26 | wildcard0 | are you binding to 127.0.0.1? or an external address? |
| 10:30.36 | wildcard0 | also if you do "reload" does it give an error in the iax section? |
| 10:30.37 | Zeeek | I've never had a probnlem before with anything connected on the net |
| 10:30.42 | ManxPower | Oh! yeah bindaddr=blah might screw it up |
| 10:30.46 | Zeeek | no error on reload |
| 10:30.54 | sambal | is there a tftp server available which can be bind at a given interface? |
| 10:31.11 | Zeeek | no bindadr in iax.conf |
| 10:31.19 | ManxPower | sambal, most tftp servers are run out of inetd/xinetd and it would be configured there. |
| 10:31.28 | wildcard0 | try adding bindaddr=<external ip> |
| 10:31.28 | Zeeek | there is a tftp server at 192.168.1.60 |
| 10:31.47 | Zeeek | errrrr I'm on the loacal network |
| 10:31.50 | *** join/#asterisk nazgool (~nazgoool@port-83-236-180-106.static.qsc.de) |
| 10:31.52 | nazgool | hi |
| 10:32.09 | wildcard0 | where external = not 127.0.0.1. like use 192.168. whatever |
| 10:32.49 | wildcard0 | also do netstat -au |
| 10:32.52 | wildcard0 | do you see it there? |
| 10:33.06 | wildcard0 | it should show 4569 |
| 10:33.19 | sambal | ManxPower: good point.. :) |
| 10:33.24 | nazgool | do i understand the syntax of this agi command correctly: can i say "CHANNEL STATUS SIP/mysipuser" ? |
| 10:33.25 | Zeeek | udp 0 0 *:4569 *:* |
| 10:34.26 | nazgool | and is there a possibility to ask the channel status for a given channel (e.g. SIP/mysipuser) from console? |
| 10:35.03 | teemu-x | jas_williams: any h323 necessary configuration changes between 1.0.2 and 1.0.6? |
| 10:35.12 | wildcard0 | Zeeek, that looks right. what's trying to connect to it? |
| 10:35.27 | Zeeek | you mean my problem phone? |
| 10:35.37 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 10:35.41 | Zeeek | There are other IAX clients coming in thru NAT btw - |
| 10:36.01 | Zeeek | no problems with my IAXy, nufone, VP, etc |
| 10:36.10 | wildcard0 | sounds like it's a client problem then |
| 10:36.13 | Zeeek | this is a Farfon |
| 10:36.30 | Zeeek | bleeding edge - my blood |
| 10:36.39 | wildcard0 | heh |
| 10:36.51 | wildcard0 | the server seems to be working fine |
| 10:36.56 | Zeeek | to me it is |
| 10:37.00 | wildcard0 | i can't help much with the client. never used it |
| 10:37.16 | Zeeek | I don't have another IAX client here but formerly the IAXy was connected in the same situation and no prob |
| 10:37.31 | wildcard0 | actually...that's a good question. what do you guys think is the best iax or sip softphone out there for free-as-in-speech ? |
| 10:37.39 | Zeeek | well maybe there is a config param on their end. Waiting to catch wasim |
| 10:37.44 | wildcard0 | Zeeek, that really sounds like a client issue |
| 10:37.46 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 10:38.08 | sambal | ManxPower: looks like it isn't supported |
| 10:38.12 | Zeeek | everyone here sbobs them, but the value for money is Grandstream |
| 10:38.32 | Zeeek | I'm looking at a Polycom ip500 now |
| 10:38.35 | wildcard0 | SOFTphone. like a program |
| 10:38.37 | sambal | grandstream isn't bad as all with the latest firmwares |
| 10:38.40 | sambal | at all |
| 10:38.48 | Zeeek | free softphone for linux? |
| 10:38.50 | sambal | especially for testing |
| 10:39.00 | wildcard0 | i'd prefer multiplatform |
| 10:39.18 | wildcard0 | but prolly windoze first as it'd have to be put on boxes that aren't mine |
| 10:39.24 | Zeeek | X-Lite is by far the best in my experience. Again, people love to trash it |
| 10:39.50 | Zeeek | I've used most of the iAX ones too IAXphone and iAXCOMM come to mind |
| 10:40.05 | wildcard0 | i was considering skinning iaxcomm, but it's a HUGE pain to compile |
| 10:40.12 | wildcard0 | like...it requires blood offerings |
| 10:40.34 | wildcard0 | x-lite isn't open source |
| 10:40.49 | Zeeek | if yiou don't specify auth in iax.conf it defaults to plain? |
| 10:40.55 | nazgool | if i'd like to know (inside an AGI script) whether the line SIP/mysipuser is busy, how do i do that? i tried "CHANNEL STATUS SIP/mysipuser", but it tells me -1 although that sip user is registered |
| 10:40.58 | nazgool | any idea? |
| 10:40.58 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 10:41.10 | Zeeek | hey ya know what? There's a new OS SIP client out here in FR |
| 10:41.19 | Zeeek | http://wengo.fr |
| 10:41.29 | Zeeek | Multiplatform - I forgot to tell people here about that |
| 10:41.49 | Zeeek | I haven't used it more than once, but it is available free |
| 10:42.03 | nazgool | do i need to append some number to "SIP/mysipuser" like "SIP/mysipuser-2343" to get a valid channel name? if so, wherefrom do i get that number? |
| 10:42.24 | wildcard0 | nazgool, it might be overkill, but a while back i wrote a patch that would allow you to run console commands in an agi script. it's prolly still in mantis somewhere |
| 10:43.13 | wildcard0 | Zeeek, i get an empty page there |
| 10:43.20 | Zeeek | I'll try |
| 10:44.32 | Zeeek | no it's up but let me try to find the dowload links |
| 10:44.49 | wildcard0 | maybe it's my proxy then |
| 10:45.20 | sambal | put www in front of it |
| 10:45.48 | Zeeek | see if this works: http://www.wengofiles.teaser-hosting.com/WengoSetup-20050207193103.exe |
| 10:46.05 | nazgool | can't i just use "CHANNEL STATUS" for that? |
| 10:46.21 | wildcard0 | no idea. never tried |
| 10:46.52 | Zeeek | wildcard0 HERE we ho: |
| 10:46.54 | Zeeek | http://developer.berlios.de/projects/openwengo/ |
| 10:47.02 | Zeeek | GPL |
| 10:47.20 | Zeeek | aplpha status like my phone |
| 10:47.48 | wildcard0 | cool. i'll check it out as soon as im done burning this disc |
| 10:48.39 | Zeeek | grab this |
| 10:48.40 | Zeeek | http://svn.berlios.de/viewcvs/openwengo/ |
| 10:48.54 | Zeeek | CVS source |
| 10:49.18 | Zeeek | Keep us posted - I forgot all about this thing |
| 10:50.16 | jas_williams | Zeeek: got your farfon going yet ? |
| 10:50.19 | Zeeek | I'll buy a coffee for the first person that compiles this phone |
| 10:50.22 | Zeeek | No! |
| 10:50.29 | Zeeek | well, half - it calls |
| 10:50.31 | Zeeek | sort of |
| 10:50.40 | Zeeek | it's registered but unreachable |
| 10:51.35 | jas_williams | Do the iax packets make it to the phone ? |
| 10:52.04 | jas_williams | What does your sniffer trace show ? |
| 10:52.08 | Zeeek | well when it calls, the call works - though it has unacceptable noise (I think it's a b0rken phone) |
| 10:52.42 | Zeeek | Actually I sniffed the tftp part but not the qualify packets |
| 10:52.50 | Zeeek | let me start the sniff again |
| 10:55.45 | jas_williams | There is a nat between you and * so the qualify is needed to keep the nat ports open |
| 10:55.56 | Zeeek | no NAT same side |
| 10:56.32 | jas_williams | oh Sorry I miss understood a coment from Manx, Then it should stay registered without the qualify |
| 10:56.46 | jas_williams | and incoming calls to the phone should work |
| 10:57.07 | Zeeek | I put a host=192.168.1.205 and asterisk now complains it isn't dynamic |
| 10:57.19 | Zeeek | why? |
| 10:57.46 | jas_williams | You cannot have a specified host and host as dynamic, have one or the other |
| 10:57.56 | Zeeek | it is one or the other |
| 10:58.26 | Zeeek | ah wait a sec - maybe that was specified in the phone conf |
| 10:58.33 | jas_williams | You need to stop the phone from registering when you have host=192,,,, |
| 10:59.15 | Zeeek | ok anyway |
| 10:59.28 | Zeeek | no packets are sent to the phone |
| 10:59.42 | Zeeek | well, none are received |
| 10:59.57 | sambal | Zeeek: did you remove the dynamic line aswell? |
| 11:00.06 | Zeeek | yeah |
| 11:00.18 | jas_williams | * will not accept registration from a phone that has host=192.168.1.205 in its config |
| 11:00.25 | Zeeek | 192.168.1.205 (S) 255.255.255.255 4569 UNREACHABLE |
| 11:00.47 | Zeeek | it works the same with no host or host=dynamic |
| 11:00.59 | wildcard0 | jas_williams, why not? |
| 11:01.01 | jas_williams | Is asterisk sending packets iax2 debug |
| 11:01.06 | Zeeek | oops no it complains "host is not dynamic" |
| 11:01.26 | jas_williams | wildcard0: That is the way * is architected |
| 11:01.49 | Zeeek | interesting verb |
| 11:01.57 | jas_williams | :) |
| 11:01.58 | Zeeek | are you claiming first use? |
| 11:02.03 | Zeeek | you should |
| 11:02.08 | wildcard0 | i don't think i've ever had that issue |
| 11:02.27 | jas_williams | yes architected (copyright jas_williams 2005) |
| 11:02.55 | Zeeek | sniffing 192.168.1.205 nothing coming in from anywhere - or going out |
| 11:03.31 | Zeeek | hmmm not much of a sniff |
| 11:03.39 | Zeeek | since I just was making a call |
| 11:04.15 | Zeeek | oops - forgot I wasn't on the hub anymore |
| 11:04.30 | jas_williams | Zeeek: ahh. |
| 11:05.09 | Zeeek | ok now I'm seeing some action |
| 11:05.29 | Zeeek | there is a POKE and an ACK |
| 11:06.06 | Zeeek | so the phone does seem to be getting and acking the poke |
| 11:06.09 | Zeeek | from ast |
| 11:06.42 | Zeeek | I have someone coming here for lunch so I'll obviously be interrupted right on the verge of a major important discovery |
| 11:07.27 | jas_williams | Zeeek: If you turn off the qualify, does the phone stay registered and can you call it ? |
| 11:07.50 | jas_williams | keep host as dynamic |
| 11:08.10 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
| 11:08.44 | Zeeek | I think so because I had qualify off thinking that'd help - it does with some SIP providers |
| 11:09.23 | jas_williams | So in that case * should at least attempt to make the call to the phone |
| 11:09.52 | Zeeek | I'm seeing REGREQ, REGAUTH and REGACK all looking normal |
| 11:09.56 | jas_williams | if staus = unreachable then * will not attempt to call the peer as far as i know |
| 11:10.04 | Zeeek | no it doesn't |
| 11:10.05 | ozJames79 | Hi can anyone help with a startup problem i am having with * i can tell it to startup by putting the command in inittab and in rc.local -vvvvc but onces its loaded i cant issue a asterisk -r yet it is running because i can make and recieve calls ...any ideas .. thanks :) |
| 11:10.21 | jas_williams | What does your trace give |
| 11:10.25 | Zeeek | ozJames look up safe_asterisk |
| 11:10.44 | ozJames79 | zeeek: i did that and console is set to yes |
| 11:10.51 | jas_williams | during the call attemot, and could it be a codec negotiation issue |
| 11:11.31 | Zeeek | if asterisk is run as root you must be root to run asterisk -r |
| 11:12.04 | ozJames79 | yep i am running as root |
| 11:12.35 | Zeeek | but are you using the safe_asterisk script ? |
| 11:13.20 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 11:13.28 | jas_williams | ozJames79: Include the line safe_asterisk in your rc.local rather than asterisk -vvvvc |
| 11:13.29 | ozJames79 | yep |
| 11:13.37 | ozJames79 | tried that also |
| 11:13.49 | ozJames79 | it does load i jsut cant get the console |
| 11:14.00 | Zeeek | what is the message? |
| 11:14.34 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 11:14.38 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:14.41 | RoyK | ~seen wasim |
| 11:14.43 | jbot | wasim <~wasim@203.81.213.118> was last seen on IRC in channel #asterisk, 1d 21h 39m 26s ago, saying: 'yay! fresh feta cheese!'. |
| 11:14.45 | ozJames79 | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 11:15.01 | Zeeek | <PROTECTED> |
| 11:15.31 | ozJames79 | hmm i am logged in as root |
| 11:15.48 | Zeeek | and you see asterisk in ps aux ? |
| 11:15.53 | jas_williams | ozJames79: do a netstat do you see unix 2 [ ACC ] STREAM LISTENING 486 /var/run/asterisk.ctl |
| 11:16.44 | Zeeek | I see connected, not listening |
| 11:16.53 | Zeeek | but I guess that's beacuse I am... |
| 11:17.02 | jas_williams | Zeeek: You have a remote session connected :) |
| 11:17.10 | Zeeek | per cise ly |
| 11:17.57 | Zeeek | jas how can I see traffic coming from 192.168.... on the asterisk box? |
| 11:18.10 | Zeeek | it's iftraf or something? |
| 11:18.24 | Zeeek | with a filter for 192.168.1.205 |
| 11:18.28 | *** join/#asterisk DHuang (~DHuang@203.46.67.60) |
| 11:18.35 | [ro]nic3try | is possible to have asterisk restarting after a crash ? |
| 11:18.58 | DHuang | yes... put in the supervise |
| 11:19.22 | ozJames79 | check this out http://www.pastebin.com/248255 |
| 11:19.57 | [ro]nic3try | ? |
| 11:20.24 | [ro]nic3try | to have asterisk restart by itself |
| 11:20.28 | DHuang | check the ctl does it exist? |
| 11:20.39 | jas_williams | Zeeek: Not sure |
| 11:20.46 | Zeeek | found it |
| 11:21.38 | ozJames79 | no /var/run/asterisk/asterisk.ctl doesnt exist |
| 11:22.20 | DHuang | is it a bug for Asterisk to unable to check register if you have 2 SIP phone under the same NAT network?? |
| 11:22.35 | DHuang | ozJ: that answer your question. |
| 11:23.07 | jas_williams | DHuang: No it is a limitation with SIP each phone has to be listening on a different sip port say 5060 and 5061 |
| 11:23.50 | DHuang | jw: I see... shall change the SIP phone and try... thanks. ;-p |
| 11:24.24 | ozJames79 | DHuang: not really no as asterisk is running i can make and recieve calls |
| 11:24.30 | jas_williams | Don't forgrt to change your port forwarding rules as well |
| 11:24.46 | DHuang | jw: thanks.. |
| 11:25.02 | DHuang | OJ: did you compie the asterisk yourself?? |
| 11:25.14 | ozJames79 | yes i did |
| 11:25.53 | DHuang | OJ: and you renamed it so safe_asterisk? |
| 11:27.09 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:27.20 | Zeeek | I'm totally DUMB-founded |
| 11:27.43 | Zeeek | On both sides, the packets are passing on 4569 REG,ACK all that stuff |
| 11:27.53 | Zeeek | yes the phone remains UNREACHABLE |
| 11:27.54 | *** join/#asterisk MuppetMaster (~muppetmas@a82-92-73-185.adsl.xs4all.nl) |
| 11:28.01 | MuppetMaster | Hello everyone |
| 11:28.11 | MuppetMaster | I am getting this parsing error: Mar 2 12:24:20 WARNING[25229]: ast_expr.y:483 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: |
| 11:28.19 | DHuang | jw: the SIP debug shows that Nat (no) <--- that should be yes?? |
| 11:28.26 | MuppetMaster | With this command in extensions.conf: exten => _9X.,5,GotoIf($[$[${ENUM:0:3} = SIP] | $[${ENUM:0:3} = IAX]]?6:8) |
| 11:28.31 | ozJames79 | i created the directory /var/run/asterisk now i can connect to cli |
| 11:28.32 | MuppetMaster | Can anyone spot the error? As I can not. |
| 11:28.52 | DHuang | OJ: did you do a "make install" |
| 11:29.32 | ozJames79 | yep |
| 11:29.40 | RaYmAn-Bx | MuppetMaster: try doing a NoOp with the same parameters to check that it returns what is expected |
| 11:29.45 | DHuang | OJ: try safe_asterisk -r |
| 11:30.09 | RaYmAn-Bx | MuppetMaster: and try the seperate parts, well..seperate :) And make sure it's assembled properly |
| 11:30.25 | Mother_ | hi all |
| 11:30.39 | Mother_ | Zeeek: that happens to me too |
| 11:30.55 | MuppetMaster | RaYmAn-Bx: How would I format the NoOp to check the syntax? |
| 11:31.02 | Mother_ | I have two * and every so often one spends a while (1 minute up to 1 hour) unreacheable |
| 11:31.17 | Zeeek | on the same local network? |
| 11:31.23 | Mother_ | ah no |
| 11:31.34 | Mother_ | they are each on their own DSL behind NAT |
| 11:31.39 | Zeeek | and I see by =sniff they are talking to each other |
| 11:31.45 | *** join/#asterisk Astinus_ (~abba@213.167.111.138) |
| 11:32.13 | *** join/#asterisk __Sparks_ (ringding@bb-195-172-54-59.ukonline.co.uk) |
| 11:32.24 | RaYmAn-Bx | MuppetMaster: well..split it up..like $[${ENUM:0:3} = SIP] for example.. |
| 11:32.25 | Mother_ | what have you, * and a phone? or both are PCs? I'm asking in case you can run ethereal on each |
| 11:32.37 | jas_williams | Zeeek: Remove the qualify lines, and restart * |
| 11:32.44 | MuppetMaster | RaYmAn-Bx: Okay, will give that a try. |
| 11:32.52 | RaYmAn-Bx | it generally helps to make sure each part is correct and if that's the case then make sure it's assembled together correctly to form the final result |
| 11:32.53 | Zeeek | I've done that - but I can do it again |
| 11:33.11 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 11:33.18 | fishboy1669 | morning guys |
| 11:33.20 | RaYmAn-Bx | MuppetMaster: my initial thought is whether or not it's safe to use | inside a $[] block |
| 11:33.28 | Mother_ | is there a way to check if the compilation of a driver with some changes has worked OK? |
| 11:33.37 | __Sparks_ | Anyone here using Xorcom Rapid? |
| 11:33.38 | Zeeek | you meant restart and not just reload? |
| 11:33.54 | tzafrir | me... |
| 11:34.13 | Mother_ | I'm playing with the debounce settings on wctdm.c to see if I can cure this hangup problem |
| 11:34.14 | MuppetMaster | RaYmAn-Bx: I have used an or('|') before and there are lots of examples of it on the Wiki... |
| 11:34.39 | jas_williams | Zeeek: Yes, To clear the registartion times may be |
| 11:34.40 | MuppetMaster | RaYmAn-Bx: The funny part is the logic works fine and just as it should. Just get the error when the case is false, but even then it goes to 8 as it should. |
| 11:35.29 | RaYmAn-Bx | MuppetMaster: I don't know then |
| 11:35.49 | Zeeek | jas_ you da man! This would have been solved hours ago if I had thought about restarting! |
| 11:36.08 | Zeeek | now it's unmonitored but callable - much better |
| 11:36.37 | Zeeek | but the clock is still wrong :) |
| 11:37.47 | Zeeek | thanks jas_ that did help a lot |
| 11:37.51 | jas_williams | Zeeek: So it looks like the farfon has an issue with the qualify code. Talk to wasim about that :) |
| 11:38.03 | Zeeek | I have one last issue - the noise |
| 11:38.10 | MuppetMaster | RaYmAn-Bx: Now it is giving me two errors now that I have split them up (same error as before). |
| 11:38.13 | Mother_ | AAAAAAARGH |
| 11:38.14 | Zeeek | see my report on FWD forum |
| 11:38.20 | MuppetMaster | exten => _9X.,5,GotoIf($[${ENUM:0:3} = SIP]?7:6) |
| 11:38.27 | MuppetMaster | exten => _9X.,6,GotoIf($[${ENUM:0:3} = IAX]?7:9) |
| 11:38.29 | Mother_ | this is SO frustrating |
| 11:38.40 | jas_williams | That could be a HW issue, |
| 11:38.41 | MuppetMaster | Mar 2 12:37:20 WARNING[25229]: ast_expr.y:483 ast_yyerror: ast_yyerror(): syntax error: syntax error; Input: |
| 11:38.54 | RaYmAn-Bx | is that the entire error? |
| 11:39.03 | MuppetMaster | <PROTECTED> |
| 11:39.15 | jas_williams | MuppetMaster: are you running the correct version of Bison ? |
| 11:39.24 | MuppetMaster | Is the bit on the next line, with ^^^ under the = sign... |
| 11:39.44 | MuppetMaster | jas_williams: Which version of Bison should I be running (I think I am, but you never know). |
| 11:39.51 | RaYmAn-Bx | actually, try putting quotes around both the ${} bit and the "SIP" bit |
| 11:40.00 | MuppetMaster | Ah, good idea. |
| 11:40.28 | MuppetMaster | Bison 1.875 |
| 11:41.33 | MuppetMaster | RaYmAn-Bx: Tried with "SIP" and "IAX" no joy, will now also add "${}" to see if that works. |
| 11:42.36 | MuppetMaster | RaYmAn-Bx: Putting " " around SIP and ${} did the trick! |
| 11:42.50 | MuppetMaster | Looks like some changes are needed to the Wiki and e164.org's examples... |
| 11:43.08 | fishboy1669 | hi zeeek hows life? |
| 11:43.31 | MuppetMaster | What is strange is that the logic worked anyway... |
| 11:44.29 | MuppetMaster | Also works as one command: exten => _9X.,5,GotoIf($[$["${ENUM:0:3}" = "SIP"] | $["${ENUM:0:3}" = "IAX"]]?6:8) |
| 11:44.32 | MuppetMaster | Thanks! |
| 11:45.47 | MuppetMaster | Another question. Is there anyway to launch an AGI command from extensions.conf in the background and continue with the next extension, or must one always wait for a return? |
| 11:47.04 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 11:48.49 | jas_williams | MuppetMaster: I just updated example 2 on the wikipage does that look good ? http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+GotoIf |
| 11:49.05 | jalsot | hi |
| 11:49.07 | Mother_ | why the heck would my SIP phones start ringing again when I hangup a call coming from a remote * over IAX2? |
| 11:49.25 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 11:49.30 | Mother_ | anyone come across this before? local calls to the PSTN work OK, phones only ring once, hangup is normal, etc. |
| 11:50.01 | MuppetMaster | jas_williams: Should actually look like this (or at least what I got to work) GotoIf($[$["${ENUM:0:3}" = "SIP"] | $["${ENUM:0:3}" = "IAX"]]?6:8) |
| 11:50.04 | Mother_ | but when the PSTN call from the other * comes in over IAX2, I get one ring, then when hanging up, a second ring, obviously to an empty line when picked up |
| 11:50.05 | *** part/#asterisk DHuang (~DHuang@203.46.67.60) |
| 11:50.15 | MuppetMaster | jas_williams: I tried without the " around the ${} and that did not work. |
| 11:50.23 | jas_williams | MuppetMaster: Just noticed that changed it to suit |
| 11:50.27 | *** join/#asterisk meppl (~mephisto@pD95428E8.dip.t-dialin.net) |
| 11:50.31 | MuppetMaster | jas_williams: So when comparing a string to a string it appears to need quotes around both sides of the comparison. |
| 11:50.41 | MuppetMaster | jas_williams: Great! |
| 11:51.03 | Mother_ | anyone? |
| 11:51.44 | Mother_ | hmmm |
| 11:52.10 | Mother_ | why do I get this "Spawn extension (iax-inbound, s, 2) exited non-zero on 'IAX2/mother@mother/2" then when the call is hungup after the second ring |
| 11:52.22 | Mother_ | I get this Spawn extension (iax-inbound, h, 2) exited non-zero on 'IAX2/mother@mother/2 |
| 11:52.34 | Mother_ | which would be the hangup extension, which I don't have configured anywhere |
| 11:52.55 | jalsot | I have a compilation problem of CVS-HEAD, can anybody take a look? http://pastebin.ca/6715 |
| 11:53.51 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 11:54.03 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 11:54.10 | Mother_ | when does the hangup extension come into effect??? |
| 11:54.21 | Mother_ | my client is screaming at me at this very minute |
| 11:54.26 | Mother_ | AAARGH |
| 11:54.36 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:56.48 | MuppetMaster | So any thoughts on how to launch an AGI in the background and carry on with the next extension? Or does it need to be done programatically within the AGI itself? |
| 11:57.16 | MuppetMaster | Where the AGI launches a process to do the work and returns to the Asterisk? |
| 11:57.48 | RoyK | ~seen wasim |
| 11:57.50 | jbot | wasim <~wasim@203.81.213.118> was last seen on IRC in channel #asterisk, 1d 22h 22m 33s ago, saying: 'yay! fresh feta cheese!'. |
| 11:57.52 | *** join/#asterisk LarsAC (~chatzilla@134.130.124.227) |
| 11:59.52 | LarsAC | where can I find info how voiceboxes are stored? |
| 12:01.31 | Mother_ | n, |
| 12:01.43 | visik7 | is pbx patented ? |
| 12:02.29 | *** part/#asterisk hanseatic (~konversat@80.171.227.2) |
| 12:06.05 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 12:06.23 | *** join/#asterisk Tili (~Tili@202-133-65-212-dialup.sat.net.pk) |
| 12:06.49 | ta[i]nted | MuppetMaster what are u trying to do? |
| 12:07.50 | MuppetMaster | ta[i]nted: I send a all notification (with CLI, etc) via a jabber server. The issue is, the jabber server can be a little slow, so it delays the call by a second or two (I playback ringing) before it actually moves on to the next extension which is a dial command. |
| 12:07.51 | fishboy1669 | anyone here in uk using x100p fxo cards? |
| 12:08.01 | nazgool | a question about groups: when i use the SetGroup command, will that in itself increment the GROUPCOUNT for that group? and how are they decremented again? when the call is hung up? or does one have to somehow do it manually? |
| 12:08.05 | MuppetMaster | ta[i]nted: So what I would like to do is launch it in the background and carry on. |
| 12:08.26 | MuppetMaster | ta[i]nted: Incidentally, I am using PHP to do this at the moment along with the http://phpagi.sourceforge.net. |
| 12:08.39 | MuppetMaster | ta[i]nted: Along with a jabber class I picked up along the way. |
| 12:12.05 | fishboy1669 | any one here from uk? |
| 12:12.11 | fishboy1669 | uk anyone? |
| 12:12.16 | *** join/#asterisk heka (~fasada@82.114.68.126) |
| 12:13.42 | jas_williams | fishboy1669: Yes |
| 12:14.16 | Astinus_ | today i called a company which uses some sort of pbx .. "press 1 for blah press 2 for bleh" i think they had 3 menu levels. i pressed a key many times fast, and the suddenly three diffrent voicefiles started playing at the same time. |
| 12:23.50 | *** join/#asterisk montoya (~montoya@200.195.80.47) |
| 12:25.14 | darkskiez | file, yes |
| 12:25.18 | darkskiez | fishboy1669, yes |
| 12:25.25 | fishboy1669 | hi dark |
| 12:25.27 | fishboy1669 | hi jas |
| 12:25.34 | fishboy1669 | do either of u use x100p cards? |
| 12:25.44 | RoyK | > |
| 12:25.46 | darkskiez | i dont |
| 12:25.47 | RoyK | < |
| 12:26.09 | fishboy1669 | jas do u use x100p? |
| 12:26.23 | fishboy1669 | hi royk |
| 12:26.27 | fishboy1669 | hows things |
| 12:26.57 | RoyK | trying to make this farfon thingie work |
| 12:26.59 | RoyK | bugs |
| 12:27.41 | Essobi | jeeez |
| 12:28.08 | Essobi | mornin |
| 12:28.23 | fishboy1669 | whats farfon |
| 12:28.23 | fishboy1669 | ? |
| 12:28.56 | fishboy1669 | http://www.firebox.com/index.html?dir=firebox&action=product&pid=1025 |
| 12:29.02 | fishboy1669 | unrelated but looks cool! |
| 12:29.41 | fishboy1669 | aha farfon iax device |
| 12:30.04 | fishboy1669 | jas u there |
| 12:30.14 | fishboy1669 | jas do u use x100p? |
| 12:30.15 | Essobi | I'm tired. |
| 12:30.28 | fishboy1669 | same here didnt get home till 10:20 last night |
| 12:30.30 | fishboy1669 | bloody work |
| 12:30.38 | fishboy1669 | then up early this morn to take car to garrage |
| 12:30.40 | fishboy1669 | befor work |
| 12:31.26 | Essobi | I wonder if that's one of those x-ten/lite supported phones. |
| 12:32.03 | Essobi | PSssh.. I wen to bed at 2:35AM waiting for billing to batch out. |
| 12:32.15 | Essobi | Here it is 7:30AM not. :| |
| 12:32.19 | RoyK | brr. -19 degrees this morning |
| 12:32.24 | Essobi | BAAH! |
| 12:32.41 | Essobi | Everyone move to Texas, |
| 12:33.12 | RoyK | been there once. that was enough |
| 12:33.15 | fishboy1669 | where r u roy |
| 12:33.20 | fishboy1669 | -19 is cold |
| 12:33.23 | fishboy1669 | is it alps |
| 12:33.23 | RoyK | .no |
| 12:33.27 | fishboy1669 | is there snowboarding there |
| 12:33.30 | RoyK | oslo |
| 12:33.42 | fishboy1669 | aha cool |
| 12:33.45 | fishboy1669 | lol |
| 12:33.48 | RoyK | exactly |
| 12:34.00 | fishboy1669 | is there any snowboarding there |
| 12:34.01 | RoyK | only -10 now. bright sun... |
| 12:34.04 | RoyK | yes |
| 12:34.05 | trym | RoyK: -19? det er drøyt |
| 12:34.05 | RoyK | lots |
| 12:34.12 | RoyK | trym: i dag morrest |
| 12:34.17 | fishboy1669 | always fancied visiting scandinavia |
| 12:34.18 | trym | var kaldt å ta røyk in att ja |
| 12:34.20 | ta[i]nted | what are valid file_formats in asterisk? |
| 12:34.24 | ta[i]nted | wav, gsm? |
| 12:34.28 | RoyK | trym: hehe |
| 12:34.31 | RoyK | trym: hvor er du fra? |
| 12:34.38 | trym | RoyK: bor på stabekk |
| 12:34.42 | fishboy1669 | essobi which phone u refering to |
| 12:35.05 | Essobi | the skype one you linked to |
| 12:35.06 | fishboy1669 | trym u from scandinavia as well! |
| 12:35.12 | trym | fishboy1669: of course |
| 12:35.15 | trym | its the only decent place to live |
| 12:35.16 | fishboy1669 | aha no i think its just skype |
| 12:35.21 | trym | ;) |
| 12:35.24 | fishboy1669 | it uses different system to the rest of us |
| 12:35.27 | trym | except during the winter |
| 12:35.29 | fishboy1669 | lol |
| 12:36.07 | *** join/#asterisk eye69 (eye69@nattmirren.com) |
| 12:36.33 | eye69 | Hello. What's a good softphone for Linux? |
| 12:36.51 | eye69 | I just need something so that my mate can get an extension on my PBX. |
| 12:39.32 | *** join/#asterisk mildenhall (~mildenhal@194.114-84-212.ippool.ndo.com) |
| 12:39.42 | Essobi | http://store.yahoo.com/cuphone/ipphone2.html Hmm... I think that might be SIP/SKYPE |
| 12:43.13 | mildenhall | When calls are transferred using a SIP phone, my pc endpoint does not see the change in where it is talking to. It still thinks it is chatting to the person that transfered the call, and not the persion who was actually transferred. Is there anything in Asterisk I've missed perhaps? It also could be my software endpoint, but I'm njust wondering. :-) |
| 12:45.27 | eye69 | Essobi: Was that for me? |
| 12:46.16 | teemu-x | if someone knows how to read traces of dropping h.323 calls, please take a look at http://pastebin.ca/6716 |
| 12:46.36 | Essobi | Is those how we ask? |
| 12:46.43 | Essobi | IS that how we ask? |
| 12:48.19 | zigman | mantis mantis mantis.. i want my mantis pass !!!! |
| 12:48.34 | Essobi | teemu-x What's the problem? |
| 12:48.37 | Essobi | Eerr |
| 12:48.43 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 12:48.47 | Essobi | What's it doing/no doing |
| 12:48.50 | teemu-x | call drops right after receiving end picks up the phone |
| 12:48.58 | Essobi | and make it fast.. Ig ot to get in the show and head to work |
| 12:49.03 | Essobi | Hmm. |
| 12:49.26 | Essobi | Go enable faststart and try it again, and I'll keep reading in the mean time. |
| 12:49.36 | Essobi | I see it answering and clicking |
| 12:49.36 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 12:49.40 | Essobi | and playback starding |
| 12:49.41 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 12:50.47 | teemu-x | yeah, but both ends (or one end in this case) just hear beep beep beep beep |
| 12:52.08 | Essobi | <PROTECTED> |
| 12:52.20 | Essobi | It's just hanging up cause it's "supposed" to. |
| 12:52.25 | Essobi | o_O |
| 12:52.49 | Essobi | Umm.. Make sure your codecs match.. just turn on G711 on both sides first. |
| 12:52.53 | teemu-x | that i can understand, but IMO there shouldnt be any RELEASE COMPLETE |
| 12:52.58 | Essobi | or somethin equivilent there. |
| 12:53.13 | teemu-x | I've got disallow=all, allow=gsm |
| 12:53.36 | *** part/#asterisk mildenhall (~mildenhal@194.114-84-212.ippool.ndo.com) |
| 12:53.44 | Essobi | <PROTECTED> |
| 12:53.54 | Essobi | teemu-x Umm. that could be the problem too. |
| 12:54.14 | Essobi | your gateway you're connecting this to is sending you every CODEC in god creation. |
| 12:55.08 | Essobi | And our GSM might not with iwth their GSM |
| 12:55.19 | Essobi | they have 4 GSMs listed in the negotiation |
| 12:55.21 | Essobi | ok |
| 12:55.25 | Essobi | I'm out peeps. |
| 13:00.52 | teemu-x | \o/ it works! \o/ |
| 13:01.24 | teemu-x | I have no idea why, but right now I dont care. |
| 13:01.53 | *** join/#asterisk Newbie___ (some@218.111.221.110) |
| 13:08.14 | *** join/#asterisk CarlosMP_ (~CarlosMP@64.40.137.60) |
| 13:08.16 | ta[i]nted | anyone? |
| 13:08.32 | ta[i]nted | what file_formats does asterisk support? only gsm and wav? |
| 13:11.09 | phaded | err.. what happened to slashdot |
| 13:12.00 | LarsAC | for voiceboxes you mean? |
| 13:12.04 | phaded | now it's working.. |
| 13:17.58 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 13:19.23 | hajekd | grandstream can't dial *5xxxx ? |
| 13:20.16 | *** join/#asterisk donis (donis@office.unique.lt) |
| 13:24.00 | *** join/#asterisk A-Married-Male (~MostWante@203.128.26.22) |
| 13:24.22 | jas_williams | hajekd: Do you see any error in the CLI |
| 13:25.00 | A-Married-Male | H5 |
| 13:25.01 | *** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net) |
| 13:26.17 | Zeeek | heh |
| 13:26.29 | A-Married-Male | Can some help me with asterisk... |
| 13:28.51 | Zeeek | of course |
| 13:29.08 | Zeeek | now that you have announced your life preferences |
| 13:31.27 | Zeeek | ho ho ho |
| 13:32.34 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:39.48 | hajekd | jas_williams: no |
| 13:40.35 | Zeeek | do so, do so |
| 13:40.49 | jas_williams | hajekd: Try a sip debug peer IPADDRESS where IPADDRESS=Grandstream and check for errors |
| 13:40.57 | RoyK | ~seen wasim |
| 13:41.06 | jbot | wasim <~wasim@203.81.213.118> was last seen on IRC in channel #asterisk, 2d 5m 49s ago, saying: 'yay! fresh feta cheese!'. |
| 13:41.06 | hajekd | funny now works...I have reload asterisk config |
| 13:41.11 | hajekd | thx |
| 13:44.34 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
| 13:52.10 | bjohnson | free shipping from http://www.eezeephone.com on iax phones for limited time. See the asterisk-biz mailing list archives for info |
| 13:53.33 | *** join/#asterisk DHuang (~DHuang@203.46.67.60) |
| 13:55.48 | DHuang | I got problem connecting 2 x SIP phone to Asterisk, A can dial to B, but B can not dial to A.. error msg = failed to Authorize user |
| 13:57.49 | DHuang | I used to get -- SIP Seeding peers from Astdb: '17004398480610801' at 17004398480610801@192.168.0.1:5060 for 180 and now got -- SIP Seeding peers from Astdb: '17004398480610801' at 17004398480610801@203.46.67.1:5060 for 180 |
| 13:57.53 | tzanger | morning |
| 13:58.02 | DHuang | morning tz |
| 13:58.22 | DHuang | is it something to do with nat settings? |
| 13:58.45 | RestLessGemini | put insecure=yes or insecure=very in sip.conf and see if this resolve the problem |
| 13:59.01 | DHuang | RLG: Thanks.. trying now |
| 13:59.15 | RestLessGemini | also check if your phone is behind nat |
| 13:59.26 | RestLessGemini | if yes., then put nat=1 or nat=yes |
| 14:00.38 | RoyK | RestLessGemini: insecure=very should never really be used |
| 14:00.55 | RoyK | RestLessGemini: it turns off all authentication.... |
| 14:03.49 | ta[i]nted | do u guys get issues converting gsm to wav? |
| 14:04.02 | DHuang | ta: no |
| 14:04.18 | ta[i]nted | i get some internal inconsistency errors |
| 14:04.33 | ta[i]nted | DHuang what version sox are u running? |
| 14:05.27 | DHuang | ta:sox-12.17.4-4.fc2 |
| 14:06.02 | ta[i]nted | strange |
| 14:06.13 | ta[i]nted | can i send u a gsm file to try to encode into wav? |
| 14:06.25 | DHuang | ok |
| 14:08.04 | DHuang | ta: can you email to david@huang.net.au |
| 14:08.17 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 14:08.23 | Alexi1 | hello |
| 14:09.09 | *** join/#asterisk mixi (~mixi@pD9E59B48.dip.t-dialin.net) |
| 14:09.48 | RestLessGemini | Thanks for update RoyK :) |
| 14:10.24 | ta[i]nted | DHuang okay sent! |
| 14:10.45 | DHuang | thanks.. |
| 14:10.50 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
| 14:11.17 | DHuang | RLG: only work when put insecure = very :-( |
| 14:13.29 | ta[i]nted | DHuang any luck? |
| 14:13.51 | *** join/#asterisk strace (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 14:13.54 | strace | hi |
| 14:13.57 | DHuang | RLG: when making the call... both phone have the same xxxx@192.168.0.l1 |
| 14:14.03 | strace | where can I see how to make an isdn cable (crossover) |
| 14:14.21 | florz | strace: look for the pbx4linux website |
| 14:14.30 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
| 14:14.41 | strace | k |
| 14:14.43 | strace | thanks |
| 14:15.31 | DHuang | ta: trying now |
| 14:15.48 | mixi | have i already mentioned? |
| 14:16.01 | mixi | asterisk rocks ;-) |
| 14:17.29 | strace | :) |
| 14:17.40 | shido6 | yes it does |
| 14:17.54 | RoyK | except all the bugs and the badly written code..... |
| 14:18.15 | ta[i]nted | RoyK and the bad documentation |
| 14:18.47 | RoyK | the docs on the wiki and asteriskdocs.org aren't that bad |
| 14:18.54 | strace | the wiki r0x |
| 14:19.14 | RestLessGemini | well, its open source, what do you expect? :) |
| 14:19.50 | RoyK | RestLessGemini: just wait :) there's a fork() creeping up |
| 14:19.54 | DHuang | ta: sox: help ! internal inconsistency - data_written 53236 gsmbytecount 53235 ... very bad recording... but the wav file works |
| 14:20.04 | RestLessGemini | ok guys, time to leave now.. next shift.. next office.. :'( |
| 14:20.26 | RestLessGemini | take care, HAppy *ing... bye |
| 14:20.30 | DHuang | RLG: later |
| 14:22.30 | ta[i]nted | hmm |
| 14:22.44 | ta[i]nted | anyone else getting inaccurate call disposition? |
| 14:22.56 | ta[i]nted | sometimes a call goes through and registers as NORESPONSE |
| 14:24.29 | ta[i]nted | i mean the call is ANSWERED, but is registered as NORESPONSE |
| 14:24.35 | *** join/#asterisk MiXi^ (~mixi@pD954532D.dip.t-dialin.net) |
| 14:24.39 | tzanger | ta[i]nted: hmm |
| 14:24.40 | ACiDV | Since yesterday, when the TE405 driver is loaded (wct4xxp) I cannot receive audio from asterisk, that all span alarm are OK but all led are off (and no cable connected), that zttest dont return any result, it's a sign that the card is crashed ? |
| 14:25.04 | tzanger | ACiDV: yes that sounds like the card is not responding |
| 14:25.09 | tzanger | I've never seen that before, personally |
| 14:25.24 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 14:26.33 | ACiDV | very weird, have try a lot of debug... and the conclusion is that when wct4xxp is loaded, asterisk stop return data (ex. dont lisen any playback sound) |
| 14:26.53 | tzanger | is the card generating interrupts properly? |
| 14:27.22 | ACiDV | tzanger in /proc/interrupts, it increase of ~1000/sec |
| 14:27.38 | tzanger | k |
| 14:28.12 | RoyK | ACiDV: the interrupt rate is ok. that's just because of the timer on the card |
| 14:28.19 | ACiDV | all my alarm are OK and no cable connected, modprobe (+dmesg) say that the card is detected, ztcfg show all my span |
| 14:28.32 | *** join/#asterisk dsmouse (~mouse@rrcs-24-199-146-243.midsouth.biz.rr.com) |
| 14:28.34 | ACiDV | and all working 2 day ago |
| 14:28.58 | tzanger | ACiDV: odd |
| 14:29.04 | ACiDV | Not related to * or zaptel version, I have try with 1.0.6 and latest cvs-head |
| 14:29.09 | tzanger | ACiDV: I'd call digium and get some support for that $1500 card |
| 14:29.15 | tzanger | Mine's bene working great for the past year |
| 14:29.26 | tzanger | which kernel and distro? |
| 14:29.39 | RoyK | tzanger: they'll prolly charge him $100 per hour for support :) |
| 14:29.41 | ACiDV | tzanger 2.6.10 / Fedora Core 2 |
| 14:29.51 | tzanger | RoyK: he's got paid support with the card |
| 14:30.01 | RoyK | ~fedora? |
| 14:30.02 | jbot | i guess fedora is RedHat's alpha/beta distro made for testing out stuff to be put into RedHat later. |
| 14:30.04 | tzanger | what's changed from 2 days ago to now |
| 14:30.22 | ACiDV | tzanger absolutly nothing :| |
| 14:30.32 | tzanger | ACiDV: I find that amazingly hard to believe |
| 14:30.45 | DHuang | RoyK: if I put insecure=very... both phone give the same callerid ?? any idea? |
| 14:30.56 | RoyK | don't use insecure=very |
| 14:31.08 | RoyK | fix the authentication problems instead |
| 14:31.08 | ta[i]nted | DHuang are both phones assigned the same IP? |
| 14:31.20 | Essobi | sometimes insecure=yes is inevitable. |
| 14:31.25 | DHuang | RoyK: insecure = yes failed auth |
| 14:31.35 | RoyK | DHuang: as I said - fix the auth |
| 14:31.43 | DHuang | ta: no.. thay are differenct IP.... private IP ie. |
| 14:32.08 | DHuang | RoyK any idea on how to fix the authentication? |
| 14:32.24 | RoyK | ~rtfm |
| 14:32.25 | jbot | extra, extra, read all about it, rtfm is read the f*cking manual... try asking me about "FAQ" |
| 14:32.28 | RoyK | ~docs |
| 14:32.29 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:32.33 | ta[i]nted | RoyK stfu |
| 14:32.35 | DHuang | RoyK: both phone are registered... |
| 14:32.43 | dsmouse | ~rtfw |
| 14:32.44 | jbot | from memory, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 14:32.49 | RoyK | ~rtfs |
| 14:32.50 | jbot | i heard rtfs is probably read the f*cking source... |
| 14:33.00 | ta[i]nted | DHuang paste your sip.conf to pastebin.ca |
| 14:33.01 | RoyK | ~lart ta[i]nted |
| 14:33.20 | Essobi | RTFS seems to be the only way I can find out most things. :) |
| 14:33.21 | Zeeek | ~watp RoyK |
| 14:33.34 | DHuang | Hmm.... no sip.conf everything is from MySQL... ie. only contain the peers info |
| 14:33.50 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 14:34.18 | Zeeek | death is like sex except you're not nauseous afterward |
| 14:34.19 | Essobi | heh |
| 14:34.33 | Zeeek | hah |
| 14:34.35 | Zeeek | whoh |
| 14:34.38 | ta[i]nted | DHuang try to manually create sip.conf then.. see if u can isolate problem |
| 14:34.54 | Essobi | ~seen Oprah's ass, but I hear it's fine. |
| 14:34.56 | jbot | i haven't seen 'oprah's ass, but i hear it's fine.', Essobi |
| 14:35.02 | Essobi | MEhe. |
| 14:35.22 | DHuang | ta: ok.. shall try now.. |
| 14:35.51 | shido6 | Burger King, McDonalds, or burn myself in the kitchen? |
| 14:36.07 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 14:36.13 | RoyK | oej: morgon |
| 14:36.14 | tzanger | shido6: burger king |
| 14:36.19 | tzanger | I wish there was one in Listowel |
| 14:36.26 | RoyK | shido6: it takes 10 minutes to make some pasta |
| 14:36.27 | shido6 | burger king it is... |
| 14:36.29 | RoyK | I'd do that.... |
| 14:36.30 | shido6 | be back in a minute... |
| 14:36.32 | tzanger | I like their boiled meat that's then charred slightly to make it look like it was bbq'd |
| 14:36.36 | Essobi | NO SMOKING (unless you're on fire) |
| 14:36.36 | shido6 | Im too hungry to cook |
| 14:36.45 | shido6 | but not too hungry to clean off the car |
| 14:36.49 | tzanger | hahaha |
| 14:37.06 | shido6 | hop in behind some horses and drive through some lights to get to a drive through that will certainly screw up my order |
| 14:37.18 | shido6 | sorry - meant to say drive thru |
| 14:37.23 | tzanger | I like the whoppers, they taste a lot better htan mcd |
| 14:37.29 | tzanger | I'm addicted to A&W chicken burgers |
| 14:37.32 | tzanger | they're really good |
| 14:38.16 | *** join/#asterisk phantam (~phantam@72.252.15.235) |
| 14:38.18 | phantam | hey guys |
| 14:38.21 | shido6 | a&w is damn good |
| 14:38.24 | phantam | everybudy still around |
| 14:38.34 | Essobi | no we are not |
| 14:38.35 | tzanger | expensive though |
| 14:38.36 | phantam | hehe |
| 14:38.47 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 14:38.51 | phantam | was wondering how to setup oh323 to talk to a gateway instead of a gatekeeper |
| 14:38.54 | phantam | or it the same way |
| 14:39.14 | pcm | gatekeeper redirects |
| 14:39.19 | pcm | it should be pretty much similar |
| 14:39.47 | *** join/#asterisk Casper_UA (~casper@as-2-22.ar43-2x.kharkov.ukrtel.net) |
| 14:40.09 | Casper_UA | hi |
| 14:40.42 | *** join/#asterisk MiXi^_ (~mixi@pD9EE1E47.dip.t-dialin.net) |
| 14:40.45 | pcm | hi |
| 14:41.32 | Essobi | Anyone pushing URLs to the desktop from *? |
| 14:42.38 | *** join/#asterisk brc-tux (~brc-tux@221.2-host.augustakom.net) |
| 14:42.52 | *** part/#asterisk DHuang (~DHuang@203.46.67.60) |
| 14:42.57 | *** join/#asterisk DHuang (~DHuang@203.46.67.60) |
| 14:43.19 | Essobi | Psh. |
| 14:43.33 | DHuang | ta: ok.. found the probelm.... the extconf.conf module has bug....ie. the sippeers part |
| 14:43.44 | *** join/#asterisk nazgool (~nazgoool@gatekeeper-e0.twc.de) |
| 14:43.45 | nazgool | hi |
| 14:44.00 | *** part/#asterisk pashah (~pashah@relay.patentica.com) |
| 14:46.05 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 14:46.10 | jalsot | does anybody know what is the pricing for Farfon devices? |
| 14:46.47 | phantam | ok |
| 14:46.58 | phantam | i need to connect asterisk to a cisco 3660 |
| 14:47.12 | nazgool | i have an agi script that returns a dialstring (such as "SIP/joe&SIP/jack&SIP/tom" ) in an asterisk variable, and then a line of extensions.conf dials out to that dialstring. is there |
| 14:47.34 | ta[i]nted | damn variable DIALSTATUS keeps returning mixed results |
| 14:49.14 | ta[i]nted | what's the best way to figure out a placed call's status using AGI |
| 14:49.21 | bjohnson | getting "Got SIP response 302 "Moved Temporarily" back from 192.168.2.6" back from a SPA 3000 fxs again .. anyone know how to fix that? |
| 14:49.34 | ta[i]nted | should i lookup the CDR or try DIALSTATUS variable.. |
| 14:52.01 | *** join/#asterisk clive-- (~pirch@myw-stp-66-18-85-146.sentechsa.net) |
| 14:53.57 | *** join/#asterisk JerJer (~jj@feth100-fw.fament.net) |
| 14:54.06 | shido6 | bjohnson |
| 14:54.08 | shido6 | I hate that |
| 14:54.16 | shido6 | see if you have to extensions with the same name |
| 14:54.17 | shido6 | ext 1 |
| 14:54.19 | shido6 | and ext 2 |
| 14:54.35 | shido6 | ta[i]nted ? |
| 14:54.39 | phantam | argggg |
| 14:54.41 | *** join/#asterisk DARP (~diegoramo@200.71.33.201) |
| 14:54.47 | DARP | Hi everyone |
| 14:54.50 | phantam | dont they make an iax2 for the cisco3660 |
| 14:54.54 | DARP | I need Help |
| 14:54.57 | shido6 | LOL |
| 14:55.04 | shido6 | sip u can do with that cisco |
| 14:55.53 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 14:55.53 | cjk | hi |
| 14:55.54 | phantam | well |
| 14:55.54 | JerJer[mobile] | hoe |
| 14:55.54 | DARP | i have that error --> Ouch ... error while writing audio data: : Broken pipe |
| 14:55.56 | phantam | im trying to get asterisk to talk oh323 to the cisco |
| 14:56.04 | DARP | when i try to start the asterisk |
| 14:56.05 | phantam | but cant figure out how to setup the oh323 to talk to a gateway |
| 14:56.12 | shido6 | not gonna happen very easily |
| 14:56.13 | DARP | i don't know why |
| 14:56.14 | Essobi | phantam Do you self a favor and use SIP |
| 14:56.17 | shido6 | got a sip load on the cisco? |
| 14:56.26 | Essobi | it |
| 14:56.34 | Essobi | it's retarded easy compared to oh323 |
| 14:57.04 | cjk | is there a way to outsource musiconhold to a different (streaming) server |
| 14:57.12 | phantam | ?hmmm |
| 14:57.13 | Essobi | sure |
| 14:57.15 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
| 14:57.21 | phantam | but the other side wants to use h323 |
| 14:57.22 | Essobi | why couldn't you? |
| 14:57.32 | Essobi | phantam GET a sip image on the router. |
| 14:57.35 | *** part/#asterisk DHuang (~DHuang@203.46.67.60) |
| 14:57.46 | Essobi | h323 is not stable, nor practical with *. |
| 14:57.50 | phantam | lol |
| 14:57.55 | phantam | h323 is what hes using in the US |
| 14:57.56 | lyroy | Does someone why I always receive that message : WARNING[24131]: chan_sip.c:752 retrans_pkt: Maximum retries exceeded on call. I have a Cisco ATA 186?? |
| 14:58.02 | phantam | and the switches are currently active |
| 14:58.07 | Essobi | I went round and round with it for 4 months. |
| 14:58.11 | *** join/#asterisk DevilFish (~me@staff211.qtm.net) |
| 14:58.19 | jalsot | I have a compilation problem with CVS-HEAD, could anybody help? http://pastebin.ca/6718 |
| 14:58.26 | *** join/#asterisk mesi (~player@dsl-082-083-129-227.arcor-ip.net) |
| 14:58.27 | phantam | Essobi but how can u get SIP to transfer a buncha lines between the cisco and asterisk |
| 14:58.28 | Essobi | It oh323 crashed a few of my playforms. |
| 14:58.29 | *** part/#asterisk mesi (~player@dsl-082-083-129-227.arcor-ip.net) |
| 14:58.32 | *** join/#asterisk mesi (~player@dsl-082-083-129-227.arcor-ip.net) |
| 14:58.39 | DevilFish | Does anyone know what happened to to the ChanSpy application? |
| 14:58.40 | Essobi | Including my Cisco routers and other softswitched. |
| 14:58.42 | JerJer[mobile] | then run a real channel driver |
| 14:58.44 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 14:58.49 | phantam | i wanted to do iax ... but the dumb 3660's dont look like they iax2 |
| 14:59.00 | nazgool | a way to know inside a following script which of the three people took up the phone? |
| 14:59.00 | JerJer[mobile] | well at least not pubicaly :P |
| 14:59.02 | Essobi | JerJer No, shit. thats what I'm trying to convince him of. |
| 14:59.32 | phantam | lol |
| 14:59.39 | pUmkInhEd | ~seen pumkinhed |
| 14:59.41 | jbot | pumkinhed is currently on #asterisk-doc #asterisk. Has said a total of 1 messages. Is idling for 2s |
| 14:59.41 | phantam | ill gladly try something else |
| 14:59.49 | phantam | but someone needs to tell me how cause im totally lost |
| 14:59.51 | ACiDV | ~itsp |
| 14:59.52 | jbot | hmm... itsp is Internet Telephony Service Provider. An ITSP is a "VoIP Phone Company" |
| 14:59.56 | phantam | cause the boxes are setup for h323 |
| 15:00.06 | Essobi | phantam SIP will re-invite as long as you don't have the T/t option on the dial command. |
| 15:00.17 | phantam | essobi got a sec for a pm? |
| 15:00.22 | Essobi | I'm pretty sure that's how it works. |
| 15:00.27 | Essobi | :P |
| 15:00.41 | Essobi | I guess. |
| 15:00.43 | phantam | lol |
| 15:01.54 | phantam | JerJer[mobile]: does oh323 do gateways? |
| 15:01.58 | phantam | or just gatekeepers |
| 15:02.13 | Essobi | WTF uses sendImage and send URL? |
| 15:02.35 | pcm | essobi: your future app |
| 15:02.41 | Essobi | ;P |
| 15:03.31 | JerJer[mobile] | phantam: i don't do oh323 |
| 15:03.37 | phantam | uh |
| 15:03.52 | phantam | the problem is we have the cisco's running h323 routed already throughout the US in LA |
| 15:04.06 | Essobi | As jerjer would say.. |
| 15:04.09 | Essobi | Sucks to be you. |
| 15:04.14 | phantam | but we need to connect asterisk to the 3660 to allow us to call from here to the US |
| 15:04.14 | *** join/#asterisk pashah (~pashah@relay.patentica.com) |
| 15:04.14 | Essobi | Tehe. |
| 15:04.15 | JerJer[mobile] | word |
| 15:04.17 | phantam | via the cisco box |
| 15:04.20 | JerJer[mobile] | SIP |
| 15:04.25 | Essobi | SIP! |
| 15:04.31 | phantam | but the boxes arent sip proxys |
| 15:04.35 | phantam | and they have no sip clients on them |
| 15:04.36 | boch | phantam yes |
| 15:04.37 | phantam | there all h323 |
| 15:04.52 | JerJer[mobile] | flash them with a SIP IOS load |
| 15:04.54 | phantam | and cisco hasnt released the sip.h323 chan till april or somesit |
| 15:04.55 | JerJer[mobile] | cisco does make them |
| 15:05.06 | JerJer[mobile] | um no |
| 15:05.09 | Essobi | no |
| 15:05.19 | JerJer[mobile] | we've had SIP loads for quite a while |
| 15:05.19 | Essobi | I've got routers running dual SIP and H323. |
| 15:05.38 | Essobi | Infact. |
| 15:06.39 | *** join/#asterisk MikeJ[Jayden] (~ircatjerr@65.170.43.34) |
| 15:07.09 | Essobi | Well shit. |
| 15:07.28 | Essobi | I need to figure out how to push a URL to a desktop.. Then on to my next pony trick. :) |
| 15:08.02 | Essobi | I got a big LED scroll bar I'm going to tie up to a set of programs |
| 15:08.09 | Essobi | and call from my dial plan. :) |
| 15:09.48 | HitTop | i wonder if there's greeting feature for voicemail? |
| 15:09.51 | ariel_ | Good morning folks. |
| 15:10.11 | HitTop | good morning ariel |
| 15:10.23 | jas_williams | HitTop: Press 0 in your voicemail box and you can record your own greetings |
| 15:10.49 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:11.02 | *** join/#asterisk felipex (~dsfdsf@host162-91.pool8533.interbusiness.it) |
| 15:11.07 | felipex | hi at all |
| 15:11.23 | eKo1 | Anyone have problems with channels not hanging up? |
| 15:11.27 | felipex | is possible to pump the volume up of zaptel device? |
| 15:11.35 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 15:11.37 | felipex | rxgain and txgain? |
| 15:11.48 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 15:11.52 | eKo1 | felipex: yeah |
| 15:11.59 | JerJer[mobile] | pump up the volume, dance dane |
| 15:12.01 | JerJer[mobile] | +c |
| 15:12.39 | felipex | eKo1 i have to use rxgain and txgain= |
| 15:12.41 | felipex | ? |
| 15:12.48 | Essobi | >->0 |
| 15:12.51 | Essobi | >-<0 |
| 15:13.02 | Essobi | JerJer Weeee. Happy? |
| 15:13.07 | eKo1 | felipex: Basically. |
| 15:13.09 | felipex | Essobi so normal is 0 |
| 15:13.29 | phantam | hehe the oh323 guy emailed me an hour ago and i didnt notice hope hes there maybe he can help |
| 15:13.35 | Essobi | What what? |
| 15:14.13 | hajekd | I'm getting every sec a notice: pbx_extension_helper: Cannot find extension context 'default' |
| 15:14.36 | Essobi | you've got something defaulting to land in [default] |
| 15:14.37 | hajekd | not every sec, but every 20-30 sec.. |
| 15:14.42 | *** join/#asterisk pUmkInhEd (~nospam@s142-179-184-59.ab.hsia.telus.net) |
| 15:14.45 | jalsot | does CVS-HEAD of asterisk compile or I'm alone with this problem? http://pastebin.ca/6718 |
| 15:14.59 | Essobi | make clean first jal? |
| 15:15.18 | *** join/#asterisk sangee (ravi@209.250.129.135) |
| 15:15.36 | jalsot | Essobi: I did |
| 15:15.43 | jas_williams | Mine makes fine CVS-HEAD-03/02/05-14:52:09 |
| 15:16.13 | MikeJ[Jayden] | i did a make from head fine yesterday |
| 15:16.56 | jalsot | ok, trying again... |
| 15:17.16 | sangee | is there any MIRC for SER? |
| 15:17.24 | *** join/#asterisk harryvv (~none@S010600055d210201.vs.shawcable.net) |
| 15:20.02 | RoyK | sangee: mirc? |
| 15:20.08 | RoyK | sangee: IRC channel? |
| 15:20.10 | loud | mirc for ser ? you mean irc chat room ? |
| 15:20.19 | jalsot | hmm, I'm getting the same compilation problem :( |
| 15:20.22 | RoyK | #SER is a good start |
| 15:20.23 | loud | try #ser |
| 15:20.31 | loud | tack |
| 15:20.32 | jalsot | ../include/asterisk/channel.h:214: error: field `varshead' has incomplete type |
| 15:21.22 | jalsot | gcc-3.3.4 |
| 15:21.25 | sangee | Hi RoyK, i am new to this, how do i get into IRC |
| 15:21.53 | sangee | #Ser |
| 15:22.01 | *** part/#asterisk clive-- (~pirch@myw-stp-66-18-85-146.sentechsa.net) |
| 15:22.37 | loud | RoyK: micket ett huvud.. |
| 15:23.00 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 15:25.25 | jas_williams | sangee: try /join #Ser |
| 15:25.45 | eKo1 | sangee: You should rtfm on irc. |
| 15:25.59 | Essobi | pssh |
| 15:26.04 | Essobi | RTFS |
| 15:26.28 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
| 15:26.49 | jas_williams | gcc version 3.2.2 |
| 15:26.52 | *** join/#asterisk randu (~randy@pool-70-16-112-236.scr.east.verizon.net) |
| 15:27.22 | randu | Hello Everyone! Is there a tutorial anywhere on how to setup after hours greeting? |
| 15:27.34 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 15:27.40 | ManxPower | in #asterisk-stable |
| 15:28.09 | randu | was that for me MaxPower? |
| 15:28.19 | ManxPower | no |
| 15:28.20 | Essobi | I think that was a broken /join |
| 15:28.21 | Essobi | :) |
| 15:28.38 | randu | oh ok :-) |
| 15:28.43 | Essobi | 1.0.6 released? You bastards. |
| 15:28.57 | eKo1 | Long ago.... |
| 15:29.56 | Essobi | I was thinking that was .5 for some recent.. |
| 15:29.59 | Essobi | reason.. |
| 15:30.18 | randu | I was wondering if there was an example or tutorial to setup after hours for asterisk. basically if it is after hours user goes to voicemail when trying to get an extension? |
| 15:30.35 | *** join/#asterisk CarlosMP_ (~CarlosMP@64.40.132.113) |
| 15:31.22 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.19) |
| 15:31.36 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
| 15:31.49 | Essobi | it's easier to do the other way around. |
| 15:32.24 | Essobi | http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime |
| 15:33.36 | randu | so the logic would be if it is after hours immediately send to voicemail? |
| 15:33.40 | CarlosMP_ | has anyone integrated asterisk with a CRM package like goldmine or Microsoft CRM? |
| 15:37.08 | *** join/#asterisk doughecka (~dheckaman@doughecka.user) |
| 15:37.12 | eKo1 | Argh, why don't these fucking channels hang up. |
| 15:37.15 | MikeJ[Jayden] | Carlos: Integrated how? |
| 15:37.47 | CarlosMP_ | Mike" Have it create an activity, throw up a screen pop with the customers name, account number, etc. |
| 15:37.50 | harryvv | eKo1 memory leak? :) jk of course thats a mswin problem. |
| 15:38.27 | MikeJ[Jayden] | You want a dialer w/ scree pops? |
| 15:38.51 | CarlosMP_ | Yes, but for incoming calls |
| 15:39.07 | MikeJ[Jayden] | such as take the caller ID and screen pop info? |
| 15:39.07 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 15:39.40 | bjohnson | shido6: do you guys sell hardware? http://www.jecinc.on.ca/RFD/rfd-voip-A-1.html |
| 15:41.06 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 15:41.06 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 15:41.28 | MikeJ[Jayden] | carlos, look at http://www.yottadot.org/download.php?op=viewsdownload&sid=10 |
| 15:41.38 | *** join/#asterisk fugitivo (~ajf@201.255.106.249) |
| 15:41.38 | MikeJ[Jayden] | I assume you are talking windows correct? |
| 15:41.45 | *** join/#asterisk santiago (~santiago@63.245.86.105) |
| 15:41.50 | CarlosMP_ | Mike: Windows=yes |
| 15:42.04 | doughecka | windows = very yes |
| 15:42.05 | doughecka | :P |
| 15:42.31 | MikeJ[Jayden] | that link will give you some C# code that can trigger event on incoming calls, then you would need somone to write your bridge to your CRM |
| 15:43.21 | MikeJ[Jayden] | I don't know CRM well, but it would really just be a sql lookup off the caller ID, a decent C# coder should be able to kick somthing out like that pretty quick |
| 15:44.20 | *** join/#asterisk fugitivo (~ajf@201.255.106.249) |
| 15:45.30 | heka | anybody using realtime sip. I have configured and compiled it also publicated the user infos, but when I try to register a user it stalles for a while and then says registration timed out |
| 15:45.33 | heka | any ide? |
| 15:45.34 | heka | any idea? |
| 15:45.53 | *** join/#asterisk tzafrir (~tzafrir@62.90.10.53) |
| 15:46.18 | CarlosMP_ | Mike: I was hoping that someone else had already done this... |
| 15:46.29 | MikeJ[Jayden] | no such luck that I know of |
| 15:46.32 | *** join/#asterisk human39 (~human39@chewie.fyi.net) |
| 15:46.53 | MikeJ[Jayden] | basically all you want is a popup on call to a configured phone? |
| 15:46.54 | *** part/#asterisk human39 (~human39@chewie.fyi.net) |
| 15:46.57 | *** join/#asterisk jason^ (jason@acs-24-154-127-188.zoominternet.net) |
| 15:47.22 | *** part/#asterisk jason^ (jason@acs-24-154-127-188.zoominternet.net) |
| 15:50.57 | CarlosMP_ | Mike: looking for popup on the users computer screen. This may be really difficult because of Terminal Services. |
| 15:51.12 | CarlosMP_ | But, the popup would be from within the CRM package |
| 15:52.07 | MikeJ[Jayden] | Citrix or Terminal services? |
| 15:52.30 | CarlosMP_ | Both...have a couple of customers with each |
| 15:52.59 | MikeJ[Jayden] | and is the * box accessable from the terminal server? |
| 15:53.32 | MikeJ[Jayden] | or is the user PC accessable to both * and the db back end of the CRM? |
| 15:53.41 | CarlosMP_ | Don't have an * box yet, but when it's in it'll be there... |
| 15:53.51 | Darwin35 | why did I get pokedin the eye |
| 15:54.18 | CarlosMP_ | They can be accessible to the DB (SQL) of the CRM, whether it's MSCRM or Goldmine |
| 15:54.19 | MikeJ[Jayden] | carlos: not done but definately doable |
| 15:54.58 | CarlosMP_ | The thing is that at a price point, * may be a better answer than Vonexus PBX, but without the CRM integration, it makes it a bit tougher |
| 15:55.30 | mesi | Is there an mp3-Jukebox for Asterisk available, which offers functionality like skip song forward/backward, fast forward/backward, play, pause, stop, perhaps even bookmarks? |
| 15:55.32 | MikeJ[Jayden] | carlos: quick and easy, just set up a SP that you toss in the caller ID, and it spits out whatever you want, then use that code in windows call manager to grab the caller ID, hit the stored procedure, and display the results in a dialog |
| 15:56.13 | MikeJ[Jayden] | for the price point, * is better than what? |
| 15:57.06 | jalsot | hmm, I did make clean, make update, make and getting the same compilation problem |
| 15:57.28 | jalsot | but now I disabled cdr_sqlite compilation and it went through :) |
| 15:57.40 | CarlosMP_ | Mike: Vonexus PBX |
| 15:57.45 | jalsot | jas_williams: do you use cdr_sqlite? |
| 15:58.16 | jas_williams | jalsot: No I do not |
| 15:58.20 | MikeJ[Jayden] | dunno about that one at all... |
| 15:58.41 | MikeJ[Jayden] | well, I guess the answer is, if you don't have a good C# coder around, grab one and pay them |
| 15:58.43 | MikeJ[Jayden] | :) |
| 15:58.56 | jalsot | jas_williams: thanks |
| 15:59.04 | CarlosMP_ | Mike: That's what I'm going to end up doing...thanks |
| 15:59.43 | MikeJ[Jayden] | np, if you have problems, let me know.... I may be able to cobble somthing together for you, but my time is pretty tight at the moment |
| 16:00.36 | hajekd | I'm getting every 20 secs the notice about Mar 2 16:57:09 NOTICE[20163]: pbx.c:1319 pbx_extension_helper: Cannot find extension context 'default' there is no load on the asterisk... |
| 16:00.39 | CarlosMP_ | thanks for the offer..will let you know. I'm ordering a box, equipment to start playing with * to have a better feel for it. |
| 16:00.55 | *** join/#asterisk didz_ (didz_@200.218.192.52) |
| 16:00.59 | hajekd | Any ideas how to debug that? |
| 16:01.17 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 16:01.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:02.05 | bjohnson | oops .. I guess playing with timing to get CID on my home machine won't actually help since I don't get CID from Bell on that line :P |
| 16:03.26 | *** join/#asterisk [ro]nic3try (~iancu@81.181.199.39) |
| 16:04.52 | Juggie | karm, i think MOH is borked in 1.0.6 and cvs head. |
| 16:05.19 | Juggie | theres a new bug on mantis, and i just tested on 1.0.6 & cvs head-2/28/05 and can confirm |
| 16:05.25 | `Sauron | ls |
| 16:05.27 | *** join/#asterisk Goshen (~Goshen@c-67-172-238-57.client.comcast.net) |
| 16:06.05 | Goshen | Where can I find documentation on the iax.conf option.... QOS=lowdelay ? I have searched all over voip wiki |
| 16:06.21 | *** join/#asterisk jayk952 (~jayk@shell.602.org) |
| 16:07.05 | jayk952 | i'm trying to make outbound calls with asterisk but i keep getting the message: Mar 2 07:55:29 NOTICE[2774]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'default' |
| 16:07.13 | jayk952 | anybody know what i might be doing wrong? |
| 16:07.38 | Juggie | it told you, your extensions.conf has no default context |
| 16:07.48 | jas_williams | hajekd: Can you post the full error lines to pastebin so we can have a look at it |
| 16:07.50 | Goshen | something between [ ] brackets is a context |
| 16:07.50 | Juggie | you need to learn a little, go read the wiki |
| 16:08.13 | Goshen | it is looking for an extension inside [default] context |
| 16:08.26 | Goshen | inside that [default] context needs to be your dial statement |
| 16:08.27 | jayk952 | ok. |
| 16:08.49 | Goshen | is it a card, or are you dialing out over voip? |
| 16:08.58 | jayk952 | i have a x100p card. |
| 16:09.17 | HitTop | I wonder if there would be a feature in asterisk that when user 1 is on call, then if another person dial in, asterisk will put that person on a queue |
| 16:09.23 | Goshen | let me point you to a fantastic page for that |
| 16:09.39 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 16:09.43 | hajekd | jas_williams: hah, the issue was that i have not defined default context in sip.conf - which is by default default ;-) |
| 16:09.51 | dsmouse | HitTop: just put everyone in a queue, and make the queue ring instead of MOH? |
| 16:09.58 | bjohnson | could someone point me to cvs upgrade instructions? Also, bkw_ likes to tell everyone to use HEAD but how realistic is that? |
| 16:10.09 | Zeeek | not |
| 16:10.42 | `Sauron | shrug, I run HEAD at home |
| 16:10.43 | `Sauron | :p |
| 16:10.47 | HitTop | dsmouse: does that means to put everyone as an agent? |
| 16:10.58 | dsmouse | no |
| 16:11.21 | dsmouse | just "member => Sip/5101" or whatnot in the queue |
| 16:11.25 | eKo1 | bjohnson: just do make upgrade inside the * src dir. |
| 16:12.31 | HitTop | dsmouse: oic.. i'll read more info for queue asterisk call queues first~ thank you~ |
| 16:12.34 | *** join/#asterisk PBXtech (~upirc@wirelessdata-167-248.mycingular.net) |
| 16:13.07 | [ro]nic3try | Help i calling a number, but when i answer, asterisk crashes.. what to do ? |
| 16:13.22 | eKo1 | Dang it, I needs me more bandwidth. |
| 16:13.39 | eKo1 | err, I need more bandwidth. |
| 16:14.44 | MikeJ[Jayden] | do you have a defualt context in extensions.conf? |
| 16:15.03 | MikeJ[Jayden] | wow.. scrollback, that was about 20 min ago wasn't it |
| 16:16.36 | Zeeek | From: "Rationalistic G. Toffies" <deliza@inkanpur.com> |
| 16:16.43 | *** join/#asterisk xai (~pasta@user-0vvdb42.cable.mindspring.com) |
| 16:17.04 | Zeeek | who wrote the script that makes these names up? |
| 16:17.06 | jayk952 | MikeJ[Jayden]: i think i fixed that part now |
| 16:21.18 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 16:21.58 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
| 16:22.16 | [ro]nic3try | <PROTECTED> |
| 16:22.24 | *** join/#asterisk ACiDV (~joel@122-64-2.dr.cgocable.ca) |
| 16:22.42 | [ro]nic3try | how do I fix that ? |
| 16:24.03 | bjohnson | eKo1: make upgrade doesn't look like it does cvs commands |
| 16:25.33 | bjohnson | never mine .. it's make update and then make upgrade |
| 16:26.35 | *** join/#asterisk Simon-- (~sim@staff-nat.netnation.com) |
| 16:26.50 | *** join/#asterisk gpowers (~glenn@static-68-162-84-101.phil.east.verizon.net) |
| 16:27.07 | gpowers | Good Day! |
| 16:27.50 | [ro]nic3try | has anyone any ideea what to do ? |
| 16:28.05 | gpowers | ?? |
| 16:28.46 | [ro]nic3try | first -- Attempting native bridge of SIP/7757-6523 and SIP/192.96.182.48-478d |
| 16:28.53 | [ro]nic3try | then Ouch ... error while writing audio data: : Broken pipe |
| 16:29.10 | [ro]nic3try | and the last Segmentation fault :( |
| 16:30.15 | tzanger | Commiting chan_zap.c patch for 2bct |
| 16:31.37 | marshall | I'm having trouble with my first PRI install. The telco is telling me they aren't receiving any digits -- Executing Dial("IAX2/5741@5741/5", "Zap/g1/") in new stack |
| 16:31.51 | [ro]nic3try | gd by all |
| 16:32.02 | marshall | should the digits be showing after Zap/g1/ |
| 16:32.20 | loud | your telco does iax ? |
| 16:32.32 | marshall | no |
| 16:32.52 | marshall | Im using IAX from phone to asterisk |
| 16:33.04 | loud | which card, signalling ? |
| 16:33.12 | Juggie | marshall, yes, your digits should be showing up after /g1/ |
| 16:33.14 | marshall | T100 / national |
| 16:33.36 | marshall | but they can't seem to decide if the signalling is 5ess or national |
| 16:33.41 | marshall | different answer each time I call them |
| 16:33.46 | Juggie | regardless |
| 16:33.51 | Juggie | you arnt sending anything at the moment |
| 16:33.57 | Juggie | fix that first |
| 16:34.05 | marshall | is it dialplan? |
| 16:34.09 | Juggie | yes |
| 16:35.06 | marshall | [local] |
| 16:35.06 | marshall | ignorepat => 9 |
| 16:35.06 | marshall | exten => 9,1,Dial(Zap/g1/) |
| 16:35.06 | marshall | exten => 9,2,Congestion |
| 16:35.13 | marshall | simple as I can make it |
| 16:35.20 | marshall | I must have tried 10 variations |
| 16:35.30 | Juggie | and yet you failed |
| 16:35.35 | Juggie | because you arnt passing a number! |
| 16:35.36 | marshall | lol |
| 16:35.41 | marshall | thanks for pointing that out |
| 16:35.49 | Juggie | Dial(Zap/g1/5551212) |
| 16:35.50 | Juggie | perhaps? |
| 16:36.10 | Juggie | go read the wiki |
| 16:36.13 | Juggie | www.voip-info.org |
| 16:36.37 | jas_williams | marshall: try exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) |
| 16:39.06 | Juggie | moh is working again in cvs-head. |
| 16:39.34 | marshall | will do |
| 16:42.40 | *** join/#asterisk brettnem (~brettnem@208.54.232.29) |
| 16:42.44 | brettnem | hello all! |
| 16:42.46 | *** join/#asterisk mnet (~zed@zed.staff.eurowan.net) |
| 16:42.49 | mnet | hello |
| 16:43.01 | *** join/#asterisk phantam (~phantam@72.252.15.235) |
| 16:43.03 | HitTop | hi |
| 16:43.03 | phantam | guys |
| 16:43.10 | phantam | i got h323 to work |
| 16:43.12 | phantam | but have a weird error |
| 16:43.23 | phantam | not sure if i fat fingered something again |
| 16:43.26 | phantam | anyone using it? |
| 16:43.32 | phantam | i know h323 sucks yadayada yada |
| 16:43.37 | mnet | not me phantam :( |
| 16:43.39 | mnet | but i have a little question about "call waiting" feature, may i ask ? |
| 16:43.47 | brettnem | my is thunderbird so slow? :( |
| 16:43.53 | brettnem | my=why |
| 16:46.29 | phantam | amd lol |
| 16:46.33 | marshall | no go jas_williams |
| 16:47.16 | marshall | Executing Dial("IAX2/5741@5741/4", "Zap/g1/") in new stack |
| 16:47.16 | marshall | <PROTECTED> |
| 16:47.16 | marshall | <PROTECTED> |
| 16:47.16 | marshall | <PROTECTED> |
| 16:48.04 | jas_williams | Have you done an extensions reload |
| 16:49.45 | marshall | yes |
| 16:49.54 | marshall | Its not going anywhere now |
| 16:50.12 | marshall | Spawn extension (default, 96619243, 1) exited non-zero on 'IAX2/5741@5741/3' |
| 16:50.32 | *** join/#asterisk Tili (~Tili@202-133-65-150-dialup.sat.net.pk) |
| 16:50.35 | *** join/#asterisk c00w (~sean@cpc1-staf1-3-0-cust86.brhm.cable.ntl.com) |
| 16:50.39 | c00w | hello all. |
| 16:50.57 | c00w | has anyone had any dealings with te410p digium cards ? |
| 16:51.11 | tzanger | c00w: I use the 405P which is pretty much the identical card but 5v |
| 16:51.20 | c00w | well i have the card installed |
| 16:51.25 | c00w | and its running |
| 16:51.33 | c00w | i have the driver loaded |
| 16:51.36 | c00w | thats all good |
| 16:51.48 | c00w | but its setting the span as t1 24 chan |
| 16:51.54 | c00w | not e1 24 chans |
| 16:52.04 | c00w | and its failing to the load in asterisk caus i have more chans |
| 16:52.26 | c00w | is there a flag or something i'm missing or any reason why its showing as t1 and not and e1 |
| 16:52.56 | c00w | SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 16:52.56 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 16:53.03 | phantam | whenever i make the h323 call |
| 16:53.05 | phantam | it ends |
| 16:53.09 | phantam | and says reason 24 (Call ended with Q.931 cause) |
| 16:53.09 | c00w | in ztcfg -vv its showing the 2 spans up |
| 16:53.15 | jas_williams | c00w: JUmper setting ? |
| 16:53.23 | c00w | is there a jumper setting |
| 16:53.44 | *** join/#asterisk brettnem (~brettnem@208.54.232.29) |
| 16:53.46 | Juggie | c00w, are you geting all your channels in ztcfg -vvv? |
| 16:53.50 | brettnem | doh.. wireless hell |
| 16:53.53 | c00w | super jas your a star |
| 16:53.55 | c00w | give me a min to test |
| 16:55.29 | phantam | anyideas? |
| 16:58.20 | brettnem | anyone using OSP? |
| 17:00.26 | eKo1 | OSP <--- Is that a Cisco protocol? |
| 17:00.48 | PatrickDK | ospf? |
| 17:00.51 | loud | no |
| 17:01.20 | Zeeek | jas advanced version of OS-X? |
| 17:01.28 | loud | http://www.voip-info.org/wiki-OSP |
| 17:01.32 | ACiDV | ~osp |
| 17:01.33 | jbot | from memory, osp is http://www.linktionary.com/o/osp.html |
| 17:01.36 | phantam | hmmm |
| 17:01.37 | loud | voip peering . |
| 17:01.40 | phantam | im at a dead end again |
| 17:01.42 | mnet | how do i put somebody "on wait" manually for example when i say "hold on please, i'm looking for your file" ? |
| 17:02.18 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-241-60.dsl.scarlet.be) |
| 17:02.24 | Zeeek | what fone? |
| 17:02.25 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 17:02.47 | mnet | me ? |
| 17:02.53 | Zeeek | ya |
| 17:03.03 | Zeeek | there is a hold button on some phones |
| 17:03.18 | mnet | i was thinking of a "pbx" features :) |
| 17:03.19 | *** join/#asterisk PCadach (~paul@212.19.157.154) |
| 17:03.23 | Zeeek | otherwise, maybe a flash or parking |
| 17:03.27 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
| 17:03.39 | mnet | when you dial *50* for example it puts someone on hold |
| 17:03.41 | Zeeek | parking |
| 17:03.49 | Zeeek | no, you hit # |
| 17:03.58 | Zeeek | and then dial a number like 700 |
| 17:04.06 | Zeeek | depends on your config |
| 17:04.07 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 17:04.13 | *** part/#asterisk Alexi1 (~alexis@www.trim.it) |
| 17:04.21 | Zeeek | that needs certain options in Dial command |
| 17:04.22 | mnet | ok thanks i'll look through sample parking.conf then :) |
| 17:04.35 | Zeeek | that's astart, and the ubiquitous wiki |
| 17:04.47 | mnet | yes, i'm already on it ? |
| 17:04.50 | mnet | on it ! |
| 17:04.51 | mnet | :) |
| 17:05.33 | phantam | is there a room somewhere for oh323 |
| 17:05.34 | phantam | lol |
| 17:05.53 | jsolares | ahhh informix sucks |
| 17:05.55 | Zeeek | yes, it's #babes-who-suck |
| 17:06.08 | jsolares | why the hell do i need to have it installed to compile the perl module to access a remote server.... |
| 17:06.18 | jsolares | meh my ivr plans have been foiled |
| 17:06.23 | Zeeek | you saaid it already, informix sucks |
| 17:06.29 | *** part/#asterisk Simon-- (~sim@staff-nat.netnation.com) |
| 17:06.39 | Zeeek | or it did in 1987 when I last looked |
| 17:06.49 | *** join/#asterisk ezabi (~ezabi@82.201.231.104) |
| 17:06.51 | Zeeek | didn't realize it even still existed :) |
| 17:06.53 | jsolares | it does indeed, and big time |
| 17:08.13 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:08.56 | *** join/#asterisk PMantis (~PMantis_C@66.251.89.34) |
| 17:09.54 | jsolares | hmm php has informix driver |
| 17:10.24 | PMantis | Can anyone point me to a website decribing how to run mirrored * servers in a hot standby scenario? |
| 17:10.25 | jsolares | now to port my standard tts/ivr code from perl agi to php agi... fun fun >_< |
| 17:11.05 | Zeeek | mv script.pl script.php and add the <?php ?> |
| 17:11.05 | phantam | jas amypme ised oh323 |
| 17:11.06 | phantam | lol |
| 17:11.10 | phantam | has anyone i meant |
| 17:11.40 | Zeeek | then look for all the unreadable gobbldy gook and make it readable |
| 17:13.24 | jsolares | hehe, wish me luck... it's been too long since i coded in php |
| 17:13.36 | Zeeek | like riding a bicycle |
| 17:13.43 | Zeeek | or sex |
| 17:13.48 | PMantis | * do redundancy, failovers? |
| 17:13.51 | Zeeek | or having sex on a bicycle |
| 17:14.00 | harryvv | pm yes |
| 17:14.03 | Zeeek | or with a bicycle |
| 17:14.09 | bjohnson | do you think a DID provider would forward to my FWD account? |
| 17:14.16 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
| 17:14.23 | Zeeek | CallUK |
| 17:14.25 | Zeeek | will |
| 17:14.27 | PMantis | harryvv, can you give me a nugde in the right direction? I need to put a proposal together. |
| 17:14.31 | sudoer | is there a java iax client or mac client? |
| 17:14.58 | PMantis | sudoer, maybe iaxcom has a mac client... |
| 17:15.10 | harryvv | PM netsurfer was the only person I know of that put one together and tested it. |
| 17:15.31 | PMantis | ~seen netsurfer |
| 17:15.35 | jbot | netsurfer <netsurfer@81-6-224-129.dyn.gotadsl.co.uk> was last seen on IRC in channel #asterisk, 13d 3h 45m 15s ago, saying: 'http://www.theregister.co.uk/2005/02/17/spam_gets_vocal_with_voip/ <-- ffs that takes the piss'. |
| 17:15.40 | PMantis | harryvv, thanks! :) |
| 17:16.02 | harryvv | Ipersonally dont know what happened to him mabey he is in the hospital mabey he is on a long extended project. |
| 17:17.17 | harryvv | so why did the authors of * select fudora |
| 17:17.53 | tzanger | harryvv: who selected fedora? |
| 17:18.18 | Zeeek | anyone know any dedicated server hosting? |
| 17:18.20 | nestAr | anyone know how to reorganize the soft keys on an IP300 with ipmid.cfg? i since there's a hard button for HOLD, i want to replace the soft hold button with transfer |
| 17:18.30 | harryvv | tzanger, Seemed I read that the first distros asterisk was installed on was fudora core |
| 17:18.33 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 17:21.05 | mnet | ianother question, is there a way to have asterix working with a "standard" modem (with 1 line and 1 phone port) connected via serial port |
| 17:21.45 | *** join/#asterisk TSCHAK (tschak@cuodan.net) |
| 17:22.08 | TSCHAK | what compiler/toolchain setup do i need to compile PWLib 1.5.2/OpenH323 1.12.2 ? |
| 17:22.22 | tzafrir_home | mnet: a "regular" modem has separate "phone" line. The "phone" line is simply connected. |
| 17:22.22 | TSCHAK | I am trying to get ANY h.323 plugin working under asterisk. |
| 17:22.26 | MENEEDoh323HelpP | why u installing that old one? |
| 17:22.30 | MENEEDoh323HelpP | TSCHAK |
| 17:22.32 | MENEEDoh323HelpP | i can help u |
| 17:22.34 | MENEEDoh323HelpP | msg me |
| 17:22.44 | MENEEDoh323HelpP | that much of it i did get to work lol |
| 17:24.05 | mnet | yes i meant can asterisk handle it ? (hanging up, menu, dtmf etc...) |
| 17:24.23 | pimpwell | anyone have the direct link to the .call file tutorial on the wiki? |
| 17:24.30 | pimpwell | I do a search and get 77 pages, 1k links |
| 17:25.16 | Zeeek | this may help |
| 17:25.17 | Zeeek | http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20wake-up |
| 17:25.18 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-11-74.d4.club-internet.fr) |
| 17:25.24 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:25.26 | pimpwell | ya thats the only one I have bookmarked |
| 17:25.28 | pimpwell | thanks though :) |
| 17:25.49 | Zeeek | well isn't it obvious when you look at the construction of the file int hat app? |
| 17:25.57 | pimpwell | theres a lot more to it |
| 17:26.01 | *** join/#asterisk lyroy (~lyroy@modemcable007.224-203-24.mc.videotron.ca) |
| 17:26.06 | pimpwell | that it doesnt cover |
| 17:26.23 | HitTop | may i ask a quesiton for queue.conf. for parameter announce = bla.. is bla suppose to be a sound file? |
| 17:26.49 | lyroy | Does someone could tell me how can I add my cell phone to a queue as a member of the queue? |
| 17:26.50 | bjohnson | do you think a DID provider would forward to my FWD account? |
| 17:26.51 | PoWeRKiLL | Hi |
| 17:27.06 | PoWeRKiLL | I suddenly get this error any idea Mar 2 17:49:44 WARNING[6164]: Unable to allocate socket: Too many open files ? |
| 17:27.14 | Zeeek | http://www.voip-info.org/wiki-Asterisk+tips+callback |
| 17:27.49 | HitTop | lyroy: under queue.conf add a queue list and add something like member => ZAP/g0/You cell number |
| 17:28.24 | lyroy | HitTop and if weth to dial via a IAX provider what will be the syntax? |
| 17:29.04 | *** join/#asterisk marshall (~test@S0106000f66563988.wp.shawcable.net) |
| 17:29.15 | Zeeek | pimpwell: |
| 17:29.17 | HitTop | lyroy: again, something like member => IAX2/bla depends on ur configuration in iax.conf |
| 17:29.20 | pimpwell | tyty |
| 17:29.23 | Zeeek | http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out |
| 17:29.29 | Zeeek | that's the one |
| 17:29.37 | lyroy | alright thanx |
| 17:29.46 | Zeeek | the secret trick |
| 17:29.49 | pimpwell | hehe |
| 17:29.58 | Zeeek | search for /var/spool/asterisk/outgoing |
| 17:30.10 | Zeeek | only 63 hits! |
| 17:30.14 | pimpwell | ty ty ty ty :o |
| 17:30.28 | HitTop | could someone tell me wat is the parameter announce stands for under queue.conf? |
| 17:30.51 | Sedorox | I would think the time that it announces their place in line |
| 17:30.56 | Zeeek | I'd guess it means a way to tell the agent who is calling? |
| 17:31.11 | Zeeek | ah, different guesses, this could lead to betting and wagering |
| 17:31.54 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 17:31.55 | bjohnson | for a simple home setup, I wonder if a DID provider would forward incoming to FWD to use their voicemail, meetme, etc and then the home user could just have a single device to worry about but get many of the * features |
| 17:32.36 | Zeeek | bjohnson I think they will, and I said many do like CallUK but who the fsck would want to depend on FWD being up to get calls? |
| 17:32.59 | Zeeek | they're about as reliable as my brother in law |
| 17:33.33 | Zeeek | the list of possibilities is on the FWD site and also ask in the forum - they'd know there |
| 17:33.54 | bjohnson | I haven't found FWD to be unreliable |
| 17:34.04 | *** join/#asterisk nikko_ (~nikko@69.85.201.170) |
| 17:34.07 | Zeeek | glad for you but many people do |
| 17:34.13 | bjohnson | this would be for a home user who wouldn't be able to deal with * |
| 17:34.40 | Zeeek | why not go with one good service instead chaing them? just my opinion |
| 17:34.50 | bjohnson | many people have problems with FWD? |
| 17:34.58 | Zeeek | ever been on the forum? |
| 17:35.13 | bjohnson | Zeeek: which good service? you mean a plan? |
| 17:35.19 | Zeeek | it's agreat and helpful place - go check it out |
| 17:35.21 | nikko_ | Hello |
| 17:35.36 | Zeeek | go ask on the FWD forum - that's where you'll get valid info |
| 17:35.46 | Zeeek | there are a lot of smart folks there |
| 17:36.01 | Zeeek | [and they give a shit about FWD] |
| 17:36.03 | bjohnson | DIDs are hard to get here in Canada .. other than vonage type providers, iax.cc seems to be only one with good selection |
| 17:36.19 | bjohnson | but they don't provide voicemail |
| 17:36.27 | Zeeek | why don't you ask them if they'd do the FWD thing? |
| 17:36.36 | bjohnson | waiting for an answer |
| 17:36.41 | Zeeek | sell it boy! |
| 17:36.45 | bjohnson | but now you've got me questioning reliability |
| 17:37.00 | Zeeek | You didn't answer - HAVE YOU been on the FWD forum |
| 17:37.02 | nikko_ | how do I determine if extensions.conf is getting parsed or has an error? show dialplan only shows the parkedcall extension |
| 17:37.13 | bjohnson | no .. I didn't even know they had one |
| 17:37.16 | Zeeek | nikko_ look at the CLI messages |
| 17:37.24 | bjohnson | despite being through their site a fair bit |
| 17:37.32 | Zeeek | http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi#general_cat |
| 17:37.46 | nikko_ | no amount od debug or vvv's will ever show it even being opened in the CLI messages |
| 17:37.48 | Zeeek | Also go to dslreports if you haven't been |
| 17:38.24 | Zeeek | nikko_ what do you see on CLI ? |
| 17:38.43 | nikko_ | nothing with extensions.conf in it |
| 17:38.44 | Zeeek | dslreports has people all over using all the various services (incl FWD) |
| 17:38.57 | Zeeek | the question is: what do you see? |
| 17:38.59 | bjohnson | well .. I must have found it before. Seems I have an account setup already |
| 17:39.02 | Zeeek | when you make the call |
| 17:39.29 | nikko_ | oh - hang onMar 2 11:37:30 NOTICE[23803]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 172.31.30.20, request '105@default' does not exist |
| 17:39.42 | Zeeek | does that tell you anything at all? |
| 17:39.58 | nikko_ | yeah, that it can't find my default context |
| 17:40.00 | Zeeek | such as there is no such number? |
| 17:40.01 | nikko_ | not much more |
| 17:40.09 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 17:40.11 | Zeeek | no there's no 105 in default |
| 17:40.12 | nikko_ | it's there |
| 17:40.17 | *** part/#asterisk ezabi (~ezabi@82.201.231.104) |
| 17:40.19 | Zeeek | reload extensins |
| 17:40.42 | JohnnyC | Hello all, I bought a Fritz PCI card but I cant find info on how to configure it with Fedora for CAPI support |
| 17:41.05 | nikko_ | I did, and restarted asterisk |
| 17:41.23 | nikko_ | is ther a way to determine if it's even being loaded on startup |
| 17:41.26 | nikko_ | ? |
| 17:41.30 | Zeeek | nikko_ a lot of the error messages are wacky and incomprehensible, but the one you show is pretty obvious |
| 17:41.31 | harryvv | anyone here running * on deb? |
| 17:41.58 | nikko_ | yeah, that's how I know it's not reading in the file, I'm trying to determine why |
| 17:42.06 | nikko_ | show dialplan gives nothing |
| 17:42.18 | nikko_ | except the parked_call extension |
| 17:42.27 | Zeeek | when you start asterisk it tells you every thing it is doing |
| 17:42.33 | nikko_ | That's what I'm trying to figure out |
| 17:43.00 | Zeeek | so go to the top of the output and read every line to see what it does when it starts up |
| 17:43.20 | Zeeek | is anything elsle working? |
| 17:43.30 | PMantis | Anyone ever use AMP and astguiclient on the same server? Does one replace the other? |
| 17:43.45 | nikko_ | yes, everything else seems to work IAX clients can register, etc |
| 17:44.26 | Zeeek | any working extensions? |
| 17:44.31 | nikko_ | no |
| 17:44.34 | Zeeek | ah. |
| 17:44.43 | Zeeek | well that isn't too useful for the pbx |
| 17:45.08 | Zeeek | have you seen this: |
| 17:45.11 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. "http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 17:45.13 | nestAr | anyone changed the softbutton configuration on a Polycom? I'm stumped. |
| 17:45.27 | Zeeek | Ask ManxPower he ahs an ip500 |
| 17:45.39 | Zeeek | may be back later |
| 17:45.42 | nestAr | yea |
| 17:46.48 | ManxPower | *sigh* I managed to cut myself open a jar of peanutbutter. |
| 17:46.52 | nestAr | lol |
| 17:48.11 | ManxPower | nestAr, I've not done it, but I've been told by someone that has done it (a long time ago), you change the button LABELS using the localization features. |
| 17:48.20 | ManxPower | I don't recall how you change the FUNCTION of the button |
| 17:48.35 | Zeeek | USELESS! |
| 17:48.49 | Zeeek | first he cuts himself, then he can't remember |
| 17:49.05 | nestAr | hrmmm.. i'm just trying to move Transfer to where Hold is currently |
| 17:49.11 | nestAr | i'll look |
| 17:49.23 | JohnnyC | anyone uses Fritz PCI card ? |
| 17:50.19 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 17:50.34 | neopher | hello everyone |
| 17:51.32 | neopher | does * support G.729? |
| 17:51.43 | nikko_ | Zeek, here's my asterisk startup: |
| 17:51.47 | nikko_ | http://pastebin.ca/6728 |
| 17:53.30 | wolfson | neopher: if you own a license, yes |
| 17:54.25 | Zeeek | nikko_ you need to make your self a list of those errors |
| 17:56.25 | JohnnyC | anyone with AVM Fritz PCI Card ? |
| 17:56.36 | mishehu | neopher: w/o a license, * only supports g729 passthru |
| 17:57.04 | Juggie | ManxPower, i was camping one time, and when i returned a squirl had eaten through the cover of the penut butter i left on the table. |
| 17:58.59 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 18:00.03 | neopher | mishehu: sorry to be dumb on this, but pass through means if i have a phone that supports G.729 and a sip provider that supports G.729 then it will work? |
| 18:01.17 | neopher | budgetone phone -> asterisk -> Broadvoice |
| 18:02.50 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 18:02.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:03.37 | *** join/#asterisk Kokomo (~databoot@wbb30.fwa3.jaring.my) |
| 18:03.59 | Kokomo | greeting everyone. I have some question about the scalability of asterisk server |
| 18:04.04 | Kokomo | anyone that can help ? |
| 18:04.17 | JerJer[mobile] | how about asking a specific question? |
| 18:04.18 | PoWeRKiLL | I suddenly get this error any idea Mar 2 17:49:44 WARNING[6164]: Unable to allocate socket: Too many open files ? |
| 18:04.29 | JerJer[mobile] | close some files |
| 18:04.31 | Zeeek | that was a specific question |
| 18:05.54 | JerJer[mobile] | Zeeek: then answer his specific question |
| 18:06.03 | Zeeek | the answer is yes |
| 18:06.17 | Kokomo | what about, can I make asterisk to work in a p2p mode, without going through the asterisk server ? |
| 18:06.31 | Kokomo | that would lessen the load at the server side |
| 18:06.32 | Zeeek | Kokomo read this |
| 18:06.33 | Zeeek | Starter tutorial: |
| 18:06.33 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 18:06.33 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 18:06.33 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 18:06.33 | Zeeek | THE reference of the moment: |
| 18:06.34 | Zeeek | http://www.asteriskdocs.org |
| 18:06.39 | PoWeRKiLL | JerJer[mobile] I got a broken pipe on asterisk |
| 18:06.57 | Zeeek | the first link expalins what asterisk is and should tell you |
| 18:07.05 | PoWeRKiLL | I did a restart and work good now but why it's happens ? |
| 18:07.51 | Kokomo | thanks Zeeek ! |
| 18:07.53 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 18:08.04 | JerJer[mobile] | PoWeRKiLL: fix t |
| 18:08.07 | JerJer[mobile] | fix it |
| 18:08.43 | PoWeRKiLL | what do I have to fix I just make a restart and it's work but I want to know why it's happens |
| 18:10.58 | Zeeek | Kokomo happy reading |
| 18:11.21 | *** join/#asterisk strace (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 18:11.23 | strace | hey you guys |
| 18:11.25 | strace | with manager.conf |
| 18:11.27 | strace | I'm getting |
| 18:11.33 | strace | Mar 2 18:09:19 NOTICE[6416]: channel.c:1817 __ast_request_and_dial: Unable to request channel sip/lemos |
| 18:11.34 | strace | why? |
| 18:11.37 | strace | :( |
| 18:12.12 | Zeeek | the lemos has left |
| 18:12.13 | ManxPower | Maybe it has to be SIP/lemos or maybe lemos is not registered with Asterisk |
| 18:12.32 | strace | <PROTECTED> |
| 18:12.34 | strace | it is... |
| 18:12.36 | strace | any more thoughts? |
| 18:13.24 | Primer | pardon my ignorance, but is there no console command to drop a sip client? this client has disconnected long ago |
| 18:14.18 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:14.30 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 18:15.00 | Defraz | I have a SPA2000 and I can't seem to get dtmf passing on when checking voice mail or anything |
| 18:17.12 | Defraz | does anyone have any ideas. |
| 18:18.09 | Delvar | set both the phone and asterisk to the same DTMF mode? info or rfc2833 |
| 18:18.18 | Primer | rfc2833 |
| 18:18.31 | Primer | should work |
| 18:19.17 | Delvar | duno then, try inband and use alaw/ulaw see if that works. |
| 18:19.53 | PatrickDK | hmm, rfc2833 works with my spa2000 |
| 18:19.57 | *** join/#asterisk innerweb (~innerweb@pcp0010181839pcs.columbus.in.indy.comcast.net) |
| 18:20.17 | *** join/#asterisk afe ([jRWUk8AW+@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:20.32 | Delvar | are you sure you have 'dtmfmode=rfc288' in your sip entity? |
| 18:20.38 | Delvar | oops |
| 18:20.43 | Delvar | are you sure you have 'dtmfmode=rfc2833' in your sip entity? |
| 18:20.46 | Defraz | yea |
| 18:20.48 | Defraz | I do |
| 18:21.01 | Defraz | I am trying inband |
| 18:23.35 | *** join/#asterisk afe ([LvxDmlSfD@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:23.52 | Primer | actually I have inband |
| 18:24.07 | Primer | I forgot my sipura connected to a different asterisk |
| 18:24.11 | Delvar | lol |
| 18:24.16 | Defraz | oh I see |
| 18:24.35 | Primer | sorry |
| 18:24.45 | *** part/#asterisk drvoip (user@S01060050baab8e4b.cg.shawcable.net) |
| 18:24.46 | Primer | but I don't see why rfc wouldn't work |
| 18:25.22 | Delvar | firmware? try upgreading |
| 18:25.30 | Delvar | asterisk? try updating :) |
| 18:25.34 | Primer | dammit, is there no way to forcibly terminate a sip client? |
| 18:25.43 | *** join/#asterisk Mneumonic (Mnemonic@ool-18ba58b4.dyn.optonline.net) |
| 18:25.43 | Zeeek | restart now |
| 18:25.47 | Delvar | yep |
| 18:25.51 | Primer | my asterisk is sending data to this long disconnected client |
| 18:26.00 | Delvar | stop now |
| 18:26.12 | Primer | but then that disconnects everyone, no? |
| 18:26.16 | Defraz | okay it works find with the local voice mail but the minute I connect to a vsystem on the end of my call it doesn't make it threw. |
| 18:26.21 | Delvar | there is a way to hangup channels but i cant remember |
| 18:26.21 | Zeeek | it's for the best |
| 18:26.31 | Zeeek | soft hangup |
| 18:26.42 | *** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 18:26.42 | innerweb | Has anyone had any experiences with the generic x100p card on a 2.6.10 kernel? |
| 18:26.46 | Primer | bah |
| 18:26.47 | Zeeek | or maybe soft towel hangup |
| 18:26.54 | *** join/#asterisk bannerman (~bannerman@dpc6682105089.direcpc.com) |
| 18:26.57 | Goshen | innerweb: mine works great with 2.6.10 |
| 18:27.02 | Primer | innerweb: the intel537? |
| 18:27.07 | Zeeek | can't you restart gracefully ? |
| 18:27.14 | Primer | I did |
| 18:27.19 | Zeeek | and? |
| 18:27.19 | Goshen | its the generic from digitnetworks.com |
| 18:27.22 | Primer | but I wanted to know how to hang up the client |
| 18:27.23 | Primer | it's fine |
| 18:27.30 | Primer | I couldn't remember soft hangup |
| 18:27.40 | Zeeek | I think there is that |
| 18:27.40 | Goshen | it is the card that is featured on the asterisk.com website |
| 18:27.44 | Primer | but it clicked in my brain as soon as you mentioned it, but it was too late |
| 18:27.51 | Primer | thanks |
| 18:27.56 | Goshen | at least it looks exactly the same when I compare the board with the picture |
| 18:28.00 | Zeeek | I just made that up. Does it really exist? |
| 18:28.10 | Primer | yes |
| 18:28.13 | Zeeek | good |
| 18:28.14 | bannerman | A couple of weeks ago someone came in asking about VoIP over a 2-way satellite, and I told him I thought it was impossible because of the latency. I was wrong. If nothing else is hitting the upstream at the same time, it's about 1.5 second delay, which isn't great, but is usable for conversation. |
| 18:28.25 | bannerman | Just fyi :) |
| 18:28.26 | Zeeek | proves my concept that the universe is a cosntruct of my mind |
| 18:28.40 | phantam | 1.5 second |
| 18:28.43 | Primer | heh, old POTS shit used to go through satelite |
| 18:28.44 | phantam | its max 800ms |
| 18:28.46 | Inv_arp | bannerman: heh think it was me |
| 18:28.47 | phantam | round trip |
| 18:28.49 | Goshen | asterisk.org |
| 18:28.49 | Primer | the latency was horrible |
| 18:28.54 | phantam | unless that sat really sucks |
| 18:29.05 | innerweb | I am doing somehting wrong. I can not get the channel up. |
| 18:29.22 | phantam | i wish i could get my damn oh323 to talk to my cisco |
| 18:29.23 | Inv_arp | ive used gsm over a modem wasnt bad... |
| 18:29.24 | phantam | but no |
| 18:29.25 | Goshen | did the drivers load? |
| 18:29.28 | Zeeek | I remember making expensive toll calls that sucked way worse than the worst voIP call... in 19801 |
| 18:29.30 | phantam | it has to give an asshole error |
| 18:29.36 | phantam | gsm is only 8k |
| 18:29.38 | innerweb | lsmod shows the drivers. |
| 18:29.40 | Zeeek | 1981 |
| 18:29.40 | phantam | i would hope it would be good |
| 18:29.45 | phantam | modem = low latency round 200ms |
| 18:29.57 | Goshen | did you edit zaptel.conf, zapata.conf? |
| 18:30.00 | *** join/#asterisk bassie (~bas@datarack.xs4all.nl) |
| 18:30.09 | bassie | hello |
| 18:30.11 | phantam | technically modem could handle 2 or 3 gsm calls |
| 18:30.21 | bannerman | satellite is minimum of like 800 ms |
| 18:30.27 | bannerman | it's impossible to get much below that |
| 18:30.30 | bannerman | like quite literally |
| 18:30.35 | bassie | has anybody succesfully setup music on hold? |
| 18:30.46 | Inv_arp | bassie: yea |
| 18:31.01 | innerweb | yep. loadzone=US \n defaultzone=US \n fxsks=1 \n |
| 18:31.04 | Juggie | bassie, its broken in 1.0.6 if you are running that. |
| 18:31.29 | bassie | cool... the problem I am having is that when I start dialing, the music sounds already.. |
| 18:31.34 | bassie | I am running 1.0.5 |
| 18:31.55 | Juggie | issue with your dialplan |
| 18:31.57 | innerweb | zapata context=demo \n group =1 \n signalling = fxs_ks \n context = incoming \n channel => 1\n |
| 18:32.09 | *** join/#asterisk afe ([1tA51miuy@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 18:32.12 | afe | help |
| 18:32.13 | bassie | Is there anyone willing to share an example of extension.conf (just 1 phone entry) |
| 18:32.18 | afe | lol - oops |
| 18:32.21 | bassie | :) |
| 18:32.23 | JerJer[mobile] | so like its march, so where are these killer cards from Atacomm |
| 18:33.02 | phantam | bassie i would but my music on hold doesnt work either |
| 18:33.02 | phantam | lol |
| 18:33.18 | phantam | bkw_: werent u the one that was helpin me before? |
| 18:33.23 | bassie | hehe :) |
| 18:33.29 | bassie | thanks anyway phantam |
| 18:33.49 | phantam | i keep gettin those mohmp3 doesnt exist errors |
| 18:33.52 | phantam | not sure how to get rid of it |
| 18:33.53 | phantam | either |
| 18:34.09 | bassie | you should have a dir /var/lib/asterisk/mohmp3 |
| 18:34.13 | Juggie | download and install asterisk-sounds |
| 18:34.23 | innerweb | then, whne I do this.. ztcfg -vv , I get 0 channels found |
| 18:34.46 | neopher | is there a why to tell what codec is being used during a call on the CLI |
| 18:34.51 | Juggie | or does mohmp3 come with the main asterisk distribution? i forget. |
| 18:34.54 | innerweb | It tends to prevent asterisk from starting. lol Had to choose the weekend to upgrade teh box. |
| 18:35.02 | bassie | mohmp3 is standard |
| 18:35.17 | Inv_arp | neopher: sip show channels |
| 18:35.30 | bassie | I placed the startrek theme in that dir for the sake of testing.... |
| 18:35.33 | neopher | tnx |
| 18:35.40 | Delvar | nn all |
| 18:35.49 | Juggie | neopher, sip show channels |
| 18:36.03 | bassie | now when I call another extension, the theme is played |
| 18:36.08 | Goshen | innerweb: see pm |
| 18:36.11 | phantam | mohmp3 folder exists |
| 18:36.17 | Inv_arp | bassie: paste your extension.conf pastebin.ca |
| 18:36.29 | innerweb | pm? (Perl Monks? |
| 18:36.31 | bassie | phantam, with fpm files in it? |
| 18:36.35 | Goshen | Juggie: I think it is in the asterisk addons |
| 18:36.37 | phantam | no3 fukes |
| 18:36.40 | phantam | mp3 files |
| 18:36.40 | phantam | lol |
| 18:37.03 | mishehu | innerweb: pm == "pokemon" |
| 18:37.19 | Goshen | innerweb: query window...personal message |
| 18:37.23 | bassie | phantam, that's good |
| 18:37.30 | neopher | hmm, anything special that needs to be installed for * to passthrough G.729 |
| 18:37.36 | bassie | have you done "make mpg123" in the asterisk source dir? |
| 18:37.47 | phantam | emerge'd it |
| 18:37.50 | innerweb | I am on a text based connection. (kind of like fancy telnet.) |
| 18:38.11 | Goshen | did you see my zapata.conf come across then? |
| 18:38.18 | innerweb | Yes. |
| 18:38.21 | Goshen | ok |
| 18:38.28 | Goshen | that is my config for my generic card |
| 18:38.35 | innerweb | Sorry. I normally use other methods to connect, not used to this. |
| 18:38.51 | *** join/#asterisk Elshar (~Elshar@ip206-91.oregonfast.net) |
| 18:39.09 | innerweb | I put those change sin and got hits:chan_zap.c:769 zt_open: Unable to specify channel 1: No such device or address |
| 18:39.11 | Goshen | you want my to put my zapata.conf at pastebin.ca? |
| 18:39.26 | Goshen | did you reload? |
| 18:39.38 | innerweb | * was not running when I did. |
| 18:39.43 | Goshen | ok |
| 18:40.13 | *** join/#asterisk Ahewes (~rsb@209.81.2.58) |
| 18:40.21 | innerweb | lsmod gives zaptel 222180 3 ztdummy,wcfxs,wcfxo |
| 18:40.31 | innerweb | Is the dummy a problem? |
| 18:40.55 | Goshen | I wouldn't load ztdummy |
| 18:41.03 | Goshen | or wcfxs |
| 18:41.18 | innerweb | Ok, its gone |
| 18:41.37 | innerweb | I tried to start * again and got the same fatal error. |
| 18:41.52 | Goshen | zaptel, crc_ccitt, wcfxo |
| 18:41.54 | dsmouse | YAY! my digium card is here! |
| 18:42.03 | dsmouse | now I just need to figure out how to use it |
| 18:42.22 | *** join/#asterisk JimVanM (~jimvanm@HSE-Toronto-ppp180870.sympatico.ca) |
| 18:42.25 | Goshen | I ahve crc_ccitt, because zaptel complained about not having it when I compiled it |
| 18:43.10 | innerweb | okI do not seem to have a crc_ccitt. But, I do remembber compiling it with the kernel. |
| 18:43.40 | innerweb | That is thee one in the kernel options, right? |
| 18:44.19 | Goshen | I beleive so yes |
| 18:44.32 | innerweb | Ok, then it is compiled. Do I use modprobe to load it? |
| 18:44.56 | Goshen | I think my programmer loaded it as a module because it wasn't compiled in |
| 18:45.00 | Goshen | not sure |
| 18:45.15 | Goshen | if you have it in the kernel you might not need to load it as a module |
| 18:45.27 | innerweb | Ok.. I will double check. Never hurts. |
| 18:46.27 | Goshen | <PROTECTED> |
| 18:46.30 | Goshen | thats good news :) |
| 18:46.49 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
| 18:48.18 | Crad|Work | Ok I have a question (well actually two) hopefully someone can help me out with... in extensions.conf one specifies an extension number such as |
| 18:48.36 | Crad|Work | exten => 123,priority,command |
| 18:48.43 | Crad|Work | where the extension number is 123, right? |
| 18:49.13 | Mneumonic | yes |
| 18:49.13 | Crad|Work | I'm trying to pass the extension number (without using the caller id variable specified in sip.conf) to pass into the voice mail bits. |
| 18:49.36 | Crad|Work | but, in reviewing pbx.c and all the variables, and playing a bit with the variables in the live config... |
| 18:49.40 | innerweb | That might be a winner. i may have compiled it wrong (as a static, not a module). |
| 18:49.40 | Crad|Work | nothing seems to be carrying that value. |
| 18:49.49 | Crad|Work | Any idea how I can get to it? |
| 18:50.01 | Crad|Work | I dont mind modifying the code, if I know where to get the value |
| 18:50.08 | innerweb | I had this system working so well last week, and I just had to see if it would work with 2.6.10 |
| 18:50.23 | innerweb | <PROTECTED> |
| 18:50.35 | Goshen | innerweb: np, good luck :) |
| 18:50.42 | bjohnson | Linksys WRT54G on sale for $90 - $15 MIR and free shipping at Staples.ca |
| 18:51.03 | Crad|Work | or is there a wayin the code to look up the "SIP/<phone id>-<unique id>" value internally in pbx.c ? |
| 18:51.22 | |Vulture| | bjohnson: you can get them for like $70 at BestBuy |
| 18:51.38 | |Vulture| | the WRT54G is the best router for the $$ I have ever had |
| 18:51.52 | bjohnson | |Vulture|: wrong country and currency though |
| 18:52.22 | |Vulture| | bjohnson: ah that makes more sense ;) |
| 18:52.56 | JerJer[mobile] | and asterisk runs beautifully on the wrt's :) |
| 18:53.02 | innerweb | Goshen: Are you linked in through any peered Internet based * solutions? |
| 18:53.14 | Ahewes | I have to agree with |Vulture| on the wrt54g |
| 18:53.21 | Goshen | innerweb: speak english? :) |
| 18:53.25 | *** join/#asterisk Laloo3 (~laloo@042.142-60-66.FTTH-SWI.surewest.net) |
| 18:53.26 | innerweb | Yes. |
| 18:53.27 | Ahewes | Although I never got asterisk to work for more than one channel. |
| 18:53.33 | JerJer[mobile] | just the stupid ipkg for asterisk whoever built really sucks |
| 18:53.35 | JerJer[mobile] | realllly sucks |
| 18:53.38 | innerweb | Nothing much else though (unfortunately). |
| 18:53.43 | Goshen | innerweb: ask again..I don't understand your question...I dial out over voip yes |
| 18:53.58 | *** join/#asterisk Schism (~schism@clt74-101-230.carolina.rr.com) |
| 18:54.00 | JerJer[mobile] | Ahewes: i've had 5 calls into a meetme |
| 18:54.05 | Goshen | innerweb: I only speak english too, and medical :) |
| 18:54.06 | JerJer[mobile] | and 4 gsm channels going |
| 18:54.11 | JerJer[mobile] | out via iax |
| 18:54.21 | JerJer[mobile] | seperately |
| 18:54.24 | innerweb | If one were to connect * servers in server to server (some called peered) network, it creates a virtual phonenet. |
| 18:54.34 | Goshen | innerweb: I dial out over nikotel(till I use up my credit) sip, FWD for 1-800 numbers, voipuser for fun |
| 18:54.39 | Mneumonic | anyone know the line of code that goes in extensions.conf to set the callerID Name? |
| 18:54.40 | *** join/#asterisk Goldenear (~goldenear@d193.dhcp212-198-200.noos.fr) |
| 18:54.43 | Laloo3 | guys. Can someone help me? There is some problem with the default voicemail setup. After listening to all the voicemail messages, asterisk prompts you with "press # to exit." When I do that, Asterisk just quits altogther. |
| 18:55.00 | Goshen | innerweb: you can do that with http://www.e164.org/ |
| 18:55.02 | Crad|Work | Mneumonic: I believe that's in sip.conf |
| 18:55.06 | Ahewes | JerJer I get serious line quality issues with three channels outbound running concurrently. Two was o.k., one was rock solid. |
| 18:55.16 | Ahewes | I'm using sipuras and g711u, nothing fancy |
| 18:55.16 | marshall | anyone here experiencing echo when using the IAXY ATA? |
| 18:55.16 | Laloo3 | I see the following message in the log file. "Maximum retries exceeded" |
| 18:55.20 | |Vulture| | put it this way I am replacing all my office routers (Netgear FVS328) with WRT54G routers... that says something |
| 18:55.21 | neopher | Crad|Work: no is in ext |
| 18:55.21 | innerweb | Some of the guys I have been talking to have not used an L/D provider in aover a year by peering their servers and provding local dial through. |
| 18:55.24 | Crad|Work | assigned to each phone, also it's in zapata.conf |
| 18:55.26 | Goshen | # is used to transfer calls |
| 18:55.34 | Ahewes | I have the wrt54gs, which i thought might be a faster, bigger version |
| 18:55.38 | innerweb | Coo. I have not been there yet. Thanks. |
| 18:55.52 | Crad|Work | neopher: I have it in sip.conf attached to each phone :x |
| 18:55.53 | Mneumonic | Crad - on my outbount part of extensions.conf i have SetCallerID("<###########>") |
| 18:55.54 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) |
| 18:55.56 | Laloo3 | Goshen, are you talking to me? |
| 18:56.02 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 18:56.07 | Goshen | Laloo3: yes |
| 18:56.12 | Mneumonic | just need the name |
| 18:56.20 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 18:56.28 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 18:56.32 | Crad|Work | interesting... any benifit in having it in extensions.conf instead of sip.conf? |
| 18:56.32 | Goshen | when I press # on my asterisk box by defauly it parks the calls to music on hold |
| 18:56.34 | Laloo3 | But I did not assign #. This is the default voicemail setup. |
| 18:56.55 | Goshen | # is a default, at least in my 1.0.6 |
| 18:57.01 | Goshen | to park the call |
| 18:57.13 | Laloo3 | then why does Asterisk quit? |
| 18:57.23 | Goshen | are you watching in your console? after starting it with asterisk -cvvv ? |
| 18:57.29 | Goshen | see what it says |
| 18:57.44 | Laloo3 | no. I can do that. I actually was looking at /var/log/asterisk/messages file. |
| 18:58.02 | Crad|Work | I'm dealing with caller id issues right now - I want all outbound calls out onto the t1 to use our external number for callerid and all internal ones to use the extension #.... it seems to be one or the other... overriding it in zapata.conf doesnt work for the caller id number just the name :| |
| 18:58.04 | innerweb | I must go now. Thanks for your help again, Goshen |
| 18:58.10 | Goshen | good luck innerweb |
| 18:58.56 | Goshen | Crad: what if you made seperate contexts for internal and external? |
| 18:58.58 | *** join/#asterisk bobx (~bobx@206.124.165.14) |
| 18:59.02 | Inv_arp | Crad|Work: SetCallerID(${IDCALLER}) |
| 19:00.19 | *** join/#asterisk zapa (zapa@200.66.21.168) |
| 19:00.32 | Crad|Work | Goshen: that's what I'm trying to figure out how to do :o |
| 19:01.10 | Goshen | put your config at pastebin.ca |
| 19:02.41 | Laloo3 | Goshen. I just ran Asterisk option. When I press #, Asterisk plays "goodbye" message. Followed by a segmentation fault :-(. |
| 19:03.59 | Laloo3 | Any idea on what I must do? |
| 19:04.52 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 19:04.52 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 19:04.53 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 19:05.12 | Goshen | Laloo: what version are you running? |
| 19:05.15 | algorithmn | if a sip ata is sending nat-keep alive data while the sip user info is in a mysql realtime database, will there be any conflicts over time? |
| 19:05.23 | Laloo3 | how can I check that? |
| 19:05.45 | Goshen | if you didn't compile it within the last couple days, it is old...upgrade :) |
| 19:05.52 | Goshen | it shows when you run asterisk... |
| 19:06.14 | JonR800 | at the cli type "show version" |
| 19:06.19 | Goshen | you could asterisk -cvvvvv > asterisk.txt |
| 19:06.26 | Goshen | and look at the top of the file, it shows you the version |
| 19:06.41 | Laloo3 | ok |
| 19:07.01 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 19:07.25 | JonR800 | if you can get it up and running.. show version will work :) |
| 19:07.30 | Goshen | lol |
| 19:07.34 | Goshen | or that :) |
| 19:08.14 | Goshen | or connect with asterisk -r, shows you the version |
| 19:08.20 | Goshen | 1.0.6 is the latest |
| 19:08.33 | JonR800 | true |
| 19:08.34 | Crad|Work | is there a way to do an if statement in extensions.conf? such as if context == internal then setcallerid(x) else setcallerid(y)? |
| 19:08.34 | JonR800 | haha |
| 19:08.41 | JonR800 | a multitude of options :) |
| 19:09.02 | algorithmn | crad|work: agi? |
| 19:09.05 | Goshen | can't you set the callerid for each call? |
| 19:09.08 | hajekd | what can be the reason for variables are not seen in dial command? |
| 19:09.27 | Inv_arp | hajekd: what type of $vars globals? |
| 19:09.55 | hajekd | yes |
| 19:10.12 | hajekd | define a varibale in [global] but this variable is not seen in dial command.. |
| 19:10.25 | Goldenear | Hello. I've read many things about IAX versus SIP... and everything I read tells the good points of IAX an the difficulties of SIP with NAT traversal. Doesn't SIP have any benefits over IAX is some situations? |
| 19:10.36 | Inv_arp | hajekd: try NoOp(${variable}) and watch cli to see if its output to screen |
| 19:10.44 | Crad|Work | ultimately what would really help is if I could find a variable to replace EXTEN |
| 19:10.47 | Crad|Work | in "exten => 121,3,VoiceMailMain(${EXTEN}) |
| 19:10.52 | wildcard0 | Goldenear, far more support with non-asterisk products |
| 19:10.54 | Crad|Work | that shows the extension that dialed |
| 19:11.01 | Crad|Work | instead of the extension that was dialed |
| 19:11.09 | algorithmn | Goldenear: market popularity |
| 19:11.34 | Laloo3 | Goshen. Mine does not say the asterisk version. It says CVS-HEAD 12/23/04. |
| 19:11.44 | hajekd | inv_arp: its empty |
| 19:11.49 | Goldenear | Really ? SIP has no technical benefits over IAX ? |
| 19:11.55 | Goshen | Goldenear: sip is common, but so are VCRs :) Go with IAX=DVD player :) |
| 19:12.04 | algorithmn | nice anology |
| 19:12.18 | Goshen | Laloo: you are using the develoupment version...downgrade to the release version |
| 19:12.28 | hajekd | inv_arp: it has been working, but I think I have changed something in other configs. |
| 19:12.48 | Laloo3 | ok. I had downloaded this from digium website as we were using Digium board. |
| 19:12.55 | Inv_arp | hajekd: LAPTOP=SIP/x151 thats the convention under globals check to see if var name didn chge |
| 19:13.06 | Goshen | just got to asterisk.org and download the tarballs |
| 19:13.18 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 19:13.21 | Laloo3 | If I get the release version and simply run "make install," will it maintain all my config files? |
| 19:13.21 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 19:13.23 | *** join/#asterisk amir (~amir@shield.guindehi.ch) |
| 19:13.37 | Goshen | Laloo3: yes, just don't make demo |
| 19:13.42 | buddah | how is nufone's international routes? are they reliable? |
| 19:13.56 | Goshen | Laloo3: might want to cp /etc/asterisk to /etc/asterisk.bak |
| 19:13.56 | Laloo3 | ok. THanks. Let me get the release version. |
| 19:14.25 | Goshen | Wacking problems is fun....this feels like a whack-a-mole game... |
| 19:14.27 | dsmouse | I have a FXO card- one of the 100 series one port cards, and a TDM400P with 3 FXS ports... I'm having problems configuring, any ideas? |
| 19:14.36 | Goshen | I am just starting to get over the learning curve, so I am learning as well |
| 19:15.03 | Inv_arp | dsmouse: err u havent told us the problem |
| 19:15.12 | Goshen | dsmouse: is this a dedicated asterisk box? |
| 19:15.13 | tzafrir_home | Goshen: it's actually called troubleshooting. |
| 19:15.26 | dsmouse | Goshen: yes; |
| 19:15.30 | Goldenear | wow, I'm very surprise. So for you SIP is "has been" (and dead born in some way) ? |
| 19:15.40 | Goshen | dsmouse: did you go into the bios and disable everything you are not using? |
| 19:15.48 | Goldenear | so I guess IAX is the standard to go |
| 19:15.53 | Goshen | dsmouse: you will want to free up all of your unused interrupts |
| 19:16.26 | Goshen | Goldenear: I love IAX, sip is a bear to configure when any NAT is involved, having all of the communications go over one port is nice |
| 19:16.27 | tzafrir_home | dsmouse: well, you can use our script for that... |
| 19:16.30 | dsmouse | I can't seem to configure zaptel.conf to address all the ports... oh |
| 19:16.31 | dsmouse | here |
| 19:16.33 | dsmouse | dug |
| 19:16.35 | dsmouse | nm |
| 19:17.07 | bassie | Goshen, do you know what "plays" the ring-ring sound when calling another extension? I mean the sound on the phone I'm calling from, not the phone being called obviously |
| 19:17.22 | tzafrir_home | http://updates.xorcom.com/genzaptelconf |
| 19:17.39 | bassie | I fooled around with MOH, and I trashed whatever makes the ring-ring sound when the phone is ringing on the other side :( |
| 19:17.40 | Goshen | bassie: not yet |
| 19:17.45 | Goldenear | Is it possible to make a direct call from one phone (soft or hard) to an other with going thru Asterisk ? |
| 19:18.05 | *** join/#asterisk Rick_Hunter (~rhunter@05-162.008.popsite.net) |
| 19:18.22 | algorithmn | if a sip ata is sending nat-keep alive data while the sip user info is in a mysql realtime database, will there be any conflicts over time? |
| 19:18.45 | Goldenear | (I mean with IAX) |
| 19:18.46 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 19:19.24 | Goldenear | (I know this is possible with SIP) |
| 19:19.33 | Goshen | Goldenear: yes |
| 19:19.36 | *** join/#asterisk fje (~fje@gabby.fullnet.com) |
| 19:19.38 | Goshen | check this out... |
| 19:19.49 | Goshen | Goldenear: http://www.e164.org/ |
| 19:20.23 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) |
| 19:20.29 | Goshen | you put your info in there...and when asterisk does a look up it sends the call over internet direct to the computer rather then going through the whole voip/PTSN/voip |
| 19:21.02 | Goshen | Asterisk can also do call briding, where it makes the two connect to eachother and Asterisk doesn't carry the call anymore(I don't know much about that yet) |
| 19:21.09 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 19:21.47 | eKo1 | Yeah, just say canreinvite=yes in the sip.conf entry on both end-points. |
| 19:21.56 | eKo1 | Assuming they use sip of course. |
| 19:22.34 | Goshen | eKo1: same with IAX right? |
| 19:22.39 | Goldenear | I have yet a e164.org occount, but I currently use it with SIP... but you are so happy with IAX, I think I will change for this :) |
| 19:23.06 | eKo1 | Goshen: Not that I know of. |
| 19:23.21 | Goldenear | is there any wifi phone that is IAX compatible ? |
| 19:24.00 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 19:24.49 | Goshen | I only know of one wifi phone, that is kind of a new field |
| 19:25.19 | Goshen | there is the IAXy, which is a little box that lets you plug a phone into ethernet |
| 19:25.24 | mnet | i have a problem with an IAX link... it says No such context/extension whereas i'm sure the extension exists and the other server authenticates me ! |
| 19:26.26 | fje | does anyone know why this no longer works SetVar(tost=white|wheat|rye), it returns this error, WARNING[11127]: pbx.c:5352 pbx_builtin_setvar: Ignoring entry 'weat' with no = (and not last 'options' entry)? |
| 19:27.08 | Goldenear | Goshen: I saw this (and some other iax phones on iaxtel.com) :) |
| 19:27.17 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 19:27.22 | eKo1 | fje: Where is it getting 'weat' from? |
| 19:27.52 | anthm | \| on the extra | |
| 19:28.01 | Ayano | Is there still a bug patch for cisco auth, or did the new version of asterisk correct that? |
| 19:28.14 | *** join/#asterisk randu (~randy@pool-70-16-112-236.scr.east.verizon.net) |
| 19:28.24 | randu | Hello! |
| 19:28.29 | fje | eKo1: it's in an extenion context(extensions.conf) |
| 19:28.40 | randu | is there anyway to have a voicemail go to two email addresses? |
| 19:28.52 | Ayano | yes |
| 19:29.28 | randu | cool :-) Do you know how? |
| 19:29.37 | fje | eKo1: I just updated the to the latest CVS and this problem started... |
| 19:30.06 | eKo1 | fje: head or stable? |
| 19:30.19 | Ayano | I have seen it somewhere. I can look for it for you, but I have to go do a few things right now. Sory. |
| 19:30.22 | fje | eKo1: head |
| 19:30.36 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 19:30.49 | mogorman | anyone here work with suse? |
| 19:30.49 | eKo1 | fje: Don't complain if you're using head. It's for development only! |
| 19:32.40 | *** join/#asterisk Jackfiber (Jackfiber@213.217.52.184) |
| 19:32.59 | Jackfiber | hello anyone know a cheap SIP adaptor for analog phones |
| 19:33.15 | mogorman | supura |
| 19:33.21 | algorithmn | asterisk RealTime anyone? |
| 19:33.32 | fje | eKo1: I'm not complaining, just wondering if anyone knows if the syntax for this built in app has changed. |
| 19:33.45 | Jackfiber | Supura? |
| 19:33.45 | JerJer[mobile] | algorithmn: no |
| 19:33.47 | anthm | \| on the extra | |
| 19:34.04 | anthm | fje---------^ |
| 19:34.08 | algorithmn | JerJer[mobile]: nufone w/no realtime? |
| 19:34.31 | Jackfiber | does anyone know any analog to SIP adaptor? |
| 19:34.40 | JerJer[mobile] | algorithmn: most certianly |
| 19:35.04 | algorithmn | JerJer[mobile]: do you dislike it? |
| 19:35.14 | bjohnson | other than networking function and * .. any other neat things you can do with a wrt54g? |
| 19:35.20 | *** join/#asterisk what-a-guy (~wayne@cpe-67-10-172-229.houston.res.rr.com) |
| 19:35.49 | __Sparks_ | Jackfiber - http://www.grandstream.com/y-286.htm |
| 19:35.51 | jsolares | anything you can do with a regular linux and can fit on the small memory of the wrt54g |
| 19:35.54 | Inv_arp | Jackfiber: sipura , handytones |
| 19:36.05 | mogorman | sipura |
| 19:36.11 | mogorman | grandstream bad... |
| 19:36.12 | Jackfiber | Thanks |
| 19:36.19 | what-a-guy | First time here...hope to get help with first asterisk setup... |
| 19:36.36 | marshall | do many people change the echo cancellation aglorithims? |
| 19:37.08 | JerJer[mobile] | algorithmn: more like hate it |
| 19:37.25 | Goshen | jackfiber: get one of these, go IAX http://voipstore.pulver.com/product_info.php?cPath=21&products_id=52 |
| 19:37.27 | Crad|Work | is it possible to assign an account code to an extension? |
| 19:37.28 | algorithmn | JerJer[mobile]: i've been tweakin with it.. i need more efficiency for cdr/call settings that change regularly |
| 19:37.35 | Inv_arp | mogorman: i use grandstream HT486 works fine |
| 19:37.43 | JerJer[mobile] | algorithmn: so then change them |
| 19:38.01 | JerJer[mobile] | no need to force asterisk to depend on a database |
| 19:38.02 | *** join/#asterisk lancey (Shady@support.net1.cc) |
| 19:38.04 | lancey | hi guys |
| 19:38.08 | algorithmn | migration will be a pain, i just forsee some quirky problems |
| 19:38.12 | Goshen | what-a-guy: welcome :) |
| 19:38.14 | lancey | what do u recommend to use for * installation of FreeBSD |
| 19:38.20 | lancey | cvs, cvs stable? |
| 19:38.46 | what-a-guy | Goshen: Getting poor sound quality with x100p OEM card. Will I do better with Digium card? |
| 19:39.24 | *** join/#asterisk Lagaffe (~mbozio@www.lagaffe.org) |
| 19:39.31 | algorithmn | JerJer[mobile]: i hang reload processes all day long w/o a db system |
| 19:39.47 | Goshen | what-a-guy: need more info, are you using a sip phone to connect to the server then dialing out over the x100p or what? |
| 19:39.51 | JerJer[mobile] | hang? |
| 19:39.57 | algorithmn | the process never ends |
| 19:39.59 | Goshen | just dialing into the asterisk server from a landline? |
| 19:40.13 | algorithmn | i need to kill the asterisk process n safe_asterisk reloads it for me |
| 19:40.30 | algorithmn | and after a while zaptel crashes... |
| 19:40.41 | algorithmn | dtmf goes haywire... |
| 19:40.55 | algorithmn | echo off the wall... |
| 19:41.00 | what-a-guy | Goshen: I have a Grandstream 101 which does fine. Get poor sound when dialing in through POTS. Sounds like a cheap answering machine. |
| 19:41.29 | Lagaffe | hi there, any1 could help with some sip trouble ? can't make it to properly register with my primus sip account.. |
| 19:43.24 | *** join/#asterisk jontow (~jontow@ws.woflsys.net) |
| 19:43.28 | modulus_ | primus? |
| 19:43.36 | Goshen | what-a-guy: try my config....http://pastebin.ca/6736 |
| 19:43.45 | jontow | hello.. been a while ;) |
| 19:44.15 | what-a-guy | Goshen: thanks.. |
| 19:44.18 | *** part/#asterisk what-a-guy (~wayne@cpe-67-10-172-229.houston.res.rr.com) |
| 19:44.30 | *** join/#asterisk BrianR___ (brianr@h006067091a61.ne.client2.attbi.com) |
| 19:44.43 | BrianR___ | Is there any way to tell from the CLI if two IAX2 channels are trunked together? |
| 19:45.03 | loud | iax2 show registry |
| 19:45.04 | jontow | in the manager API, say I was to grab the output of "Action: Queues" .. C:N, A:N, SL:N.N% within 0s |
| 19:45.06 | eKo1 | show channels |
| 19:45.44 | jontow | C:N is the total number of calls (?), A:N is the abandonment number (?), and SL:N.N% is the "Service Level" (?) .. 0s being the the total call time? .. just looking for confirmation :) |
| 19:47.40 | BrianR___ | loud: WHat is shown when there's trunking going on? |
| 19:48.00 | loud | iax2 show channels |
| 19:48.04 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 19:48.11 | *** join/#asterisk FuriousGeorge (~FuriousGe@ool-43516ebb.dyn.optonline.net) |
| 19:48.25 | FuriousGeorge | _vile: do you copy |
| 19:48.33 | BrianR___ | loud: I have two calls going on and see two lines there. Does that mean I'm trunking or not? |
| 19:49.23 | loud | yes |
| 19:49.41 | FuriousGeorge | does anybody know how well asterisk works, if at all, with a software softphone answering the calls |
| 19:49.59 | eKo1 | BrianR___: You're better of using 'show channels' |
| 19:50.31 | FuriousGeorge | inother words, can i use a softphone with asterisk to answer an analog telephone lines phone call? |
| 19:50.38 | lancey | FuriousGeorge yes you can |
| 19:50.41 | lancey | i'm using it |
| 19:50.42 | FuriousGeorge | sweet |
| 19:50.44 | lancey | no problems |
| 19:50.46 | FuriousGeorge | thanks lacey |
| 19:50.49 | FuriousGeorge | later |
| 19:50.53 | lancey | *lancey |
| 19:50.54 | lancey | :) |
| 19:50.58 | FuriousGeorge | d'oh |
| 19:51.00 | eKo1 | FuriousGeorge: * doesn't give a dime what phone you use... |
| 19:51.07 | FuriousGeorge | lancy* |
| 19:51.22 | BrianR___ | eKo1: I see two lines in show channels also... Does that mean trunked or not? |
| 19:51.28 | FuriousGeorge | eKo1, i like the sound of this * more and more |
| 19:51.29 | jontow | furiousgeorge; in fact.. i just got 3 FXO cards to use for that purpose :) |
| 19:51.39 | eKo1 | BrianR___: If they're together, i.e. on atop the other, then yes. |
| 19:51.53 | FuriousGeorge | real cool, im gonna go tell my boss were getting an asterisk based pbx |
| 19:51.54 | *** join/#asterisk Lagaffe (~mbozio@www.lagaffe.org) |
| 19:52.03 | BrianR___ | eKo1: I only have two calls... |
| 19:52.07 | jontow | in practice, if you only have 1 phone line.. you only need 1 FXO card, but I have three locations with one line each :) |
| 19:52.09 | FuriousGeorge | later all |
| 19:52.21 | *** part/#asterisk FuriousGeorge (~FuriousGe@ool-43516ebb.dyn.optonline.net) |
| 19:52.35 | fje | anthm: I tried using \| an get the same error.. |
| 19:52.38 | eKo1 | Is it just me or is the output of 'show channels' just plain annoying? |
| 19:52.51 | anthm | setvar takes mytiple sets now |
| 19:52.54 | Lagaffe | sorry, got dropped, would any1 help me with some sip configuration ? can't register my asterisk to another sip box... |
| 19:53.04 | anthm | setvar(name=fred|age=22) |
| 19:55.02 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 19:55.51 | Lagaffe | any1 have an idear why i get 101 ACK when asterisk get an incoming call from another sip ? |
| 19:57.35 | slePP | meh |
| 19:57.39 | slePP | asterisk's sip stuff annoys me |
| 19:58.52 | zipp | how so? |
| 19:58.52 | Inv_arp | anywhere i can find a prerecorded line diconnected sound file? |
| 19:58.52 | slePP | the whole auth scheme of asterisk is convoluted |
| 19:58.52 | nestAr | Inv_arp: there's not one in asterisk-sounds? |
| 19:58.56 | slePP | granted, SIP's auth is convoluted |
| 19:59.05 | slePP | but asterisk doesn't make it clear what exactly it uses to look something up |
| 19:59.05 | nestAr | : /usr/src/asterisk-sounds/sounds/discon-or-out-of-service.gsm |
| 19:59.32 | zipp | Inv_arp: find / -name *.gsm |
| 19:59.52 | Inv_arp | nestAr: nice thx |
| 20:00.02 | *** join/#asterisk aiser (~chatzilla@host85-158.pool8254.interbusiness.it) |
| 20:00.04 | eKo1 | slePP: An example? |
| 20:00.32 | aiser | Hi to all |
| 20:00.34 | slePP | of what it doesn't do? |
| 20:00.36 | slePP | SER -> Asterisk |
| 20:01.36 | eKo1 | slePP: eh, just register the extensions/user in sip.conf. |
| 20:01.44 | slePP | you'd think so |
| 20:01.57 | JerJer[mobile] | techncally a peer gets the registration |
| 20:01.57 | eKo1 | I know so, I use SER & *. |
| 20:02.11 | eKo1 | You have to make * register with the SER. |
| 20:05.22 | eKo1 | Geez, my screen is getting flooded with NOTICEs. |
| 20:05.37 | marshall | does anyone know of a ata device that supports echo cancellation? |
| 20:05.40 | slePP | and asterisk doesn't seem to register to it. mm |
| 20:07.58 | *** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 20:08.00 | slePP | SER returns a not implemented.. |
| 20:09.48 | slePP | weird |
| 20:11.11 | *** join/#asterisk lyroy (~lyroy@picachou.csaffluents.qc.ca) |
| 20:11.25 | lyroy | Is there someone who ever use Trabas for Voip Billing? |
| 20:12.33 | Darwin35 | ok this is cool |
| 20:12.34 | Darwin35 | Darwin35 Digium cal;led me about fixing the g729 on fbsd |
| 20:12.48 | Darwin35 | first time a company has called me |
| 20:12.54 | Darwin35 | was nice call to |
| 20:12.56 | eKo1 | Transmitting (no NAT):g <--- What in the world does the 'g' there mean? |
| 20:14.06 | eKo1 | Hmm...could this be a funky bug? |
| 20:14.29 | slePP | i think it printed overitself |
| 20:15.53 | Inv_arp | nah no disc sounds in * ... heh i need that "your number ##### has been disc sound" |
| 20:16.12 | tzanger | Inv_arp: heh |
| 20:16.23 | tzanger | on a PRI just don't have an exten => line that matches the DID and the telco will do it |
| 20:16.32 | tzanger | Inv_arp: actually * does have that noise |
| 20:17.09 | zipp | doesn't zapateller do something like that? |
| 20:17.11 | tzanger | it's called playtones or even zapateller if it's Zap |
| 20:17.20 | BrianR___ | The Special Information Tone (SIT)? |
| 20:17.37 | *** part/#asterisk TSCHAK (tschak@cuodan.net) |
| 20:17.54 | Inv_arp | tzanger: hmm yea got a phpagi script that will play that sound if a certain numbers call... tryin to escape my ex :) |
| 20:18.07 | BrianR___ | There's a disconnect cause which will cause the number not in service to be played by the calling party's CO too. |
| 20:18.09 | tzanger | Inv_arp: hahah |
| 20:19.32 | HitTop | hi |
| 20:20.21 | HitTop | if i use the command NoCDR(), i received this warning: WARNING[5357]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/100-f3a5' not posted. WARNING[5357]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/100-f3a5' lacks end. |
| 20:20.21 | Moc____ | hi all |
| 20:20.24 | HitTop | is this common? |
| 20:21.20 | Inv_arp | BrianR___: oh yea where can i get more info on that? |
| 20:22.06 | buddah | anyone use nufone for international termination? |
| 20:22.21 | Darwin35 | what is up with broadvoice |
| 20:22.40 | Darwin35 | 3 weeks now I have been emailing and calling and getting no reponce |
| 20:22.56 | Inv_arp | Darwin35: what probu having? |
| 20:22.58 | zipp | buddah, nufone to call from us to other nations, or the other way? |
| 20:23.10 | Darwin35 | call waiting not working |
| 20:23.20 | buddah | from us to other nations |
| 20:23.40 | zipp | buddah, I call people in germany all the time through nufone |
| 20:23.42 | Darwin35 | 3way not working |
| 20:23.49 | buddah | looking to find a new international (US to other countries) carrier since the 2 we use keep having routing problems |
| 20:23.56 | buddah | and was thinking about testing nufone |
| 20:24.09 | zipp | why not, for $5 you can test plenty |
| 20:24.11 | buddah | zipp: how is the service, reliable? quality? |
| 20:24.17 | zipp | nufone has been great for me |
| 20:24.21 | buddah | good |
| 20:24.22 | zipp | _great_ |
| 20:24.36 | Inv_arp | Darwin35: hmm works for me |
| 20:24.44 | buddah | like 3 major routes we use, singapore, banghledesh, and pakistan |
| 20:24.45 | Darwin35 | not working for me |
| 20:24.49 | buddah | both carriers are not working |
| 20:24.54 | buddah | so we are kinda in a tight spot |
| 20:25.16 | Darwin35 | also having issues with not hearing a ring when you call some one |
| 20:25.16 | zipp | buddah, I have never personally called those locations |
| 20:25.20 | buddah | yeah |
| 20:25.24 | BrianR___ | Inv_arp: Look in the voip-info wiki under Hangup |
| 20:25.26 | buddah | well if it works, then thats a huge advance |
| 20:25.27 | buddah | heh |
| 20:25.33 | zipp | however, I call germany, mexico, cananda... |
| 20:25.37 | Darwin35 | getting dead air and then 15 20 sec ltr a voice or a busy tone |
| 20:26.00 | tzanger | Darwin35: zap interface? |
| 20:26.15 | tzanger | do you have callwaiting=yes and threewaycalling=yes (or whatever it's called) in zapata.conf? |
| 20:26.23 | buddah | canada works good? |
| 20:26.44 | Darwin35 | not using asterisk on this line |
| 20:26.56 | Darwin35 | its a direct phone to broadvoice line |
| 20:26.58 | jsolares | buddah: look at livevoip.com, they have good rates |
| 20:29.55 | *** join/#asterisk Darkar (~alex@m174.net81-66-29.noos.fr) |
| 20:33.14 | *** join/#asterisk jgaviria (~jgaviria@63.245.86.120) |
| 20:36.14 | jgaviria | hi, i want to do my iax connection using a different codec not just gsm, but whe i modify iax.conf with another codec, it doesnt work and i still have gsm working |
| 20:37.08 | JonR800 | jgaviria: did you reload/restart? |
| 20:38.01 | jgaviria | JonR800 i just do restart |
| 20:38.54 | JonR800 | what does your disallow/allow setup look like? |
| 20:39.48 | jgaviria | disallow=all |
| 20:39.51 | jgaviria | allow=gsm |
| 20:39.57 | jgaviria | sorry |
| 20:40.00 | jgaviria | again |
| 20:40.06 | jgaviria | disallow=all |
| 20:40.06 | jgaviria | allow=ilbc |
| 20:40.11 | jgaviria | allow=gsm |
| 20:40.22 | jgaviria | the first priority is ilbc and it doesnt work |
| 20:40.28 | JonR800 | does the other end support ilbc? |
| 20:40.45 | jontow | wow.. that flash operator panel is damn spiffy :) |
| 20:40.48 | *** part/#asterisk nikko_ (~nikko@69.85.201.170) |
| 20:41.01 | JonR800 | sorry if these are all fairly obvious questions :) |
| 20:41.12 | jontow | (http://www.asternic.org/) |
| 20:41.16 | JonR800 | jontow: yeah the new version is sweet |
| 20:41.35 | jontow | it seems not highly polished with functionality, but the interface and configurability is.. *wow* |
| 20:42.30 | anthm | jgaviria, is it asterisk on both ends and is it HEAD? |
| 20:42.33 | jontow | ie. the parsing of the queue data and whatnot is very .. well, as-is ;) when thats polished up a bit, i have a feeling that is a VERY useful product in a call-center environment.. more so than now, even :) |
| 20:43.19 | jgaviria | anthm: yes is in both ends, what do you mean with HEAD? |
| 20:43.27 | anthm | newest cvs |
| 20:43.40 | anthm | or at least the last few weeks |
| 20:43.58 | jgaviria | anthm: 1.03 |
| 20:44.49 | anthm | same old same old |
| 20:44.50 | Juggie | anthm, what would u think about cdr using realtime/extconfig to write to the database instead of its own support? |
| 20:45.52 | anthm | Juggie, would make sense |
| 20:46.28 | Juggie | i was poking through the code but its going to require some core functionality changes and i'm not in any way famaliar with this code |
| 20:47.02 | anthm | jgaviria, upgrade to new CVS and you can control the codecs |
| 20:47.37 | *** join/#asterisk shell (shell@200.66.58.155) |
| 20:47.58 | jgaviria | anthm: are you sure?... 1.03 had this problem? |
| 20:48.07 | anthm | Juggie, also take in to consideration my new cdr vars addition |
| 20:48.09 | Darwin35 | man still noone at broadvoice |
| 20:48.15 | Juggie | anthm, i already did. |
| 20:48.17 | Darwin35 | this is bullshit |
| 20:48.28 | Juggie | i forse one cdr module, or two at most. |
| 20:48.31 | anthm | well i added support for codec preferencing myself |
| 20:48.33 | Darwin35 | they should havesome one there if they offer tech support |
| 20:48.35 | Juggie | cdr_db and cdr_csv |
| 20:49.29 | anthm | if it has that patch, you can say codecpriority=caller in the friend on the server |
| 20:49.37 | Juggie | cdr_db should use extconfig/realtime and do the default insert first, then do updates to add in the custom CDR vars. |
| 20:49.47 | anthm | and the codecs are honored in the order they are defined |
| 20:49.48 | sivana | anyone know if notransfer=yes can go under [general] and affect everyone? |
| 20:50.11 | Juggie | i dont think it would be wise to do it all on the same line anthm as if someone fools up their dialplan and tries to use an invalid field, then they will looose the record alltogether |
| 20:50.23 | Juggie | so initial insert then record updates i would think would be best. |
| 20:50.26 | *** join/#asterisk Ayano (~erik_leee@209.143.187.254) |
| 20:50.48 | jgaviria | anthm: ok thanks! |
| 20:51.04 | Ayano | I have a cisco phone that I'm testing and it wont authenticate. Where can I find the patch that corrects this? |
| 20:52.14 | anthm | probably you would need to specifg what cols to try and send to th engine |
| 20:52.30 | anthm | so it only used those and it would match the table you are using |
| 20:53.28 | sivana | Darwin35: what's wrong with BV |
| 20:53.45 | Juggie | anthm, you already did it |
| 20:53.59 | Juggie | SetVarCDR(var=value) |
| 20:54.04 | Juggie | so treat var as a column name |
| 20:54.35 | anthm | yah that got yanked the final cut was SetVar(CDR(var)=value) |
| 20:54.47 | Juggie | well, still.... |
| 20:54.49 | Juggie | do that |
| 20:54.53 | Juggie | so if u did |
| 20:54.57 | anthm | i mean since anyone can add more cols at will |
| 20:55.02 | Juggie | right... |
| 20:55.09 | Juggie | they need to put it in their db |
| 20:55.13 | Juggie | thats not our problem |
| 20:55.18 | anthm | you need to pick ahead of time which ones will be used to insert into the db |
| 20:55.28 | anthm | so you can make sure it only uses the ones you have |
| 20:55.53 | Corydon-w | Not necessarily... if you had an excessively normallized schema |
| 20:56.01 | Juggie | i dont agree... |
| 20:56.11 | Juggie | why not do the standard insert as usual, as is done now |
| 20:56.18 | Juggie | and then run an update for the custom vars. |
| 20:56.42 | Juggie | its an extra hit on the database, but least if the user asses it up, they will have most of the record |
| 20:56.44 | anthm | you mean so it can fail if they dont exist ? |
| 20:57.19 | anthm | or you could extend realtime to tell you all the cols you have to work with |
| 20:57.21 | Juggie | its going to fail if the column doesnt exist iregardless |
| 20:57.28 | anthm | send back as a string of ast_vars |
| 20:58.00 | Corydon-w | The other problem is that the varname has to exactly match the column name... so what happens if you have multiple CDR modules loaded, with different types of fields in each with the same name? |
| 20:58.32 | Juggie | anthm, yah, realtime should keep a list of fields in every table its pointed to |
| 20:58.35 | Juggie | that would be good. |
| 20:58.48 | Juggie | then you can fail on the set in the dialplan |
| 20:58.52 | Juggie | if the var doesnt exist. |
| 20:58.54 | anthm | maybe like struct_ast_variable var = ast_realtime_get_cols(family,db); |
| 20:58.58 | *** part/#asterisk Moc____ (~mochouina@64.235.210.66) |
| 20:59.02 | Corydon-w | Juggie: what if a table gets modified on the fly? |
| 20:59.18 | Corydon-w | Juggie: should it just ignore that new column until a reload? |
| 20:59.21 | Juggie | Corydon-w, its not possible to prevent every scenario |
| 20:59.27 | Juggie | i think it should yes. |
| 20:59.38 | Juggie | modify your tables, u have to reinit realtime so it can get table structs again |
| 20:59.43 | Corydon-w | But that's not consistent with the nature of RealTime |
| 21:00.06 | Juggie | sure it is, no where in realtime does it say we should modify tables while its live. |
| 21:00.09 | afe | anyone knows how high I can set RING_DEBOUNCE in wctdm.c without asterisk going haywire on me? I now have 128ms, but that's sometimes not enough |
| 21:00.29 | Corydon-w | No, but the nature of RealTime is that it gets rows as it needs them |
| 21:00.29 | afe | And when I tried with 256 asterisk wouldn't start properly |
| 21:00.54 | *** join/#asterisk Matt-E- (~Matto@66-224-125-137.atgi.net) |
| 21:00.55 | Juggie | maybe add a realtime command to reparse table structs. |
| 21:01.11 | *** join/#asterisk outsidefactor (~blah@203-173-32-225.dyn.iinet.net.au) |
| 21:01.39 | Corydon-w | Juggie: why not just put the fields that you need into cdr_customwhatever.c and leave it at that? |
| 21:02.00 | Corydon-w | Or go for the normalized layout of separate rows for each custom variable? |
| 21:02.04 | Juggie | Corydon-w, because theres already a patch on head for custom CDR variables. |
| 21:02.27 | *** join/#asterisk djrzulf (13-2355@82.160.40.3) |
| 21:02.33 | Corydon-w | Juggie: custom CDR variables are how to notify your custom CDR module of the values |
| 21:02.45 | djrzulf | hello |
| 21:02.52 | djrzulf | any polish people here ? ;] |
| 21:03.00 | Corydon-w | Juggie: but the matter of the backend database; how it should deal with those set variables is an entirely different matter |
| 21:03.19 | Juggie | Corydon-w, i am trying to envision a system where you dont need custom modules. |
| 21:03.42 | Corydon-w | Juggie: okay, so go with the normalized database schema, in that case |
| 21:04.11 | Corydon-w | That way, you always store the variables set, even if the main table isn't set to receive them |
| 21:04.15 | Juggie | thats an option, but then thats a hassle to relink those tables to your cdr record, it should just be one database. |
| 21:04.32 | Juggie | er, one table |
| 21:04.32 | Juggie | i ment |
| 21:04.39 | Corydon-w | It's not a hassle... it's called a relational database |
| 21:04.47 | Juggie | o |
| 21:05.10 | Juggie | *i'm famailiar with a rational database design thanks, i just dont consider it an advantage in this situation |
| 21:05.33 | Qwell | y0, y0, y0 |
| 21:05.43 | djrzulf | ;) |
| 21:05.57 | Juggie | i agree with what anthm said, maintain a list of columns within the table and allow those to be changed. |
| 21:06.13 | eKo1 | What are you guys talking about? |
| 21:06.23 | Qwell | exten => s,1,Playback(Qwell-is-now-a-daddy--w00t) |
| 21:06.26 | Juggie | fail on any SetVar(CDR(var)=val) which includes one not available. |
| 21:06.28 | tuxinator_linux | kram: Hey Mark |
| 21:06.29 | Corydon-w | eKo1: storing extra CDR fields |
| 21:06.33 | anthm | i like to write my cdr 1 file per record with the vars hashed out |
| 21:06.33 | kram | hi tux |
| 21:06.40 | eKo1 | Corydon-w: Like a rate field? |
| 21:06.48 | Juggie | see, i have backup :) |
| 21:06.51 | Corydon-w | eKo1: exactly |
| 21:07.05 | tuxinator_linux | kram: How is VON prep going? I will be at Meet *. |
| 21:07.06 | Juggie | cdr has no business being spread across multiple tables, with a one to many relationship. |
| 21:07.18 | kram | tux: cool, it's been busy, but then when isn't it |
| 21:07.29 | eKo1 | Huh? CDRs across multiple tables? |
| 21:07.41 | Corydon-w | Juggie: sure it does, but it depends upon what you want to have stored |
| 21:07.46 | Juggie | that was Corydon-w's suggestion to dealing with custom cdr vars. |
| 21:08.14 | eKo1 | I would just store the extra 'vars' in columns in the cdr table. |
| 21:08.15 | Corydon-w | Juggie: No, my suggestion is to have custom cdr_*.c modules, per backend accounting package |
| 21:08.42 | shmaltz | anybody interested in looking at this: |
| 21:08.44 | shmaltz | http://lists.digium.com/pipermail/asterisk-users/2005-February/092049.html |
| 21:08.46 | Juggie | Corydon-w, i respectfully disagree, users should be able to modify their cdr table to add the fields they want.... |
| 21:09.00 | *** join/#asterisk Red_6 (~alex@m174.net81-66-29.noos.fr) |
| 21:09.02 | Juggie | then SetVar should set values in those fields, if they exist. |
| 21:09.15 | jontow | nice.. linux-mozilla/flashplugin works with the asternic.org flash stuff :) |
| 21:09.51 | eKo1 | I agree with Juggie. |
| 21:10.28 | Juggie | 3 out of 4 dentists agree. |
| 21:10.29 | eKo1 | Or they can be filled elsewhere, e.g. with a stored procedure. |
| 21:10.40 | Corydon-w | Gee, why don't we describe the database schema with XML? |
| 21:10.58 | Corydon-w | Then parse that schema inside cdr_whatever.c |
| 21:11.15 | Juggie | because the ideal design is to reduce the number of required modules |
| 21:11.41 | Juggie | there should just be one cdr module, or at most, cdr_db & cdr_csv |
| 21:12.19 | Corydon-w | Uh, that's a little much |
| 21:12.22 | Juggie | _db idealy then uses the database connectivity provided by realtime |
| 21:12.27 | Juggie | and works using extconfig |
| 21:12.47 | Juggie | i dont see why... database driver code shoudnt be rewritten all over the place. |
| 21:13.19 | Juggie | realtime only supports reads and updates atm so it would need to be updated to create new records. |
| 21:13.19 | Inv_arp | hmm anyone know the disc cause code for number disconnected usa and using BV for incoming |
| 21:13.25 | Corydon-w | I think we should have cdr_customaccountingpackage34.c and people interested in using that package should load that CDR module |
| 21:13.26 | anthm | like i said i like the serialized data dump way here is an example of the way I do it |
| 21:13.30 | anthm | http://www.asterlink.com/eg/105902.1109797744.0 |
| 21:14.06 | Juggie | server's down |
| 21:14.11 | Corydon-w | That makes it simpler for the person deploying the package |
| 21:14.16 | anthm | i didnt conver to use my own patch for cdr vars yet but i use channel vars for the time being |
| 21:15.02 | anthm | server may not like you it's pretty paranoid |
| 21:15.21 | Juggie | Corydon-w, thats fine if your needs are specific, but why should someone who wants one extra field be required to write a custom module |
| 21:16.28 | Juggie | i needed one extra field to tell me which server the record came from, i had to patch cdr_mysql, it shoudnt be that hard. |
| 21:17.04 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-224-247.dsl.scarlet.be) |
| 21:17.35 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-219-154.dsl.scarlet.be) |
| 21:18.34 | Juggie | why shoudn't i be able to do a SetVar(cdr(servernum)=1) in my dialplan, add servernum as a field in the table, and i'm done... |
| 21:18.59 | anthm | how you gonna do that ? |
| 21:19.14 | *** join/#asterisk hotoke (~hnic@208.179.86.5) |
| 21:19.42 | HitTop | anyone using asterisk's sound card for intercom? |
| 21:20.46 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 21:21.22 | Juggie | what do u mean how? in the case of right now, ignoring the fact i think cdr should use realtime, you would need to patch cdr_addon_mysql to loop through the set cdr(var)'s if any exist, and put them into the cdr table, provided the fields exist within the database. |
| 21:21.25 | *** join/#asterisk bowman (~rsp@195.46.47.202.static.cablesurf.de) |
| 21:21.31 | bowman | hi. any quadBRI users here? |
| 21:21.35 | zigman | me |
| 21:21.50 | *** join/#asterisk afe_ ([SDgLXkygd@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 21:22.08 | bowman | I have audio "holes" on the ISDN side, any idea how to get rid of them? |
| 21:22.19 | zigman | yes |
| 21:22.23 | zigman | turn dma on |
| 21:22.37 | bowman | for what? for the card? |
| 21:22.38 | afe_ | this is very off topic, but why did my username just get an underscore after it? |
| 21:22.56 | Juggie | you likely reconnected and your previous client hadnt dropped? |
| 21:22.58 | Goshen | type /nick afe |
| 21:22.59 | buddah | 'do a /whois afe |
| 21:23.05 | buddah | and youll see why |
| 21:23.11 | bowman | afe_: because there is someone with the nick afe online and your client chose afe_ as the alternative ;) |
| 21:23.48 | Juggie | anthm? |
| 21:24.02 | anthm | ? |
| 21:24.08 | afe_ | ah... thanks - my putty crashed and I thought I killed all processes :) |
| 21:24.14 | bowman | zigman: I only know how to switch DMA mode for hard disks - how about PCI cards? |
| 21:24.15 | Juggie | oh, agree/disagree? |
| 21:25.02 | zigman | no for your harddisk |
| 21:25.10 | zigman | do hdparm -d /dev/hda |
| 21:25.13 | zigman | is it on ? |
| 21:25.24 | bowman | yep |
| 21:25.26 | afe | :P |
| 21:25.37 | *** join/#asterisk zoa (~zoa@ip-212-239-162-26.dsl.scarlet.be) |
| 21:25.44 | zoa | yo yo |
| 21:26.21 | afe | zoa can you reset my mantis pass ? |
| 21:26.31 | zoa | i will have a look in a sec afe |
| 21:26.34 | afe | i'm going ;) |
| 21:26.46 | zigman | hehe no thats my pass |
| 21:26.49 | zigman | my user ;) |
| 21:26.55 | *** join/#asterisk afe_ ([OQ48nGiB0@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 21:27.06 | zigman | you see why ? ;) |
| 21:27.14 | afe_ | lol |
| 21:27.30 | afe_ | gimme my nick back :) |
| 21:27.31 | *** join/#asterisk mishehu (mishehu@cshells.shavedgoats.net) |
| 21:27.36 | zigman | ;) |
| 21:28.00 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 21:28.02 | fje | anthm: thanks for you help, I was able to fix my problem by modifying my agi scripts, and using a different delimiter |
| 21:28.11 | afe_ | I promise to change the pw again |
| 21:28.13 | HitTop | hi all |
| 21:28.32 | kram | n |
| 21:28.36 | goatmilk | y |
| 21:28.42 | eKo1 | c |
| 21:28.58 | anthm | fje, np |
| 21:29.00 | goatmilk | eKo1: :) |
| 21:29.18 | tuxinator_linux | I feel like eating some alphebet soup |
| 21:29.25 | *** join/#asterisk afe ([7k2BDR3Lj@c-e616e055.123-1-64736c12.cust.bredbandsbolaget.se) |
| 21:29.32 | HitTop | i've received a warning when loading chan_oss.so, I cannot use this channel for paging right now, can anyone help me please??? |
| 21:29.57 | afe | ah... now it works again (sorry about that I haven't used irc in ages :)) |
| 21:31.24 | *** join/#asterisk LarsAC (~chatzilla@pD95005F0.dip0.t-ipconnect.de) |
| 21:31.56 | ManxPower | Looks like another newbie. Doesn't paste the actual error message.... |
| 21:32.52 | *** join/#asterisk phantam (~phantam@72.252.15.235) |
| 21:33.10 | phantam | hmmm |
| 21:33.12 | HitTop | WARNING[3821]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable |
| 21:33.19 | phantam | still cant get this darn oh323 to connect |
| 21:33.22 | phantam | same error everytime |
| 21:33.29 | HitTop | sorry. i was afraid to spam this channel |
| 21:34.16 | *** part/#asterisk fje (~fje@gabby.fullnet.com) |
| 21:34.40 | ManxPower | Himeko, That means either you don't have the oss drivers loaded or some other application has that device in use or you don't have permission to open the device. |
| 21:34.52 | ManxPower | Himeko, Can you use the sound card in other applications? |
| 21:35.11 | phantam | wtf -- H.323 call 'ip$localhost/24978' cleared, reason 24 (Call ended with Q.931 cause) |
| 21:35.17 | phantam | every time |
| 21:35.18 | phantam | argggggg |
| 21:35.33 | ManxPower | Well find out what the cause is. |
| 21:35.39 | ManxPower | Usually ${CAUSECODE} |
| 21:36.35 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 21:37.05 | phantam | huh |
| 21:37.15 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 21:37.39 | ariel_ | afternoon everyone. |
| 21:37.48 | tuxinator_linux | Hey ariel_ |
| 21:38.37 | phantam | how am i supposed to do that |
| 21:38.55 | phantam | ariel_: sup im rippin my h323 hair out |
| 21:38.56 | *** join/#asterisk santiago (~santiago@63.245.86.120) |
| 21:39.35 | phantam | no one knows what that error means? |
| 21:40.19 | eKo1 | Maybe looking at the source code will help. |
| 21:40.35 | zigman | kram |
| 21:40.38 | zigman | damn |
| 21:40.44 | zigman | MANTIS |
| 21:40.50 | zigman | i want my mantis back ! ;) |
| 21:41.09 | drumkilla | ? |
| 21:41.13 | eKo1 | You mean the bug tracker is down? |
| 21:41.14 | zigman | my pass |
| 21:41.17 | zigman | it was reset |
| 21:41.23 | zigman | but i never received the mail for it |
| 21:41.24 | eKo1 | oh... |
| 21:41.33 | phantam | drumkilla = op maybe he knows |
| 21:41.34 | phantam | :) |
| 21:41.56 | drumkilla | zigman: want me to reset it? |
| 21:42.07 | zigman | yes pls |
| 21:42.09 | zigman | user is zigman |
| 21:42.20 | zoa | mantis is not down |
| 21:42.22 | zoa | i was on there |
| 21:42.29 | drumkilla | zigman: ok. You should get a new password by email |
| 21:43.00 | zigman | thx |
| 21:43.04 | drumkilla | no problem |
| 21:43.06 | zoa | zigman i cant fix it for you |
| 21:43.08 | zoa | aha |
| 21:43.12 | zoa | drumkila is already here |
| 21:43.13 | drumkilla | who wants to guess how many accounts there are? |
| 21:43.18 | zoa | me me me |
| 21:43.21 | drumkilla | :) |
| 21:43.21 | zigman | 3000 |
| 21:43.22 | zoa | 10.000 ? L |
| 21:43.24 | phantam | 3 |
| 21:43.29 | phantam | ;) |
| 21:43.39 | zoa | 500 |
| 21:43.46 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
| 21:43.48 | phantam | 0.2 |
| 21:43.48 | zoa | i just counted them rainman style |
| 21:43.49 | phantam | ? |
| 21:43.54 | zigman | how many are there? |
| 21:43.56 | drumkilla | .... 2138 |
| 21:43.57 | zoa | 534 |
| 21:44.06 | phantam | u're rain man skills are well off |
| 21:44.07 | phantam | lol |
| 21:44.10 | drumkilla | 34 new in the last week |
| 21:44.20 | zoa | thats 34 times bkw reapplying |
| 21:44.27 | phantam | drumkilla: do u know anyone using oh323 i cant get past this error |
| 21:44.41 | drumkilla | plenty of people use it ... |
| 21:44.45 | drumkilla | I don't, though :) |
| 21:44.50 | file | Russelllllllllllll |
| 21:44.52 | zigman | zoa ;) |
| 21:44.52 | phantam | any that might be around lol |
| 21:44.55 | drumkilla | file!!!! |
| 21:45.05 | drumkilla | phantam: this is the best place to ask ... |
| 21:45.07 | ctooley | Sometimes I wonder if Microsoft doesn't have it right. Some customers really do deserve it. |
| 21:45.08 | phantam | lol |
| 21:45.09 | file | Russelllllllllllll!!!!!!!!!!!!!!!!!!!!! |
| 21:45.11 | phantam | i been askin all day |
| 21:45.12 | drumkilla | you can also try the -users mailing list |
| 21:45.15 | phantam | and no one said nuttin |
| 21:45.22 | *** part/#asterisk djrzulf (13-2355@82.160.40.3) |
| 21:45.28 | drumkilla | phantam: have you searched the mailing list archives? |
| 21:45.31 | zoa | oh323 and h323 -> i dont think anyone will want to give you support for that |
| 21:45.34 | phantam | yes |
| 21:45.37 | phantam | i saw the question |
| 21:45.40 | phantam | but there was no answer |
| 21:45.44 | drumkilla | heh |
| 21:45.46 | JerJer[mobile] | not oh323 that's for sure |
| 21:45.58 | zoa | hehe jj is also awake |
| 21:46.01 | JerJer[mobile] | h323 is a different story |
| 21:46.05 | phantam | lol JerJer[mobile] well if h323 would compile i'd happily use it |
| 21:46.20 | phantam | but needless to say its bound to such anchient pwlibs it wont compile |
| 21:46.36 | drumkilla | are you running a stable release or cvs head? |
| 21:46.50 | phantam | stable |
| 21:46.52 | phantam | 6.5 |
| 21:46.59 | drumkilla | 6.5? |
| 21:47.01 | mogorman | who in there right mind would run stable drumkilla ^_^ |
| 21:47.02 | phantam | 0.6.5 |
| 21:47.03 | phantam | lol |
| 21:47.07 | drumkilla | woah |
| 21:47.09 | drumkilla | dude. |
| 21:47.10 | zoa | is anyone using app_icd ? |
| 21:47.10 | drumkilla | update |
| 21:47.12 | phantam | no |
| 21:47.14 | phantam | lol |
| 21:47.15 | drumkilla | haha |
| 21:47.15 | zoa | wow |
| 21:47.17 | phantam | u meant of asterisk |
| 21:47.17 | phantam | ? |
| 21:47.18 | zoa | 0.6.5 |
| 21:47.20 | zoa | amazing |
| 21:47.20 | drumkilla | yes |
| 21:47.23 | phantam | oh |
| 21:47.23 | phantam | lol |
| 21:47.25 | phantam | 1.0.5 |
| 21:47.26 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 21:47.28 | zoa | aha |
| 21:47.39 | drumkilla | ok, well for chan_h323, you'll want to try cvs head ... |
| 21:47.47 | drumkilla | the one in stable hasn't been touched in a long time |
| 21:47.55 | drumkilla | head has had a lot of improvements |
| 21:48.04 | phantam | ye i heard that before lol jerjer dont wanna touch it no more |
| 21:48.04 | phantam | lol |
| 21:48.07 | zoa | did anyone here try the latest chan_iax2 jitter buffer patches ? |
| 21:48.24 | zoa | i have a suspicion there are some deadlocks there |
| 21:48.39 | zoa | but only happening very seldomly |
| 21:49.16 | phantam | so cvs version of asterisk compiles clean with h323? |
| 21:49.25 | drumkilla | as far as I know |
| 21:49.38 | phantam | what version of pwlib etc do i need JerJer[mobile] |
| 21:49.49 | bowman | zigman, pm? |
| 21:50.13 | zigman | bowman not workinß |
| 21:50.34 | zigman | drumkilla can you find out to what email adress the password gets send ? |
| 21:50.40 | jsolares | meh i moved my quad fxo card from one pci slot to the other and now it's sharing an irq, and not working anymore |
| 21:50.43 | bowman | zigman: just to check out some details :) |
| 21:51.00 | jsolares | would irq sharing cause the card to no longer work? |
| 21:51.04 | jsolares | or maybe i fried it... :X |
| 21:51.24 | ManxPower | jsolares, Shareing IRQs can cause ANY sort of problem. |
| 21:51.28 | zigman | bowman KalD|Work |
| 21:51.31 | zigman | bowman k |
| 21:51.33 | jsolares | any huh? |
| 21:51.39 | jsolares | well crap |
| 21:51.52 | *** join/#asterisk Dalion (~DaLion@HSE-QuebecCity-ppp3497400.sympatico.ca) |
| 21:52.06 | ManxPower | Digium cards do not support shareing IRQs. |
| 21:52.54 | jsolares | i figured hisses, pops, crackles, or any sort of weird sounds, not it dying, hehe oh well, i think i'll move the network card which doesnt show up as sharing an irq |
| 21:53.40 | ManxPower | jsolares, Remember that if you plug a phone line into and FXS port on the card and a call comes in, it will blow up the module. |
| 21:53.57 | Goshen | I am looking for a voip provider that supports IAX, and can use number portability to transfer my number, and tips? |
| 21:54.37 | Goshen | nufone doesn't port numbers, and doesn't provider 801 area code numbers |
| 21:55.12 | ManxPower | Goshen, As far as I can tell, all VoIP providers suck in some way. |
| 21:55.22 | Goshen | hmm |
| 21:55.39 | harryvv | Manx #1 issue would be avaible bandwith? |
| 21:55.43 | Goshen | I guess for business use, it would be best to keep my PSTN line then |
| 21:55.52 | Goshen | for incoming anyhow |
| 21:55.58 | ManxPower | harryvv, For various reasons. Quality, Support, DID numbers, etc. |
| 21:56.14 | ManxPower | PLUS all the issues of the internet between you and the provider. |
| 21:56.20 | harryvv | You mean avaiable dids or the one you want :) |
| 21:56.35 | ManxPower | DIDs in cities I want them in. |
| 21:56.50 | ManxPower | or they require you to use their equipment. |
| 21:56.52 | harryvv | I was lucky to find a local did providers and going to inquire in there service. |
| 21:57.04 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 21:57.18 | ManxPower | Lingo has good prices, good DId coverage, cheap rates, but they require their own equipment. |
| 21:57.50 | JerJer[mobile] | Goshen: we port numbers |
| 21:57.52 | Goshen | if I go with a provider, I would require IAX |
| 21:57.56 | Goshen | who is we? |
| 21:58.03 | JerJer[mobile] | i own nufone |
| 21:58.18 | Goshen | ahh, you will want to change your voip wiki entry then :) |
| 21:58.22 | file | JerJer _is_ NuFone |
| 21:58.29 | dca[laptop] | Goshen: have you look at teliax? |
| 21:58.30 | eKo1 | hehe |
| 21:58.39 | Goshen | have not looked at teliax yet |
| 21:58.41 | file | he is 'da bomb |
| 21:58.47 | Goshen | jerjer: can you port 801 numbers? |
| 21:58.48 | twisted[work] | file |
| 21:58.48 | harryvv | JerJer[mobile] when did you start selling service to the public |
| 21:58.50 | mogorman | hey jerjer can i get any 1800 numbers? which ones can you get at your rate? |
| 21:58.56 | dca[laptop] | Goshen: http://www.teliax.com |
| 21:58.56 | file | twisted |
| 21:59.00 | twisted[work] | file |
| 21:59.04 | file | twisted @ work |
| 21:59.07 | Goshen | afk, patient |
| 21:59.19 | twisted[work] | file, ready for von/ |
| 21:59.23 | JerJer[mobile] | harryvv: um we have always sold service to the public |
| 21:59.27 | twisted[work] | s/\/? |
| 21:59.29 | twisted[work] | er |
| 21:59.29 | file | twisted[work]: well yesssssssss, actually no |
| 21:59.35 | twisted[work] | no? |
| 21:59.44 | harryvv | jerjer even before voip came along? |
| 21:59.51 | file | I'm going to the bank tomorrow to get some cash, and getting a haircut |
| 22:00.02 | twisted[work] | file, we can cut off your hair |
| 22:00.04 | JerJer[mobile] | Goshen: we only deal with DIDs in the state of Michigan and US48 Toll-Free numbers |
| 22:00.10 | file | twisted[work]: HA |
| 22:00.11 | twisted[work] | it'd be fun |
| 22:00.18 | twisted[work] | we'd sacrifice your hair to the VoIP gods |
| 22:00.38 | file | hrm lemme think |
| 22:00.39 | file | ... |
| 22:00.40 | file | NO! |
| 22:00.53 | drumkilla | you can cut mine |
| 22:01.06 | drumkilla | It's long |
| 22:01.08 | phantam | ARG |
| 22:01.09 | phantam | wtf |
| 22:01.14 | phantam | cant i get anything proper |
| 22:01.19 | JerJer[mobile] | this is why a wiki should not be the source for information |
| 22:01.25 | *** join/#asterisk darkskiez (~mhb@host-84-9-70-212.bulldogdsl.com) |
| 22:01.27 | file | drumkilla: if bkw was here he'd use that oppurtunity to so make fun of you |
| 22:01.38 | phantam | cvs is compiled against 1.8.1 thats not in portage so i'd have to manually do it... and the openh323 as well |
| 22:01.41 | drumkilla | file: can't make fun of people yourself? |
| 22:01.41 | JerJer[mobile] | whoever created those entries for us has no clue |
| 22:01.49 | JerJer[mobile] | lord knows I didn't do that crap |
| 22:01.57 | file | drumkilla: I don't wanna! |
| 22:02.12 | Dalion | i use teliax.. works incredibly well |
| 22:02.21 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net) |
| 22:02.25 | phantam | JerJer[mobile]: how come cvs isnt against the latest pwlib/oh323? |
| 22:02.57 | JerJer[mobile] | because nobody has informed me it works |
| 22:03.06 | Juggie | anthm, do you have an example cdr.conf for your cdr_csv2? i cant find one in the cvs. |
| 22:03.19 | *** join/#asterisk TSCHAK (tschak@cuodan.net) |
| 22:03.19 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 22:03.38 | *** join/#asterisk Dr-Linux (~sshah@202.125.141.6) |
| 22:03.53 | Dr-Linux | i need help |
| 22:03.58 | *** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net) |
| 22:04.04 | HitTop | i wonder if res_musichold.so can be unload |
| 22:04.06 | Dr-Linux | i'm using x-lite |
| 22:04.36 | jontow | dr-linux; pretty vague.. :) |
| 22:04.41 | Dr-Linux | my friend is in other city he can hear me good, but i can't hear him, both are using x-lite |
| 22:04.45 | Dr-Linux | what cound be problem ? |
| 22:04.47 | phantam | lol |
| 22:04.52 | jontow | both ends are full-duplex soundcards? |
| 22:05.00 | MikeJ[Jayden] | what's a good price on a polycom 500 in the US? |
| 22:05.06 | *** join/#asterisk Tarox (someone@pD9E79FB1.dip.t-dialin.net) |
| 22:05.08 | MikeJ[Jayden] | and user experience of them? |
| 22:05.25 | pr0m | got a tdm400 installed and asterisk up and running with "make samples". i'm not getting a dial tone on my analog phone connected to the fxs port on the tdm. |
| 22:05.37 | jontow | dr-linux; and furthermore.. is 'dtmf=inband' set? |
| 22:05.42 | phantam | well |
| 22:05.42 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 22:05.46 | phantam | ill let u know in a few seconds |
| 22:05.56 | Goshen | jerjer: here is the page to edit... http://www.voip-info.org/tiki-index.php?page=Nufone |
| 22:05.58 | pr0m | does anyone have a working set of configs for the tdm400 with a single fxs port? |
| 22:06.00 | phantam | gonna try it with latest cvs and latest pwlib/openh323 |
| 22:06.09 | JerJer[mobile] | Goshen: fire away |
| 22:06.09 | phantam | JerJer[mobile]: are u still actively workin on the h323 thing |
| 22:06.29 | JerJer[mobile] | define actively |
| 22:06.32 | ManxPower | pr0m, Other than the kernel module name, it would be the same config as an X100P |
| 22:06.33 | Dr-Linux | jontow: where i can set this >> 'dtmf=inband' ? |
| 22:06.47 | pr0m | ManxPower: ok that helps. |
| 22:06.57 | ManxPower | Dr-Linux, Are you using the ulaw or alaw codec? |
| 22:06.58 | phantam | lol |
| 22:07.02 | JerJer[mobile] | Goshen: i dispise that damn wiki so i refuse to even go to it |
| 22:07.06 | phantam | as in are u fixing bugs to keep it working with newer pwlibs |
| 22:07.07 | phantam | lol |
| 22:07.19 | phantam | hehe "define actively" |
| 22:07.28 | Goshen | jerjer: ok, I will do it for you then :) |
| 22:07.38 | Dr-Linux | ManxPower: i'm using defualt all |
| 22:07.45 | ManxPower | JerJer[mobile], Yeah, but you should at least make sure the information about your own company is not wrong. |
| 22:07.46 | Dr-Linux | i didn't changed any |
| 22:07.48 | Dr-Linux | jontow: where i can set this >> 'dtmf=inband' ? |
| 22:07.54 | ManxPower | Dr-Linux, NEVER use allow=all |
| 22:08.02 | Dr-Linux | okey |
| 22:08.05 | jsolares | no longer irq sharing, still not working... well that rules out the card, i'll try kicking the avaya definity |
| 22:08.12 | Dr-Linux | ManxPower: it can be effect on voice quality |
| 22:08.25 | ManxPower | Dr-Linux, English is not your native language? |
| 22:08.36 | Dr-Linux | but voice quality is very very good, but problem is this, i can't hear him, he can hear me good |
| 22:08.56 | Dr-Linux | ManxPower: yeah, english is not my native language |
| 22:09.01 | ManxPower | Dr-Linux, That is either a NAT issue or an allow=all issue or a firewall issue. |
| 22:09.37 | eKo1 | or a phone issue. |
| 22:09.56 | Dr-Linux | ManxPower: we both have live Ips |
| 22:10.17 | phantam | JerJer[mobile]: u never responded to my definition of actively |
| 22:10.18 | phantam | lol |
| 22:10.27 | TSCHAK | hey guys, I have a cisco callmanager and I need to route a pattern from it (701) to an extension in asterisk. |
| 22:10.32 | TSCHAK | any ideas? |
| 22:10.46 | TSCHAK | i've got a trunk set up in CM, as ASTERISK |
| 22:10.51 | TSCHAK | and I've routed a pattern to it.. |
| 22:10.55 | phantam | 701 prefix? |
| 22:11.29 | phantam | damn pwlib takes forever im tired of switchin versions lol |
| 22:11.37 | TSCHAK | phantam nah, just dialing '701' |
| 22:11.43 | zigman | phantam don't ;) |
| 22:11.44 | TSCHAK | I get a fast busy immediately. |
| 22:12.07 | JerJer[mobile] | phantam: I no longer see a need for H.323, thus anything in chan_h323 is going to have to come from the community or they are going to have to find a reason to motivate me |
| 22:12.11 | phantam | u mean when u dial 701 u want to get the callmanager |
| 22:12.17 | phantam | hmmm |
| 22:12.34 | phantam | why is h323 dead considering alot of pbx's dont support iax yet |
| 22:12.34 | zigman | JerJer[mobile] like t.38 ? |
| 22:12.39 | zigman | or do you want more |
| 22:12.48 | phantam | and in some cases sip really isnt possible due to machines in heavy production |
| 22:12.51 | JerJer[mobile] | there is no need for T.38 |
| 22:12.57 | JerJer[mobile] | app_tx and rxfax works damn good |
| 22:12.57 | phantam | i cant call a provider and tell them "switch u're box to sip" |
| 22:12.58 | phantam | lol |
| 22:13.09 | JerJer[mobile] | then find a real provider |
| 22:13.16 | phantam | lol |
| 22:13.34 | phantam | tell me how to get cisco 3660's to do iax2 and ill try that instead lol |
| 22:13.55 | JerJer[mobile] | load the firmware on it |
| 22:14.03 | ManxPower | This Celestial Seasonings Chai has got to be the best damn beverage I've had in a VERY long time. |
| 22:14.14 | ManxPower | phaded, They can do SIP. |
| 22:14.44 | ManxPower | Pretty much any Cisco box that is reasonably modern can do SIP. |
| 22:14.56 | JerJer[mobile] | and h.323 at the same time |
| 22:15.00 | Dr-Linux | ManxPower: what ports i should open to hear from outside ? |
| 22:15.12 | jsolares | well crap, my zap is dead |
| 22:15.13 | sivana | JerJer[mobile]: did you get my email? |
| 22:15.24 | ManxPower | Dr-Linux, I thought you just said you were both on public IPs. |
| 22:15.37 | JerJer[mobile] | sivana: you are going to have to be more specific than that...we deal with a stupid amount of email |
| 22:15.51 | ManxPower | Dr-Linux, By default Asterisk uses port 5060/UDP for SIP Signaling, and ports 10,000 - 20,000 for SIP/RTP Audio |
| 22:16.54 | Dr-Linux | ManxPower: yeah we are using public ip |
| 22:17.22 | sivana | JerJer[mobile]: about the CDR issues |
| 22:17.46 | Dr-Linux | ManxPower: so what port i should open for audio ? |
| 22:18.13 | eKo1 | didn't he just say 10000-20000 |
| 22:18.15 | ManxPower | Dr-Linux, By default Asterisk uses port 5060/UDP for SIP Signaling, and ports 10,000 - 20,000 for SIP/RTP Audio |
| 22:18.21 | jsolares | would having an uncofigured tormenta pci card have anything to do with my quad fxo digium card not working? |
| 22:18.43 | sivana | JerJer[mobile]: come to find out, when bridging two IAX2 systems, the call is then transferred between each other, cutting out the middle man :) |
| 22:18.52 | eKo1 | jsolares: maybe you're having irq problems. |
| 22:19.01 | jsolares | nope, no longer sharing irq |
| 22:19.02 | ManxPower | sivana, That is the way IAX@ transfers work. |
| 22:19.12 | ManxPower | If you don't like that set notransfer=yes in iax.conf |
| 22:19.16 | jsolares | 209: 937658 IO-APIC-level wctdm |
| 22:19.19 | sivana | yes, I realize that now :) |
| 22:19.21 | eKo1 | jsolares: so if you take out the tormenta card, does it work? |
| 22:19.29 | sivana | it should be no transfer by default I would think |
| 22:19.35 | jsolares | no idea... bah i knew putting the server in the server room was a bad idea |
| 22:19.59 | ManxPower | sivana, It's a matter of correct CDRs .vs. bandiwdth savings. I think the default of bandwidth savings is a good one. |
| 22:20.01 | eKo1 | Is it a big server? |
| 22:20.16 | pr0m | oops. i just called digium using iax from the demo. hehehe. well. the operator seems nice enough. she just laughed at me. ;-) |
| 22:20.17 | sivana | ManxPower: I see... the other IAX2 box is on my LAN extension |
| 22:20.18 | Juggie | if a user can have a one digit menu while they wait on a queue, what happens if a dial occurs in that context? when the agent answers the call, will the dial be interupted? |
| 22:20.26 | jsolares | not really, it's just a pain to remove stuff in the room |
| 22:21.05 | eKo1 | well, nobody said this was going to be soothing |
| 22:21.12 | sivana | ManxPower: do you know if the setting can be made under [general] or each entry individually? |
| 22:21.16 | jsolares | it was... damn tor2 card |
| 22:21.42 | ManxPower | sivana, I don't know, but I would assume you can put it either place. |
| 22:23.13 | jsolares | thanks eKo1, i'll go yank the tormenta card |
| 22:24.19 | eKo1 | Man, the lack of bandwidth here is killing me. |
| 22:24.43 | TSCHAK | do I need to add something to modules.conf to enable SIP ? |
| 22:24.52 | eKo1 | no |
| 22:25.09 | TSCHAK | should 5060 show up in port scans with nmap? |
| 22:25.16 | eKo1 | yes |
| 22:25.26 | TSCHAK | i'm not seeing anything. :-( |
| 22:25.54 | eKo1 | well, apart from that, does it work? |
| 22:26.01 | *** join/#asterisk Blackvel (~blackvel@dsl-082-082-059-240.arcor-ip.net) |
| 22:26.21 | TSCHAK | eKo1 no.. I can't seem to get a routed pattern to dial to it. |
| 22:26.22 | Dalion | TSCHAK yes |
| 22:26.36 | Dalion | aint it udp |
| 22:26.38 | Blackvel | hi, anyone knows, how I can remotely access the manager API? does that work over TCP/IP sockets? is there some interface to java or something else? |
| 22:26.38 | Dalion | ? |
| 22:27.07 | eKo1 | Blackvel: yes, yes, telnet |
| 22:27.27 | TSCHAK | what port do I telnet to ? |
| 22:27.49 | eKo1 | TSCHAK: turn of your firewall, make sure the phones are registered and do a 'sip debug' |
| 22:27.49 | Blackvel | telnet? interesting |
| 22:28.04 | eKo1 | Blackvel: check the wiki. |
| 22:28.04 | TSCHAK | eKo1 ok... what's the port to telnet to? |
| 22:28.08 | Primer | I wrote a perl script to interface with my asterisk console |
| 22:28.11 | TSCHAK | ok |
| 22:28.14 | Primer | which is actually an IRC bot |
| 22:28.29 | eKo1 | the port is 5081 or something like that. |
| 22:28.37 | Primer | [general] |
| 22:28.37 | Primer | enabled = yes |
| 22:28.37 | Primer | port = 5038 |
| 22:28.38 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 22:28.45 | Primer | from /etc/asterisk/manager.conf |
| 22:28.52 | Primer | you can set it to whatever you want |
| 22:29.02 | Beirdo | heya, lilo ;) |
| 22:29.36 | mikegrb | heya, Beirdo |
| 22:30.02 | Beirdo | hehe |
| 22:30.18 | TSCHAK | I type sip debug, and nothing happens. |
| 22:30.44 | LarsAC | how are the voiceboxes stored ? |
| 22:31.29 | *** join/#asterisk Markie (Mark@208.247.106.165) |
| 22:33.18 | Goshen | JerJer[mobile]: Just to be clear, you can port US48 1-8** numbers to your service? |
| 22:33.21 | Markie | if i'm monitoring a channel, and i transfer the call, it then records MOH..how do i make the monitor app follow a channel |
| 22:33.50 | *** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net) |
| 22:34.45 | *** join/#asterisk gezick (gezick@sartre.ispvip.biz) |
| 22:35.07 | gezick | is it possible to have meetme rooms with pure voip (i.e. not using a zaptel card) |
| 22:36.50 | tuxinator_linux | gezick: timing |
| 22:37.06 | tuxinator_linux | gezick: * uses zaptel for timing I think |
| 22:37.17 | CoaxD | gezick: It is possible, but not the best solution |
| 22:37.31 | CoaxD | gezick: You must, at the very least, use the zaprtc module to provide a timing source |
| 22:37.46 | CoaxD | but taht one is only 97% or so.. To get a 100% accurate timing source, you need to at least have an X100P or sometihng |
| 22:39.04 | *** join/#asterisk jsolares (~jsolares@200.30.141.85) |
| 22:39.14 | jsolares | well it wasnt the tormenta card |
| 22:39.21 | *** part/#asterisk PMantis (~PMantis_C@66.251.89.34) |
| 22:39.28 | *** join/#asterisk anthm (~anthm@68.29.19.1) |
| 22:39.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:39.32 | Blackvel | wow |
| 22:39.39 | Blackvel | this manager aPI socket thing is really cool! |
| 22:39.47 | phantam | JerJer[mobile]: are u there? |
| 22:39.50 | Blackvel | is it working as stable as .call files are? |
| 22:40.04 | Markie | i jsut started using the manager API and i'm very happy with it. |
| 22:40.08 | jsolares | time to try lastest cvs, new kernel, recompiled drivers |
| 22:40.17 | Markie | i'm using both right now..call files and the manager API |
| 22:40.37 | Markie | cal lfiles are for "automated" stuff, whereas manager API i use for more "realtime" stuff |
| 22:40.38 | CoaxD | dont understand the need for .call files |
| 22:40.40 | CoaxD | maybe i should read up on that |
| 22:41.07 | Blackvel | Markie, where do you use the manager API from? |
| 22:41.23 | Markie | ummfrom a web page. |
| 22:41.28 | Blackvel | perl/php? |
| 22:41.40 | Markie | thats the nice thing, cause i can initiate a call from a remote machien, cant do that with .call files |
| 22:41.41 | Markie | php |
| 22:41.57 | Blackvel | looks great |
| 22:42.03 | tzafrir_home | Markie: sure you can. Using ssh |
| 22:42.18 | Markie | sorry..let me rephrase.. |
| 22:42.22 | phantam | it cant find the header file |
| 22:42.25 | phantam | files |
| 22:42.25 | Blackvel | maybe it is just the issue about performance at end end with many actions |
| 22:42.27 | Blackvel | and events |
| 22:42.37 | Blackvel | but there is a way to test around :) |
| 22:42.38 | Markie | you CAN, but it's UGLY, unmanagable, and needs to be maintained |
| 22:42.55 | Markie | one thing i actually just read today is the ActionID |
| 22:43.07 | phantam | where the hell is h323.h |
| 22:43.09 | Markie | i'm not using it now, but i'm going to use it to help with the parsing |
| 22:43.47 | Blackvel | thanx for tip |
| 22:44.10 | Markie | i use .call files for text-to-speach automated outgoing calls. |
| 22:44.22 | Markie | i.e. send an email, it'll dial out and "play" the email |
| 22:44.27 | tzafrir_home | phantam: if you've installed libopenh323 from a decent package you'd have a clue as to file locations... |
| 22:44.52 | Markie | night(ish) |
| 22:45.11 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 22:45.16 | Markie | so..how about anyone knowing the answer to my problem? |
| 22:45.34 | phantam | what is it with this room |
| 22:45.37 | Markie | if i'm monitoring a channel, and i transfer the call, it then records MOH..it doest "follow" the channel. how do i make the monitor app follow a channel |
| 22:45.41 | phantam | every has to be extremely sarcastic |
| 22:45.56 | CoaxD | Markie: Hmmmm. interesting |
| 22:46.12 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 22:46.21 | CoaxD | Markie: I've had that problem too |
| 22:46.26 | CoaxD | Markie: Annoying isnt it |
| 22:46.40 | Markie | yea...we record all calls nthe company..and we just noticed that any calls that get transferred form one internal phone to another doesnt get followed |
| 22:46.51 | Markie | CoaxD: any solution you've run across? |
| 22:47.01 | CoaxD | Markie: I don't think there's a way to MAKE ti follow. technically, internally to asterisk, it is actually a different call |
| 22:47.15 | Markie | but the "different" call is not recorded either :( |
| 22:47.17 | CoaxD | The only way would be to start a record session on each individual extension you transfer to |
| 22:47.40 | Markie | hrmm..but the transfer is done on the phone side.. i.e. they flash-transfer |
| 22:47.48 | Markie | so i dont evne know ehre it even touches asterisk |
| 22:48.13 | CoaxD | markie: if you're transferring, asterisk is most certainly in the media path |
| 22:48.31 | anthm | you can put a box upstream and monitor that or call over a loopback interface and monitor that otherwise you need a chainsaw and the src |
| 22:48.36 | Markie | yea, i know it's in the media path..but i dont know where/how it hits something like an extensions.conf |
| 22:48.55 | Markie | anthm: ok..so you mean there aint nothing built in currently.. |
| 22:48.57 | CoaxD | markie: If you're transferring, obviously you're transferring to an extension number |
| 22:49.06 | Markie | CoaxD: hrmmmm... |
| 22:49.22 | Markie | CoaxD: we're not recording internal-to-internal calls |
| 22:49.28 | Markie | maybe i neeed to do that? |
| 22:49.29 | CoaxD | markie: if in each extension you set up a record macro.. it'd work |
| 22:49.38 | CoaxD | markie: it'd start a new file |
| 22:49.42 | CoaxD | markie: But it'd work |
| 22:49.48 | *** join/#asterisk Tarox (someone@pD9E79FB1.dip.t-dialin.net) |
| 22:49.54 | Markie | so you think it's treated as an internal-to-internal call? |
| 22:50.05 | CoaxD | markie: How else do you think it knows the extension number? |
| 22:50.21 | CoaxD | Asterisk doesn't care whats internal or external. those are terms YOU defined |
| 22:50.29 | Markie | yea..correct.. |
| 22:50.31 | Markie | hrm.. |
| 22:50.38 | anthm | when you transfer it moves the soul of the call to a new object and the carcass of the old chan will fall lifeless and make the monitor stop |
| 22:50.39 | Markie | i'm jsut looking at the CDR info..triyng to figure it out |
| 22:50.48 | CoaxD | anthm: haha |
| 22:51.01 | anthm | like i said if you *need* it bridge over a loopback first and record that |
| 22:51.18 | Markie | hrm.. |
| 22:51.19 | CoaxD | anthm: What is a loopback? |
| 22:51.37 | CoaxD | anthm: You're a an asterisk hacker; you know all those neat things; many of us are so green its not even funny |
| 22:51.50 | file | soylent green |
| 22:51.50 | Markie | what's the difference between "channel" and "source" as far as the CDR is concerned? |
| 22:51.51 | CoaxD | I make * go for my office, but.. i'd love to know how to do everytihng |
| 22:52.20 | anthm | loopback wouild be a peer on iax or sip that points at the same box or a physocal zap loopback tunnel using ztd_local |
| 22:52.40 | CoaxD | anthm: Hmmmm. thank you |
| 22:52.52 | Markie | but it looks like a simple recording of "internal" numbers will solve it.. |
| 22:53.06 | file | anthm: it's sad you're not going to VON :( |
| 22:53.12 | tuxinator_linux | <PROTECTED> |
| 22:53.16 | Markie | question.. is there a way i can do something like if "came from a transfer", record, |
| 22:53.21 | Silik0n | is realtime for vm still borked? |
| 22:53.29 | anthm | yah i wanted to |
| 22:53.50 | tuxinator_linux | how sweet |
| 22:54.02 | anthm | you can try the -b thing i added |
| 22:54.06 | *** join/#asterisk apaneiro (~apaneiro@bl4-65-237.dsl.telepac.pt) |
| 22:54.08 | anthm | to res_monitor |
| 22:54.29 | anthm | that says automonitor whenever a bridge happens |
| 22:54.29 | Markie | well...ok...one other Q...the pre-transfered call didnt end after the transfer.. |
| 22:54.40 | buddah | is there a way i can direct traffic coming from 1 ip to be routed to gateway A, and any other traffic calling the same area to go to gateway B? |
| 22:54.42 | anthm | if the new chan inherits the var it stores that in then it will auto record |
| 22:54.47 | gezick | CoaxD: what is an X100P |
| 22:54.48 | Markie | the Monitor app kept recording MOH until i was done with the transferred phone conversation |
| 22:54.53 | anthm | or you can set that var with that _ goodies |
| 22:55.34 | Markie | umm...yea, i aready use the -b now. |
| 22:55.48 | buddah | also, is nufone always bad about communication? i've sent 2 emails, called, left a message, and no response. are they always like that? |
| 22:55.48 | Markie | thats what allows it to work in the first place for the initial call.. |
| 22:56.06 | anthm | -b just sets a var so if that var started with __ i think it would survive tranfers and keep happening |
| 22:56.43 | BrianR___ | tzanger: Around? |
| 22:56.48 | flewid | sup |
| 22:57.05 | Markie | ok..i'll try that..one other question..when someon's doing a flash-transfer from the phone, what context in the extensions.conf does it use? |
| 22:57.12 | flewid | anyone here going to von? i have a silly question, i've never been to a conference like this before, i've already registered and everything, but i haven't received anything in the mail |
| 22:57.22 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
| 22:57.22 | flewid | should i worry? or is there just a table i go up to and give my name and get a pass or something? |
| 22:57.24 | BrianR___ | buddah: I think nufone may be a very small shop... There's a few other similar services that are just as cheap though. |
| 22:57.26 | Markie | oh..i think i ansered it myself..if i pretend that a flash-transfer is the same as an originated call. |
| 22:58.29 | buddah | heh, they arent cheap though |
| 22:58.29 | Markie | i still wonder why the original pre-transferred channel continues to record MOH until the call is over |
| 22:58.37 | buddah | they are actually on the expensive side |
| 22:58.37 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 22:58.44 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 22:58.47 | Markie | like it doesnt release that channel until the entire call is completed. |
| 22:58.49 | *** join/#asterisk orpheusp (~oren-pins@200.204.120.198) |
| 22:59.16 | ManxPower | Markie, get the latest version of whatever Asterisk branch you use. There were some MoH and transfer issues fixed recently. |
| 22:59.51 | orpheusp | hello - looking for help in setting a simple goto/context. any volunteer to guide me? using AMP and X100P |
| 23:00.02 | Markie | ManxPower: thanks! |
| 23:00.15 | Markie | i think i'm using latest cvs..how late is relative.. |
| 23:00.27 | Markie | Connected to Asterisk CVS-HEAD-02/16/05-14:03:59 |
| 23:00.43 | Markie | you think later than that? |
| 23:00.46 | BrianR___ | I got MWI working between a norstar MICS key system and asterisk voicemail |
| 23:01.46 | ManxPower | Hmmph! Cox Cable New Orleans resets the TOS bits on outgoing packets |
| 23:01.48 | *** join/#asterisk opinsky (~opinsky@200.204.120.198) |
| 23:02.03 | Markie | is thers a cvs log easily accessible? |
| 23:02.08 | Markie | or cvsweb? |
| 23:02.09 | *** part/#asterisk Dalion (~DaLion@HSE-QuebecCity-ppp3497400.sympatico.ca) |
| 23:02.13 | ManxPower | Markie, Yes, the fix was in the past 10 days. If you use CVS you should be on the asterisk-cvs mailing list. |
| 23:02.28 | ManxPower | Markie, The asterisk-cvs mailing list archive at lists.digium.com |
| 23:02.37 | Markie | ok |
| 23:02.48 | Markie | i'm just on the devel and users |
| 23:02.51 | Markie | danke. |
| 23:03.36 | jontow | BrianR; does it use SMDI? |
| 23:03.48 | jontow | if not. .has anyone done SMDI-over-RS232 with asterisk and any other equipment? |
| 23:04.11 | gezick | if my * server is behind a firewall, and i'm behind a different firewall, what's a relatively easy way to do SIP calls (with x-lite or similar)? |
| 23:04.20 | gezick | or is there one? |
| 23:04.24 | mishehu | vpn |
| 23:04.24 | mishehu | heh |
| 23:04.40 | gezick | without doing a vpm ;-) |
| 23:04.46 | gezick | er. or a vpn. |
| 23:04.58 | jontow | tried any strangeness with IAX2? |
| 23:05.40 | jontow | got a third hop that isn't firewalled? :) |
| 23:05.52 | JohnnyC | anyone with a extensions.conf with CAPI ? How can I dial out ? |
| 23:06.01 | ManxPower | gezick, There is no such thing as "easy" if you have double-NAT. |
| 23:07.57 | opinsky | Hello - I am having trouble setting-up a simple extension in a context that is called through a goto command. I have included on my extensions_custom.conf |
| 23:08.17 | *** part/#asterisk Markie (Mark@208.247.106.165) |
| 23:08.30 | opinsky | [custom-count2four] s,1,Wait(2) s,2,SayDigits(1234) |
| 23:08.45 | opinsky | in a different context i "goto (custom-count2four,s,1) |
| 23:08.48 | opinsky | doesn't work |
| 23:08.59 | jontow | opinsky; whats the console say? |
| 23:09.07 | opinsky | it say executing (goto) |
| 23:09.12 | opinsky | and nothing else |
| 23:09.16 | opinsky | and hangs-up |
| 23:09.18 | gezick | ManxPower: it's all been so 'easy' so far, i was just hoping. |
| 23:09.46 | gezick | can someone point me to a good guide for punching holes in a firewall for *? |
| 23:09.55 | JohnnyC | does anyone has a ISDN example to dial out from a SIP ? |
| 23:10.12 | opinsky | it says exactly: -- Executing Goto("SIP/202-1b78", "custom-count2four|7777|1") in new stack |
| 23:10.12 | opinsky | <PROTECTED> |
| 23:10.19 | opinsky | and stops |
| 23:10.27 | jontow | ok.. 7777 ? |
| 23:10.28 | opinsky | oops change 7777 for s |
| 23:12.19 | Juggie | hrm, i cant seem to get agent groups working... |
| 23:12.45 | jontow | opinsky; in extensions.conf .. [custom-count2four] is on its own line yes? you just concatenated for pasting? |
| 23:12.56 | Juggie | anyone have any problems with that, i am running cvs head.... |
| 23:13.04 | *** part/#asterisk apaneiro (~apaneiro@bl4-65-237.dsl.telepac.pt) |
| 23:13.11 | jontow | juggie; i've got them working to some extent with -r v1-0 |
| 23:13.11 | Juggie | i can put agents in a queue if i specify their agent id |
| 23:13.19 | opinsky | yes, every instruction has its own line |
| 23:13.23 | jontow | but i had the problem you speak of in HEAD ages ago :) |
| 23:13.28 | Juggie | but if i do Agent/@1 |
| 23:13.31 | Juggie | it doesnt work |
| 23:15.29 | TSCHAK | can someone help me with a cisco callmanager to asterisk problem? |
| 23:18.16 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 23:19.33 | hajekd | can you recommend some reliable pstn termination in europe? |
| 23:23.02 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 23:23.02 | tzanger | BrianR___: am now |
| 23:24.08 | Juggie | is anyone running cvs-head with agents/queues i may have found a bug, need to confirm, because i may be a retard :) |
| 23:25.07 | *** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 23:26.32 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 23:26.33 | anthm | nah, if you were a chimp who had accidently typed in the queue configuration with shakespere in the comments i'd still suspect it was a bug if it didn't work |
| 23:27.18 | Juggie | i'll pastebin |
| 23:27.23 | Juggie | to remove the retard factor |
| 23:27.31 | tzanger | you need confirmation that you're a retard? :-) |
| 23:27.42 | CoaxD | somebody oughtta set THAT in the topic. haha |
| 23:27.58 | *** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com) |
| 23:28.43 | *** join/#asterisk clord (~Clint@166.70.78.242) |
| 23:28.45 | Juggie | tzanger, i've been wrong before :) |
| 23:29.04 | Juggie | http://pastebin.ca/6748 |
| 23:29.07 | CoaxD | all of us have been retards at one time or another; thats not a big thing. heh |
| 23:29.11 | anthm | i can tell you for sure chan_agent has a segfault hiding in it |
| 23:29.12 | tzanger | agreed |
| 23:29.14 | clord | What kinds of issues can cause "Unable to create channel of type 'Zap'"?? |
| 23:29.24 | Qwell | clord: not having the module loaded |
| 23:29.24 | clord | I have the chan_zap.so file. |
| 23:29.24 | tzanger | clord: you don't have your zap channels set up |
| 23:29.35 | tzanger | clord: the module's not loading (see last statement) |
| 23:29.38 | Juggie | that looks right no? |
| 23:29.42 | tzanger | clord: all channels are in use |
| 23:30.03 | clord | ok |
| 23:30.04 | clord | thanks. |
| 23:30.07 | clord | I'll look into those. |
| 23:30.15 | tzanger | clord: btw that was actually a very good question |
| 23:30.31 | tzanger | not great but certainly acceptable in these circles as far as newbie questions go |
| 23:30.36 | *** join/#asterisk Muiz (someone@dhcp185-1-186.dsl.ucc-net.ca) |
| 23:30.44 | tzanger | I can't think of a single digium resource which would answer that question short of scanning the source |
| 23:30.46 | Qwell | tzanger: As opposed to "Help! I get an error!"? |
| 23:30.51 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 23:30.52 | tzanger | Qwell: exactly |
| 23:31.03 | Juggie | tzanger, would that be correct what i pastebin'ed i assume., since its pretty damn simple. |
| 23:31.06 | clord | he he! Thanks... my biggest issue is that it was working just fine, and then I rebooted the server and shazam... not working. |
| 23:31.10 | Juggie | er, *that would |
| 23:31.23 | tzanger | clord: in that case I'd suspect you didn't load your zaptel modules |
| 23:31.25 | Juggie | clord, not loading your modules? |
| 23:31.26 | CoaxD | clord: Ah! You probably forgot the 'ztconfig -vv' step |
| 23:31.28 | Qwell | clord: sounds most like the zap kernel modules aren't loaded. I'm not muchhelp though usually |
| 23:31.37 | CoaxD | clord; or, yes, you forgot to load your zaptel stuff |
| 23:31.39 | tzanger | Juggie: not sure, I don't know anything about your particular problem |
| 23:31.41 | tzanger | I've never used agents |
| 23:31.43 | clord | I did do ztcfg -vv already |
| 23:31.47 | clord | and it's showing as loaded. |
| 23:31.49 | CoaxD | clord: and THAT worked? |
| 23:31.51 | clord | "configured" |
| 23:31.53 | gezick | which ports do i have to expose to initiate SIP calls through a firewall? |
| 23:31.53 | clord | yes |
| 23:31.59 | CoaxD | clord: No errors? |
| 23:32.01 | tzanger | clord: hmm interesting |
| 23:32.02 | clord | nope |
| 23:32.05 | CoaxD | clord: nifty |
| 23:32.05 | clord | ztcfg -vv |
| 23:32.08 | tzanger | clord: it says x channels configured? |
| 23:32.09 | CoaxD | clord: /dev/zap stuff exists? |
| 23:32.10 | clord | works fine. |
| 23:32.11 | tzanger | don't spit out the entire output |
| 23:32.13 | clord | yes |
| 23:32.15 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 23:32.19 | Muiz | does anybody know what happened to Loligo's web site? I can't get a conection |
| 23:32.25 | tzanger | clord: interesting |
| 23:32.31 | tzanger | what's 'zap show channels' say in the asterisk CLI |
| 23:32.32 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
| 23:32.33 | clord | could be an "in use" issue. |
| 23:32.39 | clord | as you said. |
| 23:32.44 | clord | right? |
| 23:32.52 | tzanger | could still be a number of things |
| 23:32.58 | clord | So the switch that line is hooked to could have it off hook or something. |
| 23:33.07 | tzanger | what's 'zap show channels' say in the asterisk CLI |
| 23:33.13 | clord | checking. |
| 23:33.20 | *** join/#asterisk vickers (~dsimmeth@216.57.217.138) |
| 23:33.27 | tzanger | I'm trying out red curry chicken tonight |
| 23:33.31 | tzanger | smells alright |
| 23:33.45 | clord | pseudo incoming |
| 23:33.48 | clord | 1 incoming |
| 23:33.51 | *** join/#asterisk ptblank (~MURDER1@68-169-176-29.lmdaca.adelphia.net) |
| 23:33.52 | Muiz | hmmm....i guess nobody knows what happened to the loligo website.... |
| 23:34.04 | tzanger | clord: that's fine, what's zap show channel 1 say |
| 23:34.07 | Muiz | can somebody help me out by sending me a couple of the sound files from their package? |
| 23:34.13 | tzanger | especially in regards to hookstate (I'm assuming it's an FXO card) |
| 23:34.15 | tzanger | oh wait |
| 23:34.19 | tzanger | fxo cards don't show hookstate |
| 23:34.28 | clord | offhook |
| 23:34.39 | tzanger | clord: yeah ignore that fxo cards don't show hookstate |
| 23:34.44 | clord | ah |
| 23:34.46 | tzanger | I should patch asterisk so that it doesn't even show that line |
| 23:34.56 | CoaxD | yeah, its purely cosmetic |
| 23:35.12 | tzanger | clord: is the channel in red alarm? |
| 23:35.13 | CoaxD | although it realy should be able to tell whether it is on-hook or off-hook |
| 23:35.18 | tzanger | CoaxD: agreed |
| 23:35.26 | CoaxD | guaranteed, the card knows |
| 23:35.28 | clord | where would I see that? |
| 23:35.31 | clord | red state |
| 23:35.34 | *** part/#asterisk Muiz (someone@dhcp185-1-186.dsl.ucc-net.ca) |
| 23:35.38 | clord | red alarm |
| 23:35.38 | CoaxD | clord: zttool |
| 23:35.38 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 23:35.51 | clord | exit |
| 23:35.52 | CoaxD | clord: Comes with zaptel |
| 23:35.53 | tzanger | /proc/zaptel/1 should show it too I think |
| 23:35.53 | clord | he he! |
| 23:35.55 | clord | sorry |
| 23:35.56 | vickers | I'm looking for information on configuring Sphinx with asterisk, but have found very little. |
| 23:36.06 | tzanger | CoaxD: only if you have that goofball libnewt installed |
| 23:36.09 | clord | yes |
| 23:36.09 | clord | RED |
| 23:36.18 | tzanger | clord: that means there's no phone line plugged in to it |
| 23:36.21 | CoaxD | clord: you dont have a line hooked to it |
| 23:36.30 | vickers | anyone have experience using voice recognition software with asterisk? |
| 23:36.34 | CoaxD | clord: asterisk wont make a connection to a zap channel in red alarm state, i wouldnt imagine |
| 23:36.36 | tzanger | vickers: not me |
| 23:36.38 | clord | perfect... so it's our switch tech's fault... perfect. |
| 23:36.39 | clord | he he! |
| 23:36.41 | tzanger | CoaxD: nope it won't |
| 23:36.51 | clord | Love to hear it's them and not me. |
| 23:36.56 | dalabera | hi, anyone here with experience programing AGI scripts in PERL? |
| 23:36.56 | CoaxD | tzanger: Sweet. i havent tested that |
| 23:37.08 | tzanger | well red alarms take the channel(s) right out of commission |
| 23:37.23 | tzanger | at least on t1/e1 so I imagine if they put red alarm in there for a mere POTS line they'd treat it the same |
| 23:37.26 | *** join/#asterisk ReVoK (ReVoK@82.224.60.46) |
| 23:37.33 | ReVoK | hi |
| 23:37.39 | dalabera | I'm doing a perl script and weirds are happenning to me, please help |
| 23:37.41 | clord | ok, so once the line is setup it should be back to normal. |
| 23:37.53 | tzanger | clord: yes you'll see a message about the zap channel coming out of red alarm |
| 23:37.56 | tzanger | and it'll be available |
| 23:38.09 | clord | nice... thanks as always, this channel is always helpful. |
| 23:38.13 | ManxPower | dalabera, You are not reading STDIN when you start your perl script. |
| 23:38.46 | ManxPower | dalabera, Use asterisk-perl and let it handle all the ugly stuff for you. |
| 23:40.30 | dalabera | that's what I'm using |
| 23:41.06 | dalabera | you meant reading STDIN by issuing this command foreach $key (keys %input) { |
| 23:41.06 | dalabera | <PROTECTED> |
| 23:41.06 | dalabera | <PROTECTED> |
| 23:41.25 | dalabera | my %input = $AGI->ReadParse(); |
| 23:41.38 | Juggie | no pasteing plz, use pastebin.ca |
| 23:41.44 | ManxPower | dalabera, It's been at least a year since I've used AGI. You only need to "read stdin" if you are not using asterisk-perl. |
| 23:41.57 | ta[i]nted | anyone having inaccurate DIALSTATUS in AGI? |
| 23:42.01 | ManxPower | Follow the exmaple(s) that come with asterisk-erl. |
| 23:42.13 | ManxPower | ta[i]nted, What's happening with it? |
| 23:42.45 | ta[i]nted | calling the same number and receiving the same response i get different DIALSTATUS results |
| 23:42.56 | ta[i]nted | sometimes 'noresponse', sometimes 'ANSWERED' |
| 23:43.01 | ManxPower | ta[i]nted, VoIP or PSTN? |
| 23:43.05 | ta[i]nted | voip |
| 23:43.33 | ta[i]nted | what actually determines whether a call is answered |
| 23:43.38 | ManxPower | So it's not an agi issue at all. |
| 23:43.38 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 23:43.49 | ManxPower | ta[i]nted, Watch the console, does it show answered? |
| 23:43.50 | ta[i]nted | if it goes to voicemail, is that call 'answered' or no |
| 23:43.55 | ta[i]nted | yes |
| 23:44.02 | ManxPower | yes, voicemail is an answer |
| 23:44.04 | ta[i]nted | but agi still returns 'noresponse' |
| 23:44.27 | ManxPower | Um...IIRC there is no NORESPONSE value for DIALSTATUS |
| 23:44.28 | ta[i]nted | at least $AGI->get_variable("DIALSTATUS") does anyway |
| 23:45.11 | *** join/#asterisk mesi (~player@dsl-082-083-061-248.arcor-ip.net) |
| 23:45.11 | machinehd | how hard is it to stumble accross sip formware for the cisco 7960? |
| 23:45.12 | ta[i]nted | interesting... that's the returned value |
| 23:45.12 | ManxPower | ${DIALSTATUS} Status of the call, one of: |
| 23:45.12 | ManxPower | <PROTECTED> |
| 23:45.17 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 23:45.29 | ta[i]nted | let me audit my code.. see if i've missed something |
| 23:45.55 | ManxPower | ta[i]nted, Try a SetVar(MY_DIALSTATUS=${DIALSTATUS}) before calling your AGI. |
| 23:46.05 | ManxPower | See if there's a difference in the value of the two of them in your AGI. |
| 23:47.31 | ta[i]nted | good idea |
| 23:47.41 | ManxPower | ta[i]nted, Getting automatically set dialplan variables is a VERY recent addition to AGI. |
| 23:48.20 | Dr-Linux | tzanger |
| 23:48.31 | tzanger | Dr-Linux: what's up |
| 23:48.49 | Dr-Linux | how are you tzanger :) |
| 23:48.56 | tzanger | alright |
| 23:49.49 | TSCHAK | this is driving me |
| 23:49.50 | TSCHAK | crazy |
| 23:49.57 | *** join/#asterisk Grant2 (~grant@adsl-39-232.swiftdsl.com.au) |
| 23:49.57 | *** join/#asterisk DevilFish (~DevilFish@kleanmail.com) |
| 23:50.05 | psywar | TSCHAK: is it a long drive? |
| 23:50.07 | TSCHAK | cisco callmanager refuses to talk to asterisk OR the GNUgk |
| 23:50.19 | psywar | ;) |
| 23:50.23 | DevilFish | can anyone have a look at this and tell me what they think? |
| 23:50.24 | DevilFish | http://lists.digium.com/pipermail/asterisk-users/2005-January/083456.html |
| 23:50.32 | Grant2 | anyone have some nice tutorials? |
| 23:50.40 | DevilFish | got disconnection problems and I cannot seem to beat them |
| 23:51.10 | gezick | i'm trying use x-lite to make SIP calls on the mac, i have it so that incomming calls are working, but outbound calls don't even generate any log information in asterisk. i'm sure that it's a set up problem in x-lite (at least) does anyone know where a likely source of the problem might be |
| 23:51.21 | DevilFish | Also is anyone using MetaSwitch? |
| 23:51.39 | *** part/#asterisk darkskiez (~mhb@host-84-9-70-212.bulldogdsl.com) |
| 23:52.52 | mishehu | I'm using the LightSwitch |
| 23:53.05 | mishehu | it does amazing things, like bring brightness into the room. ;-) |
| 23:54.04 | tzanger | http://homepage.ntlworld.com/nathanroberts1/fark/horsey2.gif |
| 23:54.09 | tzanger | how to make a horse go faster :-) |
| 23:54.30 | DevilFish | har har har :) |
| 23:55.16 | TSCHAK | has anyone in here worked with Cisco CM? |
| 23:56.06 | Dr-Linux | ManxPower: there ? |
| 23:56.16 | ManxPower | Dr-Linux, Sort of |
| 23:56.47 | JohnnyC | can someone hep me wth CAPI ? |
| 23:56.56 | JohnnyC | I cant seem to find any example to Dialout |
| 23:56.59 | JohnnyC | at leat updated |
| 23:57.06 | Dr-Linux | okey |
| 23:57.52 | Dr-Linux | ManxPower: can you tell me how i can fix this NAT issue ? having audio problem as i asked your b4 ? |
| 23:57.52 | ManxPower | ~google site:lists.digium.com CAPI Dial |
| 23:57.53 | ManxPower | Dr-Linux, No, I cannot. If I could, I would have. |
| 23:59.10 | ta[i]nted | ManxPower same results -- both DIALSTATUS and MY_DIALSTATUS posts 'noresponse' randomly |
| 23:59.54 | ManxPower | ta[i]nted, Is that any different than what they say in the dialplan? NoOp(DIALSTATUS=${DIALSTATUS}) |
