00:00.07 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
00:00.29 | file[mac] | aye |
00:04.39 | *** part/#asterisk Hydroxide (user@Hydroxide.developer.debian) |
00:06.44 | *** join/#asterisk Inv_arp (junya@adsl-3-245-209.mia.bellsouth.net) |
00:14.08 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
00:16.24 | TUplink | ERRRR this program is starting to get to me..... when i enter an extnetionat my IVR it takes it as 20 instead of 20002 |
00:16.31 | FuriousGeorge | nothing is more confusing, or more poorly laid out than zapata.conf |
00:18.17 | MichaelCat | Can you capture and use the return codes if you call a shell script from a dialplan? |
00:19.38 | three55ml | PTG1234: No Rio? |
00:19.47 | PTG1234 | rio was too big compared to the mini |
00:19.56 | three55ml | Most things are :) |
00:19.58 | PTG1234 | and the mini's battery life of 18hours just sold me aloing with the size |
00:20.04 | three55ml | Ah |
00:20.20 | PTG1234 | hey three would you have a few moments to help me with courier later? |
00:20.35 | PTG1234 | or do you have no idea how to set it up |
00:20.38 | *** join/#asterisk file[mac] (~file[mac]@mctn1-3046.nb.aliant.net) |
00:20.44 | PTG1234 | file: welcome |
00:20.56 | three55ml | I do, what are you trying to do and I'll tell you if I've done it before |
00:21.07 | PTG1234 | i need it to use the vpopmail format |
00:21.13 | PTG1234 | and have no idea how to tell it which auth module to use |
00:21.19 | three55ml | Latest version? |
00:21.22 | PTG1234 | yah |
00:21.28 | PTG1234 | actually one sec let me reboot |
00:21.35 | PTG1234 | so my ipod works :) |
00:21.38 | three55ml | Let me look |
00:21.39 | three55ml | ok |
00:21.46 | file[mac] | meep |
00:21.56 | file[mac] | ooh iPod |
00:22.59 | *** join/#asterisk Skarmeth (~Skarmeth@201009017181.user.veloxzone.com.br) |
00:23.04 | Skarmeth | hi all |
00:23.33 | Skarmeth | I'm trying to set a lab asterisk with a md3200 board, but not found docs about it |
00:23.47 | Skarmeth | can anyone point me to any source of info? |
00:26.34 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
00:27.25 | *** join/#asterisk In-Side (~Lowgitek@es-217-129-30-41.netvisao.pt) |
00:27.27 | In-Side | Hi guys |
00:27.45 | In-Side | I have a problem that i will describe now to know if anybody can give me a clue |
00:28.15 | In-Side | when i use iax2 to connect fwd everything works nice |
00:28.28 | In-Side | when i use sip i get always 503 as answer |
00:28.46 | Sedorox | you can't use both to FWD |
00:28.48 | Skarmeth | ok, get it working |
00:28.52 | In-Side | and if i try to configure a nated client directly it simple doesn't have sound as supposed |
00:29.03 | In-Side | I'm jsut using one |
00:29.27 | Skarmeth | Zapata Telephony Interface Registered on major 196 |
00:29.27 | Skarmeth | wcfxo: DAA mode is 'FCC' |
00:29.27 | Skarmeth | Found a Wildcard FXO: Generic Clone |
00:29.28 | Skarmeth | Registered tone zone 0 (United States / North America) |
00:29.42 | Skarmeth | I've two card's in this machine |
00:29.53 | Skarmeth | a Intel 537 and a Ambient md3200 |
00:30.13 | MikeJ[Jayden] | ~pastebin |
00:30.15 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
00:30.15 | bugbot | pastebin is assigned nothing and reported nothing. |
00:30.17 | Skarmeth | if I put fxsks=2, I'll get all two working? |
00:33.29 | MichaelCat | Can you capture and use the return codes if you call a shell script from a dialplan? |
00:34.31 | MikeJ[Jayden] | http://www.voip-info.org/wiki-Asterisk+cmd+System |
00:36.01 | MichaelCat | yes it looks like system will not allow me to use custoemr return codes from a shell script, am i missing something or is there a way to do it |
00:37.14 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
00:37.17 | PTG1234 | ok back |
00:37.24 | PTG1234 | ipod won't detect but i'm back :) |
00:37.33 | file[mac] | how rude of it |
00:37.45 | PTG1234 | haha |
00:37.52 | PTG1234 | well i told you i bought a mac just for my ipod |
00:37.56 | three55ml | PTG1234: You just need to know where to specify to use vpopmail with Courier or what? |
00:37.58 | PTG1234 | maybe i should use that again :) |
00:38.03 | file[mac] | haha |
00:38.05 | three55ml | Haha |
00:38.13 | three55ml | I think I might get a Mac mini |
00:38.15 | PTG1234 | three ya |
00:38.18 | PTG1234 | the docs for courier suck |
00:38.24 | MichaelCat | the shell script does stuf and returns different codes 0 thru 9 depending on what it finds, i need to have the extension do differnt things based on the return codes from the script |
00:38.31 | PTG1234 | file[mac]: shouldn't the ipod just show up in the list? |
00:38.38 | PTG1234 | in itunes |
00:39.08 | three55ml | PTG1234: On Linux at least, you'd edit /usr/local/etc/authlib/authdaemonrc |
00:39.18 | file[mac] | yes |
00:39.21 | PTG1234 | hmm |
00:39.23 | PTG1234 | let me look at that |
00:39.46 | PatrickDK | hmm, I have a fun email setup :) |
00:39.53 | three55ml | PTG1234: There's a line that says "authmodulelist="...add "vchkpw" to the list. I just use vpopmail, so mine reads: authmodulelist="authvchkpw" |
00:40.38 | PTG1234 | ok its in there so it will use that? |
00:40.41 | *** part/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
00:40.47 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
00:40.48 | PTG1234 | whoops |
00:40.50 | PTG1234 | what did i miss :) |
00:40.58 | three55ml | nothing, yeah...that should be it |
00:40.58 | *** join/#asterisk sivana (~sivana@165.154.13.35) |
00:40.58 | PTG1234 | authmodulelist="authuserdb authpam authvchkpw" |
00:40.58 | PTG1234 | <PROTECTED> |
00:40.58 | PTG1234 | ##NAME: authmodulelistorig:2 |
00:40.58 | PTG1234 | # |
00:40.58 | PTG1234 | # This setting is used by Courier's webadmin module, and should be left |
00:40.58 | PTG1234 | # alone |
00:41.00 | PTG1234 | <PROTECTED> |
00:41.03 | PTG1234 | authmodulelistorig="authuserdb authpam authvchkpw" |
00:41.07 | three55ml | Leave orig |
00:41.34 | PTG1234 | file you alive |
00:41.40 | PTG1234 | my ipod has a folder icon on it with a ! |
00:41.42 | PTG1234 | what does that mean |
00:41.55 | PTG1234 | and now usb won't detect it |
00:42.09 | PTG1234 | think the batteries just need to be charged? |
00:42.21 | file[mac] | I haven't a clue |
00:42.21 | file[mac] | lol |
00:42.38 | file[mac] | but yes, the batteries probably do need to be charged |
00:43.18 | PTG1234 | ok |
00:43.22 | PTG1234 | so courier just should WORK? |
00:43.26 | PTG1234 | how do i add an account? |
00:44.23 | PatrickDK | dunno, doesn't vpopmail use some kind of flat file or mysql database? |
00:44.36 | PatrickDK | I've used vm-pop and now I am using ldap |
00:44.36 | PTG1234 | flat file |
00:44.40 | PTG1234 | i think :) |
00:44.41 | unterx | can you configure asterisk to allow multiple calls to phone sip phone? |
00:44.48 | PatrickDK | ptg, probably just edit it |
00:44.58 | PatrickDK | unless couriour has some build in gui for it |
00:45.05 | PatrickDK | I know vpopmail doesn't |
00:46.43 | MikeJ[Jayden] | MichaelCat, you can use agi and set variables and check them after return |
00:46.44 | PTG1234 | hmm |
00:48.48 | MichaelCat | <MikeJ[Jayden]>,that is what i am looking at now, but not sure the cleanest way to do it |
00:49.20 | MichaelCat | it looks like i can only pass one arg with agi |
00:49.33 | *** join/#asterisk jterrero (~jt@ool-43576e0d.dyn.optonline.net) |
00:49.59 | MichaelCat | i pass the caller id and another arg, so I guess i could try to read the caller id from in the agi script |
00:52.27 | MichaelCat | i am looking for a sample agi script to pass variable back and forth and have the agi call my other shell script instead |
00:52.39 | *** join/#asterisk file[mac] (~jcolp@mctn1-3046.nb.aliant.net) |
00:53.33 | *** join/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
00:55.14 | MikeJ[Jayden] | http://www.voip-info.org/wiki-Asterisk+AGI |
00:56.31 | PTG1234 | can you hard reset and ipod? |
00:56.40 | file[mac] | yeah it's in the manual |
00:57.28 | *** join/#asterisk topping (~topping@cpe-24-210-82-196.columbus.res.rr.com) |
00:57.54 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
00:58.50 | PTG1234 | ugh |
00:58.52 | PTG1234 | its broke |
00:59.46 | RoyK | aweirnjcu |
01:01.16 | PTG1234 | hmm now a battery icon |
01:01.19 | *** part/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
01:01.24 | PTG1234 | i guess i'll leave it alone for an hour and hope it works when i come back :) |
01:04.46 | Mavvie | somebody here with an Patton 2977 card? |
01:10.42 | RoyK | <PROTECTED> |
01:15.22 | *** join/#asterisk file[laptop] (~file@mctn1-3046.nb.aliant.net) |
01:20.36 | PTG1234 | well looks like i am going back to the store already |
01:20.39 | PTG1234 | yippee |
01:20.49 | file[laptop] | why? |
01:20.58 | PTG1234 | its dead |
01:21.02 | PTG1234 | flashing folder icon with a ! |
01:21.08 | PTG1234 | no computers reconize it, etc |
01:21.36 | file[laptop] | use the iPod Updater |
01:21.37 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
01:22.00 | PTG1234 | doesn't see it |
01:22.18 | file[laptop] | ooh eek you broke it! |
01:22.32 | PTG1234 | yah |
01:22.34 | PTG1234 | explain to me how :) |
01:22.42 | PTG1234 | stupid f'cking ipod :) |
01:23.07 | denon | PTG1234: your wife get a #? :) |
01:23.11 | file[laptop] | http://www.macosxhints.com/article.php?story=20050309134113686 |
01:23.15 | file[laptop] | look at that! :p |
01:23.28 | PTG1234 | oh i think so denon, remind me in like 30 minutes, she ran to the store :) |
01:23.28 | denon | its .. an ipod .. apple product .. you deserve what ya get :) |
01:23.37 | denon | PTG1234 : im gonna be afk in like 20 seconds |
01:23.42 | denon | so if you remember .. remember ;) |
01:23.57 | file[laptop] | PTG1234: take a look at that page and give it a try |
01:24.01 | file[laptop] | and my Apple stuff works 100% :p |
01:24.05 | file[laptop] | Mac, iPod, and iSight |
01:24.47 | PTG1234 | file: to do any of that i need the pc to reconize it :) |
01:24.48 | PTG1234 | which it doesn't |
01:24.56 | file[laptop] | ah PC |
01:25.12 | file[laptop] | yeah it's all your fault it's broken too |
01:25.20 | PTG1234 | why because i own a pc? :) |
01:25.28 | file[laptop] | sure |
01:25.48 | PTG1234 | hah this notebook is so sweet :) no mac could compete :) |
01:26.02 | file[laptop] | meh I care not |
01:26.18 | PTG1234 | ok going to go jump in my pool and cool off fromy my stupid ipod problems |
01:26.19 | PTG1234 | :) |
01:26.44 | denon | /timer 1 1800 /msg PTG1234 Automated Message: did she get a tracking #? |
01:26.45 | denon | :) |
01:27.18 | file[laptop] | PTG1234: what did denon get from you? |
01:27.45 | *** join/#asterisk greyarea (~vgklsdklg@ip68-2-199-60.ph.ph.cox.net) |
01:28.21 | greyarea | yo anyone around? |
01:28.27 | file[laptop] | maybe. |
01:29.26 | greyarea | well Im having a bit of a problem getting musiconhold to work in freeBSD in meetme.I was thinking maybe I could paste my configs and someone could point out what Im doing wrong?pls |
01:29.32 | *** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net) |
01:29.52 | file[laptop] | using mpg123 0.59r? |
01:30.06 | file[laptop] | got a timing device? |
01:30.08 | greyarea | um I believe its changed let me see |
01:30.20 | file[laptop] | use 0.59r, and 0.59r only |
01:30.23 | greyarea | yeah using ztdummy |
01:30.37 | greyarea | the conf works but musiconhold does not |
01:30.55 | greyarea | ok let me see what player Im using that might be the problem |
01:31.04 | file[laptop] | you HAVE to use mpg123 0.59r |
01:31.11 | file[laptop] | only then will I further help you |
01:31.24 | greyarea | ok thanks |
01:32.04 | PTG1234 | isn't there something better them mplayer now |
01:32.16 | PTG1234 | besides you should really convert it to a raw file and play from it, so it doesn't use cpu cycles |
01:32.30 | Sedorox | mplayer or mpg123? |
01:32.30 | file[laptop] | mpg123 and mplayer are two different things ;) |
01:32.36 | Sedorox | lol |
01:32.37 | PTG1234 | er mpg123 |
01:32.39 | PTG1234 | whatever :) |
01:32.42 | PTG1234 | i am still seeing red from my ipod |
01:32.47 | Sedorox | hmmm |
01:32.54 | file[laptop] | I just use mpg123 because it always works for me and requires no effort usually |
01:32.56 | Sedorox | what happened to it? |
01:33.00 | *** join/#asterisk BitShopSteveR (icechat5@207.176.47.113) |
01:33.08 | PTG1234 | its dead |
01:33.10 | PTG1234 | and its less then an hour old |
01:33.25 | Sedorox | how is it deasd? |
01:33.42 | PTG1234 | flashing folder icon |
01:33.53 | Sedorox | hmmm |
01:34.03 | PTG1234 | http://www.archos.com/products/overview/gmini_xs200_tech_specs.html |
01:34.07 | PTG1234 | maybe i should just go buy that instead |
01:34.11 | PTG1234 | i got a bad taste from the ipod already |
01:35.03 | tzanger | PTG1234: they're not edible you twitt |
01:35.07 | Sedorox | http://www.ipodhacks.com/modules.php?op=modload&name=Forum&file=viewtopic&topic=3453&forum=2 |
01:35.10 | Mavvie | Status: Provisioned, Down, Active |
01:35.14 | Mavvie | go up you stupid link! |
01:35.39 | Mavvie | oh look! |
01:35.45 | Mavvie | Status: Provisioned, Up, Active |
01:35.47 | Mavvie | woohooo! |
01:35.48 | PTG1234 | yah i thinhk thats my issue |
01:35.54 | PTG1234 | the device_control folder is gone |
01:36.02 | PTG1234 | but the computer won't reconize it for the updater to work |
01:36.19 | *** join/#asterisk david (~david@gw-djc.davidcoulson.net) |
01:37.40 | Sedorox | hmm |
01:37.40 | RoyK | <PROTECTED> |
01:37.42 | greyarea | hmm,I'm using MPG123 but I can't tell what version. |
01:37.43 | RoyK | grr |
01:39.06 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
01:39.13 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
01:39.32 | *** join/#asterisk ozzy_cow (~industria@69.0.97.83.adsl.snet.net) |
01:39.56 | ozzy_cow | howdy |
01:42.00 | PTG1234 | ok well if it does ahve an explanation mark and youve had it for less than a year apple will take it back and usuallly send you a new one |
01:42.14 | PTG1234 | thats the onluy thing i can find :) |
01:43.19 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
01:43.47 | *** join/#asterisk ZX81_Laptop (~ZX81@222-153-115-253.jetstream.xtra.co.nz) |
01:44.00 | PTG1234 | ok screw it i am going to go buy the archos :) |
01:44.26 | ZX81_Laptop | anyone know why I get too many open files when ulimit is unlimited with 500 calls per second? |
01:44.43 | PTG1234 | yah b/c linux sucks under load |
01:44.45 | ozzy_cow | is someone running asterisk with "connect" voicepulse ? |
01:44.51 | ZX81_Laptop | ok |
01:44.53 | ZX81_Laptop | ta |
01:44.54 | ZX81_Laptop | :) |
01:44.56 | PTG1234 | use freebsd |
01:45.09 | cypromis | morn kiwiman |
01:45.12 | ZX81_Laptop | :) |
01:45.13 | cypromis | :P |
01:45.14 | ZX81_Laptop | morning |
01:45.38 | greyarea | yeah Voice pulse sucks |
01:45.45 | ozzy_cow | i keep getting Call rejected by 66.234.228.160: No such context/extension errors |
01:45.59 | ozzy_cow | greyarea: really, who do you recommend? |
01:46.01 | ZX81_Laptop | [root@pabx ~]# /usr/sbin/safe_asterisk: line 40: 16623 Aborted (core dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} |
01:46.06 | greyarea | Did you add the key? |
01:46.12 | ZX81_Laptop | :) |
01:46.14 | cypromis | ZX81_Laptop: now that wasn't very hard, was it ? |
01:46.26 | ZX81_Laptop | :) |
01:46.47 | greyarea | I hate how the DID I have with VP won't get CPN information. |
01:46.56 | *** join/#asterisk want561or772did (~aoiahsdf@68.71.213-36.atlsfl.adelphia.net) |
01:47.00 | ZX81_Laptop | also |
01:47.02 | ZX81_Laptop | Apr 17 13:40:17 WARNING[16623]: channel.c:505 ast_channel_walk_locked: Avoided initial deadlock for 'Local/6601@internal-1ce5,2', 10 retries! |
01:47.21 | ZX81_Laptop | and |
01:47.21 | ZX81_Laptop | Apr 17 13:40:17 WARNING[16623]: res_musiconhold.c:601 mohalloc: Failed to create pipe: Too many open files |
01:47.23 | ZX81_Laptop | :) |
01:47.42 | ZX81_Laptop | my poor asterisk box |
01:47.44 | ZX81_Laptop | :) |
01:47.51 | ozzy_cow | well, i can't even get voicepulse to work |
01:48.22 | greyarea | I couldnt either until I found out you have to add a key. |
01:48.23 | ozzy_cow | greyarea: whats CPN? |
01:48.31 | greyarea | Calling party Number |
01:48.40 | greyarea | whats behind private tags |
01:48.48 | greyarea | if someone tried to call with *67 |
01:48.49 | ozzy_cow | oh |
01:48.55 | greyarea | Not ANI but CPN |
01:49.05 | greyarea | Voip doesnt get ANI |
01:49.20 | ozzy_cow | what key do i need? |
01:49.35 | greyarea | I dont know I had someone else do that while I was out of town |
01:50.10 | greyarea | Anyone else with VPC should know what that is tho :/ |
01:51.20 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
01:51.50 | ozzy_cow | oh, i have the key |
01:52.09 | greyarea | werd you get it? |
01:52.10 | ozzy_cow | it's in /var/lib/asterisk/keys |
01:52.27 | ozzy_cow | http://connect.voicepulse.com/keys/voicepulse01.pub |
01:52.55 | greyarea | You see before I got this DID with VPC I had a acct with Nufone :) |
01:53.13 | greyarea | I like Nufone out of all the voip providers :) |
01:53.22 | ozzy_cow | i can't even get outgoing to work |
01:53.35 | ozzy_cow | i get "all circuits busy error" |
01:53.47 | ozzy_cow | and the logs show up with Call rejected by 66.234.228.160: No such context/extension |
01:53.50 | greyarea | You tring to dial in then out? |
01:54.00 | ozzy_cow | what do you mean |
01:54.17 | ozzy_cow | i didn't even set up dialing in yet :-) |
01:54.17 | greyarea | like you tring to call your pbx then enter an extension and dial of it? |
01:54.31 | greyarea | "off |
01:54.33 | ozzy_cow | hmm let me try that |
01:54.47 | ozzy_cow | btw, my asterisk box is behind nat, do i need to forward ports? |
01:54.56 | greyarea | boo |
01:55.09 | greyarea | I don't have NAT set-up :( |
01:55.38 | unterx | anyone have any experience with meetme? I defined the proper conference room in meetme.conf, but when I use it as an extension in extensions.conf i get "that is not a valid conference number" |
01:56.00 | greyarea | If your having problems with your extensions or iax.conf I can copy paste what I have. |
01:56.31 | greyarea | unterx:try this in you extensions.conf |
01:56.31 | greyarea | exten => 1,1,MeetMeCount,1 |
01:56.31 | greyarea | exten => 1,2,MeetMe,1 |
01:56.43 | docelmo | Who said you dont get ANI with VOIP? |
01:57.16 | Chuji | ~ANI |
01:57.17 | bugbot | ANI is assigned nothing and reported nothing. |
01:57.37 | sivana | ~ani |
01:57.38 | bugbot | ani is assigned nothing and reported nothing. |
01:57.46 | greyarea | and in your meetme.conf |
01:57.46 | greyarea | conf => 1, |
01:57.57 | greyarea | voip doenst get ani |
01:58.07 | Chuji | jbot ani is Automatic Number Identification Systems |
01:58.08 | jbot | okay, Chuji |
01:58.10 | greyarea | SO your telling me that it gets both CPN and ANI |
01:58.15 | greyarea | your crazy |
01:58.21 | docelmo | So grey your telling me if I recieve a call via voip I will not get the ANI? |
01:58.29 | file[laptop] | you can get lots of stuff |
01:58.40 | greyarea | what provider and on what switch is your DID coiming off of? |
01:59.03 | greyarea | thats what determines it |
01:59.40 | docelmo | No provider should block ANI unless someone did a *67 or something. You should always get ANI or CPN.. |
01:59.42 | greyarea | I wish that Nufones shit would send along the OCI digits or also know as the ANI II digits to me :( |
01:59.59 | greyarea | ANI and CPN are 2 different things |
02:00.22 | docelmo | Ani is the calling party number. |
02:00.24 | unterx | greyarea: what is the point of 1, in meetme.conf? |
02:00.36 | greyarea | just for pressing 1 |
02:00.49 | unterx | oh, just for testing |
02:01.01 | greyarea | Now traditionaly ANI is the CPN but you can change the CPN to generate a different callerid |
02:01.07 | docelmo | I deal with large scale voip deployments and I always get ANI I dont always get OLI but just the same.. |
02:01.44 | greyarea | So if you call me and I have my phone forwarded to tom,tom will see your CPN (Calling party number. |
02:01.46 | greyarea | but if |
02:02.05 | greyarea | you call me and I forward my line to NPA5551212 they wil see my number b/c IM the ANI |
02:02.19 | greyarea | I originated the call to them and Im the BTN the Builling telephone number |
02:02.32 | greyarea | But your the CPN |
02:02.40 | greyarea | theres a difference my friend |
02:03.09 | docelmo | In this instance this is true.. However ANI is all anyone looks at and normally the orignal ANI is passed along if via tandem |
02:03.29 | greyarea | VOip gets CPN |
02:03.33 | greyarea | not ANI |
02:04.11 | docelmo | sigh.. |
02:04.17 | file[laptop] | docelmo: don't bother |
02:04.17 | greyarea | ofcourse if someone puts private tags on their line in most carriers you can see their number but that does not mean that it is ani its justy CPN |
02:04.25 | file[laptop] | it's not worth it. |
02:04.30 | file[laptop] | I gave up long ago |
02:04.34 | greyarea | If you dont beleive me test it yourself |
02:04.35 | *** join/#asterisk doonut (~chatzilla@adsl-67-123-166-100.dsl.sntc01.pacbell.net) |
02:05.07 | docelmo | Dude I run 20 * boxes load balanced.. My front end is cisco or direct VOIP and I get ANI and OLI |
02:05.28 | greyarea | ok one sec |
02:05.32 | greyarea | we can test this |
02:05.46 | Silik0n | jesus fucking christ argue over stoopid shit why dont you |
02:05.46 | docelmo | greyarea, what is your area of expertise in VOIP? |
02:06.12 | docelmo | I think its funny.. Cause ANI and CPN are the same.. |
02:06.12 | sivana | my area of expertise is breaking shit |
02:06.17 | greyarea | Im going to call passing ANI and a differnt CPN let me know what you see. |
02:06.45 | docelmo | Im not gonna see shit cause I am in a chat channel not on your box ass.. |
02:07.03 | docelmo | file knows I am right.. :) |
02:07.20 | file[laptop] | all I'm saying is you can get lots of things |
02:07.29 | docelmo | Anywho.. I am the Senior Network Data/Voice engineer for my company so I kinda know what I am talking about.. ANI and CPN are one in the same.. |
02:07.36 | greyarea | I know about this man,you cant get ANI only CPN |
02:07.50 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
02:07.51 | greyarea | they always hire noobs :/ |
02:07.53 | docelmo | ANI IS CPN! ARGH! Im going to eat.. |
02:07.58 | greyarea | No it is not |
02:08.09 | sivana | ding ding ding |
02:08.22 | newl | round two... |
02:08.24 | greyarea | ANI is the BIlling number the number that orginated the call CPN is the person calling |
02:08.44 | file[laptop] | Silik0n: So how's life treating you? |
02:08.46 | greyarea | I only wish voip could get ANI |
02:09.15 | file[laptop] | *cough* it can *cough* |
02:09.19 | file[laptop] | I didn't say that |
02:09.24 | sivana | who said that |
02:09.28 | file[laptop] | not me!!! |
02:10.11 | greyarea | So.... your telling me that you can spoogf ANI? HAha,no you cant you can only spoof CPN. |
02:10.19 | sivana | lol |
02:10.28 | file[laptop] | did I say that? |
02:10.29 | greyarea | ANI is the originating number. |
02:10.36 | greyarea | not the calling party number |
02:10.44 | greyarea | anywho |
02:10.52 | greyarea | youll find out when you test |
02:11.16 | newl | heh cpn? that's a new acronym to me. :) |
02:11.36 | Sedorox | Looks.. its a grey area... so leave it alone.... |
02:11.37 | Sedorox | O:-) |
02:11.54 | drumkilla | file[laptop]: !!!!!!!!!!!!! |
02:11.59 | file[laptop] | drumkilla: !!!!!!!!!!!!! |
02:12.12 | drumkilla | how you doin' ? |
02:12.23 | file[laptop] | I'm peachy, you? |
02:12.31 | drumkilla | goooooood |
02:12.38 | greyarea | If you think you get ANI,call from your landline thru 10102880 (it makes the ANI a NPA that the call was homed out of) and does a anifial but the info is still there call yourself and see if you see that 3 digit number |
02:12.52 | greyarea | you wont |
02:13.26 | greyarea | and or call yourself thru voicepulse passing CPN that you choose and see if it reads it or if it reads the ANI the VPC has set for their outgoing calls |
02:13.35 | greyarea | you are going to see the CPN that you choose |
02:13.35 | file[laptop] | voicepulse? pfft |
02:14.07 | greyarea | nufone passes a anifail but its a 3 digit number and you wont see that either |
02:14.29 | greyarea | with voip there is always a ANI behind your CPN |
02:14.42 | greyarea | that is differnt |
02:14.53 | file[laptop] | who has a landline and wants to test this? |
02:15.04 | greyarea | I can call anynumber |
02:15.10 | *** join/#asterisk _solstice_ (~solstice@dsl-cap-209-5-169-205-cgy.nucleus.com) |
02:15.15 | greyarea | who wants me to call their box? |
02:15.36 | greyarea | I will pass a different ANI than the CPN and you can tell me what you see the CPN or the ANI |
02:15.39 | file[laptop] | just call that and hangup, that's all I need |
02:16.00 | file[laptop] | I'm curious to see what the equipment will send me |
02:16.20 | greyarea | ok Ill pass 4803915000 as the CPN and 2124007300 as the ANI |
02:16.27 | file[laptop] | k |
02:16.32 | _solstice_ | Anyone here ever setup a asterisk box behind a cisco router on a nated network? Trying to get mine to work behind either a Cisco 837DSL / 3640 .. |
02:16.51 | *** join/#asterisk voiper (~none@pcp0010158348pcs.eatntn01.nj.comcast.net) |
02:17.35 | greyarea | ok called |
02:17.37 | file[laptop] | 'datz enough |
02:17.40 | greyarea | thats a voip number right? |
02:17.52 | greyarea | oh btw I can pull tollfree info :) |
02:17.53 | greyarea | heh |
02:18.44 | file[laptop] | that was TDM |
02:18.53 | greyarea | ? |
02:18.55 | file[laptop] | going through a Lucent APX |
02:19.03 | greyarea | ah |
02:19.07 | greyarea | what did it say? |
02:19.09 | file[laptop] | I wanted to see if it would give me the ANI in the SIP messages |
02:20.15 | greyarea | Im going to smoke msg me what it said pls |
02:20.24 | file[laptop] | nah it had the 480 |
02:20.32 | file[laptop] | the SIP messages just looked a little odd |
02:20.39 | file[laptop] | indented weirdly... |
02:24.50 | greyarea | right see |
02:24.53 | greyarea | its simple |
02:25.10 | file[laptop] | but anyway, depending on the equipment you can get both |
02:25.15 | file[laptop] | you can get LOTS of things |
02:25.19 | greyarea | if a call comes straight from a PSTN line The ANI will generate the CPN and the CPN will generate the Callerid |
02:25.56 | greyarea | if a call is forwarded then the CPN iwll be diffent than the ANI or if a call comes from a PBX the CPN will be different if you change it |
02:26.07 | Silik0n | whores |
02:26.11 | Silik0n | misfire |
02:26.33 | greyarea | well no it depends on the backbone provider sending the information along to the reseller and if the reseller sends that info to you. |
02:26.39 | *** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com) |
02:26.53 | newl | If it is forwarded, it's up to the exchange what to do with it. Some are configured to presend the B party diverting, some present the A party. |
02:27.02 | newl | s/end/ent/ |
02:27.17 | sivana | so then basically CPN = ANI.. right |
02:27.28 | greyarea | well right that depends on how your carreir handles forwarded calls |
02:27.30 | sivana | lol |
02:27.30 | greyarea | no |
02:27.32 | newl | sivana: no. |
02:27.40 | sivana | I'm just stirring up the pot |
02:27.43 | greyarea | heh |
02:27.44 | newl | :) |
02:27.47 | greyarea | no need to fight |
02:27.51 | sivana | hehe |
02:28.25 | greyarea | but traditionaly ANI is the CPN but CPN can always be changed. |
02:28.37 | FuriousGeorge | so this snd_intel8x0 driver. i cant seem to find an option in the kernel for it. i found under OSS "Intel ICH 8xx" but i doubt thats it, since its for oss. anyone got an idea what option in the kernel i should be looking for |
02:28.37 | greyarea | to be something different than the ANI |
02:28.56 | FuriousGeorge | generic inte/nvidia/sis/blah support |
02:28.57 | greyarea | anywho tho |
02:28.59 | newl | I can program up a 6 line hunt group that'll present the 5 auxillary numbers as the prime for presentation purposes. That doesn't mean the ANI will be the same. |
02:29.10 | greyarea | right |
02:29.13 | FuriousGeorge | whoops im in the wrong channel |
02:29.17 | greyarea | a "lead" number? |
02:29.28 | newl | FuriousGeorge: snd_intel8x0 is an ALSA module. |
02:29.31 | sivana | right, but then ANI = CPN though |
02:29.38 | greyarea | in ss7 alot of info is being sent |
02:29.46 | sivana | and E = MC2 |
02:29.46 | greyarea | like 4 different numbers |
02:30.10 | sivana | and my ass = going home |
02:30.14 | greyarea | ok yo can you help me know with my musiconhold pls :) |
02:30.14 | FuriousGeorge | newl: so its not in the kernel |
02:30.46 | newl | FuriousGeorge: Not by default, no. |
02:31.02 | *** join/#asterisk iq{tablet} (~iq@63-230-45-178.omah.qwest.net) |
02:31.10 | newl | FuriousGeorge: Well, not vanilla kernel anyway. It may be provided by the distributor though. |
02:31.24 | FuriousGeorge | yeah, i think its in my gentoo kernel |
02:31.37 | FuriousGeorge | Intel/SiS/nVidia/AMD/ALi AC97 Controller (NEW) |
02:31.51 | FuriousGeorge | i have it built in, but still nothing |
02:35.00 | greyarea | ok Im in freeBSD and I have my extensions.conf like this for music on hold while in a conference |
02:35.00 | *** join/#asterisk jbAU (~johnblade@c210-49-42-214.rochd2.qld.optusnet.com.au) |
02:35.15 | greyarea | [conf] |
02:35.15 | greyarea | extensions.conf |
02:35.15 | greyarea | exten => 1,1,MeetMeCount,1 |
02:35.15 | greyarea | exten => 1,2,MeetMe,1 |
02:35.24 | greyarea | shit |
02:35.35 | greyarea | exten => 1,1,MeetMeCount,1 |
02:35.35 | greyarea | exten => 1,2,MeetMe,1|M |
02:35.47 | greyarea | and in my musiconhold.conf |
02:35.56 | greyarea | musiconhold.conf |
02:35.56 | greyarea | [classes] |
02:35.56 | greyarea | default => quietmp3:/usr/local/share/asterisk/mohmp3 |
02:36.13 | greyarea | and my zapata.conf [channels] |
02:36.13 | greyarea | musiconhold=default |
02:36.14 | elric | greyarea, use pastebin.ca to paste stuff |
02:36.17 | *** join/#asterisk Tuplink_ (~dsfsk@68-232-92-239.chvlva.adelphia.net) |
02:36.34 | Tuplink_ | i need to dial a user in another context how do i do that? |
02:36.44 | greyarea | werd |
02:36.48 | greyarea | What am I doing wrong? |
02:37.24 | greyarea | The conf works fine but the musiconhold isnt when the only caller. |
02:37.42 | greyarea | and mp3's play fine |
02:37.45 | Tuplink_ | mpg123 is crap |
02:38.04 | Tuplink_ | do a ps -aux | grep mpg123 |
02:38.15 | Tuplink_ | and see what its chewing of you cpu |
02:39.21 | greyarea | thats for FreeBSD? |
02:39.38 | Tuplink_ | im using freeBSD |
02:39.47 | Tuplink_ | and MOH works interminitly |
02:40.04 | Tuplink_ | its mpg123 that is the prob |
02:40.10 | greyarea | oh ok,I swear man I have taken this box offline somemany times using linux commands heh |
02:41.15 | greyarea | my extensions shit is correct though right? |
02:41.18 | greyarea | exten => 1,1,MeetMeCount,1 |
02:41.18 | greyarea | exten => 1,2,MeetMe,1|M |
02:41.25 | greyarea | how I did the command? |
02:41.32 | greyarea | |M |
02:41.41 | greyarea | or option whatever |
02:42.10 | *** join/#asterisk Legend (~Legend@24.244.142.134) |
02:43.51 | greyarea | Tuplink:What player are you using? |
02:44.12 | Tuplink_ | player for? |
02:44.16 | greyarea | mp3 |
02:44.26 | Tuplink_ | not using any... that i know of |
02:44.30 | greyarea | oh |
02:44.53 | Tuplink_ | im still setting up my box... diagnosed hte prob dont know how to fix it |
02:45.11 | greyarea | I have like over like 700 mp3's that play fine on my box |
02:45.19 | greyarea | thats what I use it for a media player |
02:45.24 | greyarea | by phone :) |
02:45.38 | greyarea | but its in the conference I cant get it to work. |
02:45.40 | Tuplink_ | oh... ok |
02:45.49 | Tuplink_ | music on hold work? |
02:45.51 | *** join/#asterisk kapanol (~edit@203-118-142-32.pppoe.ihug.co.nz) |
02:45.54 | greyarea | Im pretty sure I have the options setup correctly but Im not sure |
02:46.00 | greyarea | right |
02:46.15 | Tuplink_ | no idea then sorry |
02:46.30 | greyarea | oh ok thx though |
02:46.35 | Tuplink_ | im having a prob of my own... |
02:46.40 | greyarea | whats your problem? |
02:47.29 | greyarea | Maybe its something I have done. |
02:47.50 | Tuplink_ | i have a switchboard... if i dail it from internal it works great... if i dial to it from FWD it works.... but if i use a service to dial a FWD ext it dosnt work get hte menues and all just dosnt recive the user input |
02:48.09 | greyarea | hmm |
02:48.17 | greyarea | I want a switchboard :D |
02:48.26 | Tuplink_ | VIM |
02:48.34 | Tuplink_ | i call it a switchboard |
02:48.49 | *** join/#asterisk tonya (~tonya@203.177.148.153) |
02:48.55 | Tuplink_ | any ideas |
02:49.04 | greyarea | so dtmf isnt passing when you call from something else? |
02:49.26 | Tuplink_ | dtmf? |
02:49.33 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
02:49.36 | greyarea | dual tone multifreq |
02:49.38 | greyarea | um |
02:49.44 | Tuplink_ | sounds rite |
02:49.44 | greyarea | buttons on your phone |
02:49.48 | Tuplink_ | nope... |
02:49.51 | greyarea | oh |
02:50.08 | Tuplink_ | i have tryed it from a land line and a cell |
02:50.19 | tonya | hi huys.....do you know where i could get a developers kit or any tutorials about developing SIP application in windows to connect to asterisk |
02:50.22 | greyarea | and what happends? |
02:50.26 | Tuplink_ | i orderd 2 FXO cards yesterday |
02:50.27 | greyarea | it doenst respond? |
02:50.43 | Tuplink_ | nope |
02:50.51 | Tuplink_ | you have fwd? |
02:50.54 | greyarea | does it answer but wont listen to you pressing anything? |
02:50.58 | greyarea | no |
02:51.08 | Tuplink_ | it anwsers and you can hear it |
02:51.12 | Tuplink_ | but cant respond |
02:51.32 | greyarea | sounds like it has to do with DTMF not passing |
02:51.43 | Tuplink_ | so how do i get that to pass? |
02:51.52 | greyarea | I havent had that problem |
02:51.58 | greyarea | Let me google a little for you |
02:52.04 | Tuplink_ | or is it the service that im using |
02:52.09 | brc__ | http://www.dgp.toronto.edu/~ravin/videos/uist2003_tabletop.mpg |
02:52.53 | newl | tonya: the SIP RFCs are probably a good start, more info here: http://www.voip-info.org/tiki-index.php?page=SIP |
02:53.08 | greyarea | Tuplink look around here : http://www.google.com/search?hl=en&q=DTMF+not+passing+when+I+call+up+my+asterisk+box&spell=1 |
02:53.27 | greyarea | there is some things you can change,I just never had to. |
02:53.49 | greyarea | Sorry I couldnt be more help. |
02:55.21 | _solstice_ | anyone know how to setup a nat'ed asterisk box behind a cisco router? |
02:55.43 | Tuplink_ | its ok... |
02:56.00 | Tuplink_ | _solstice_ i want to know how to to... |
02:56.06 | Tuplink_ | what ports have to be mapped |
02:56.19 | Tuplink_ | 5060 for SIP |
02:56.22 | Tuplink_ | rite |
02:57.26 | _solstice_ | yea .. exactly , and a range in the 16k - 20k range .. but for registration etc.. i can't seem to get that to work, unless i use the router as a full VPN unit, which is a extra overhead here .. |
02:59.04 | Tuplink_ | i can get a SIP phone to register to my box... w/o hte 16k-29k range |
02:59.39 | Tuplink_ | and can call... while watching the debugscreen and see asterisk respond but client gets no voice |
03:00.29 | Tuplink_ | greyarea could it be a NAT ISSUE? |
03:00.40 | *** join/#asterisk zignig (~zignig@the.interthingy.com) |
03:00.43 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr) |
03:03.29 | _solstice_ | Tuplink: hmm .. maybe it is possible to setup a "keep state" rule for cisco to track the connections and pass the data out correctly |
03:03.50 | Tuplink_ | i have a keep state on mine. |
03:03.59 | Tuplink_ | mine is a openbsd w/ PF |
03:04.34 | *** join/#asterisk toddf (amrthhyfix@default.fries.net) |
03:04.58 | _solstice_ | hmm .. openbsd should work .. Cisco is the one that is causing grief .. i knew a few people who made the openbsd thing work together with authpf shell for login, and call routed .. |
03:06.18 | Tuplink_ | hum... i have had openBSD on my firewall for 3 days and asterisk on FreeBSD for 4 days |
03:06.19 | Tuplink_ | hehe |
03:09.30 | _solstice_ | the problem seems to be cisco's don't support even states when routing .. kinda blows .. specially when you deal with DMZ's, etc ... |
03:09.40 | *** part/#asterisk zignig (~zignig@the.interthingy.com) |
03:09.55 | Tuplink_ | i need to make a DMZ with my pd |
03:09.56 | Tuplink_ | pf |
03:10.20 | Tuplink_ | hummm.... |
03:10.31 | Tuplink_ | i dont rember much about NAT and cisco |
03:10.31 | _solstice_ | quite simple there .. suggestion having seperate nic all together in machine .. then DMZ becomes easy to setup .. |
03:10.47 | Tuplink_ | not really |
03:10.56 | Tuplink_ | just use ip's :) |
03:11.11 | Tuplink_ | i dont want secure rite now i want it to work |
03:11.25 | Tuplink_ | then secure |
03:11.38 | _solstice_ | hehe ... i just read the faq pdf on openbsd's site .. had a few examples that made this work quite nicely .. hehe |
03:12.05 | Tuplink_ | hum.... |
03:12.10 | Tuplink_ | whats the link |
03:12.40 | Tuplink_ | FAQ/PF cause i have that printed |
03:13.20 | *** join/#asterisk coppice (~chatzilla@235.193.17.210.dyn.pacific.net.hk) |
03:13.44 | greyarea | Tuplink: I dont know,mine doesnt have NAT set-up :/ |
03:13.52 | Tuplink_ | hum.... |
03:13.57 | greyarea | Im just behind a cisco router |
03:14.00 | _solstice_ | ftp://ftp.openbsd.org/pub/OpenBSD/doc/pf-faq.pdf |
03:14.03 | Tuplink_ | im on cable ;) |
03:14.24 | Tuplink_ | than whats the prob? |
03:14.27 | _solstice_ | greyarea: you setup natting and are able to access your asterisk box from outside ? |
03:14.29 | greyarea | yeah I have a DS3 piped into mine w00t :D |
03:14.50 | greyarea | no man,nat is not set-up |
03:14.53 | _solstice_ | 100MB fiber link here and DSL circuit as backup .. hehe |
03:15.02 | greyarea | heh |
03:15.05 | _solstice_ | you just route live ip ? |
03:15.11 | greyarea | yeah |
03:15.26 | greyarea | your handle looks familar |
03:15.35 | _solstice_ | routable ip .. hmm .. greyarea: do you know if it is possible on cisco's to do a live route aswell as a nat ? |
03:15.53 | sivana | yes |
03:16.01 | sivana | which router |
03:16.10 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
03:16.14 | sivana | I'm doing it on the 1710 and 1720s |
03:16.14 | greyarea | I dont know man sorry |
03:16.39 | _solstice_ | sivana: example plz .. . Got a 3640 (100MB intake) and a Cisco 837 DSL .. so atm -> ether .. |
03:17.14 | sivana | you should be able to just add the subnet to the fastether |
03:17.27 | sivana | but I'm not sure about atm -> ether |
03:17.44 | *** join/#asterisk theHub (~karl@ool-182cba82.dyn.optonline.net) |
03:18.12 | _solstice_ | don't worry about the atm -> ether part, basically dsl circuits are nothing more then atm protocol based circuits, ... |
03:18.13 | want561or772did | i modified MeetMe to allow a response timeout option, t |
03:18.43 | want561or772did | now all my pbx menus throw callers into meetme musiconhold with a responsetimeout |
03:18.47 | want561or772did | i call this musiconholdhell |
03:19.21 | want561or772did | if they press a button, though, they are pulled out of hell and advanced to a new extension |
03:19.49 | sivana | _solstice_: what subnet are you running NAT.. 10.x.x.x?\ |
03:20.09 | _solstice_ | 192.168.8/24 |
03:20.21 | sivana | ok.. and that's on fastether? |
03:20.33 | sivana | faste0 |
03:20.58 | _solstice_ | internal .. outside i get 2 live Public IP's pushed to the outside interface .. 1 for nating one for DMZ |
03:21.30 | sivana | you'll need another subnet then, but you put that one on the fastether |
03:22.15 | _solstice_ | yea that would be fine .. i could use a 10.x.x.x for the DMZ .. but that would be nated then .. which i guess is what causes most of the grief with asterisk box.. |
03:22.27 | sivana | you'll need a subnet from your upstream |
03:23.02 | sivana | let's a /29, take the first usable and put it on your fastether |
03:23.09 | sivana | that becomes your gateway basically |
03:24.32 | greyarea | latez ppl ; also strom put up his ANI II and CPN readback number 1-877-TADA-LOL it reads II digits and CPN NOT ANI |
03:24.35 | greyarea | peace |
03:24.37 | *** part/#asterisk greyarea (~vgklsdklg@ip68-2-199-60.ph.ph.cox.net) |
03:26.12 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
03:28.24 | _solstice_ | hmm ..ok ill see what my provider can do ... |
03:30.06 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
03:33.55 | *** join/#asterisk cjk (~cjk@80.92.75.232) |
03:34.37 | cjk | hi, asterisk goes down every 4 jours. i just traced the whole thing using strace and i see that i have too many openfiles. but it is 05:34 and there is no activity here |
03:34.40 | cjk | any idea |
03:35.33 | _solstice_ | cjk, look at your processes and see if you have anything in the inittab that respawns and opens a new file .. btw, where are you at ? |
03:36.39 | *** part/#asterisk doonut (~chatzilla@adsl-67-123-166-100.dsl.sntc01.pacbell.net) |
03:38.12 | cjk | _solstice_, europe luxembourg |
03:38.46 | cjk | _solstice_, whats the command to check openfiles and the realted processes, could be attack too? |
03:39.34 | Luhiwu | cjk, try "lsof" |
03:39.36 | ManxPower | lsof |
03:40.13 | unterx | what the hell, when I call a pstn phone they can not hear me |
03:40.13 | cjk | damn i get this problem really since we have a bigger presentation in germany |
03:40.16 | unterx | this is getting ridiculous |
03:40.51 | cjk | unterx, who do you blame |
03:41.37 | unterx | myself, obviously |
03:42.20 | cjk | ah ok |
03:42.27 | *** join/#asterisk MarkS__ (~MarkS@cpe-70-112-81-84.austin.res.rr.com) |
03:42.40 | iq{tablet} | hi, anyone using AMP... http://www.pastebin.com/272261 |
03:43.13 | cjk | is this normal? voipgate1:~# lsof | grep asterisk | wc |
03:43.13 | cjk | <PROTECTED> |
03:44.01 | MarkS__ | so- |
03:44.06 | _solstice_ | seems ok ... |
03:44.06 | _solstice_ | [root@asterisk1 etc]# lsof | grep asterisk | wc |
03:44.07 | _solstice_ | <PROTECTED> |
03:44.38 | cjk | _solstice_, ok i see |
03:44.50 | MarkS__ | anyone from the 512 in here? -- whats a good voip service that is SIP / IAX compatible |
03:45.51 | cjk | _solstice_, whats the maxopenfiles your system allows |
03:46.24 | iq{tablet} | I'm trying to patch snapdsp for AMP/Asterisk. It is failing. Any idea? |
03:47.57 | iq{tablet} | Makefile.rej : http://www.pastebin.com/272261 |
03:56.40 | MarkS__ | anyone from the 512 in here? -- whats a good voip service that is SIP / IAX compatible |
03:57.32 | cjk | whats 512 ? |
03:57.59 | Sedorox | area codwe? |
03:58.29 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
03:58.30 | *** mode/#asterisk [+o bkw_] by ChanServ |
03:59.17 | MarkS__ | 512 |
03:59.22 | MarkS__ | Central Texas |
03:59.31 | MarkS__ | Austin, TX to be exact, the capitol of texas |
04:04.25 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
04:13.02 | *** join/#asterisk barshad (kkhhaannuu@202.134.140.30) |
04:14.48 | barshad | Hello, can some one help me installing OpenH323 with asterisk ? |
04:25.51 | want561or772did | has anyone signed this digium disclaimer and if so did you really fax it? can't i email it or some crap |
04:27.57 | Moonwick | emails aren't legally binding |
04:28.19 | want561or772did | what's the difference between faxing an image and emailing an image |
04:28.44 | Moonwick | oh, I wasn't thinking about that. that might be good enough. |
04:29.11 | want561or772did | yeah, plus i think congress made "digital" signatures legal so that makes a lot of web forms legally binding somehow |
04:29.28 | want561or772did | maybe a text email would be, too, if its legitimacy were believable |
04:30.17 | *** join/#asterisk Rick_Hunter (~rhunter@01-199.008.popsite.net) |
04:32.04 | want561or772did | well besides that, is there a way to transmit a fax to digium via their asterisk gateway |
04:32.13 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
04:34.21 | newl | legally binding emails are dependant upon the country in which you reside. :) |
04:37.25 | want561or772did | how many countries besides the us would humor a lawsuit against digium for patches submitted |
04:38.09 | want561or772did | i mean of the countries digium wants to do business with |
04:38.29 | FuriousGeorge | newl: got the ole sound working. took it out of kernel and made everything as modules |
04:41.50 | FuriousGeorge | on a realted note, i am all proud of myself for registering my x-lite softphone with a user on *, but sipphone tells me i dial an unknown user, even when i dont. something must be wrong with this line...exten=>_.,1,dial(sip/sipout/$(EXTEN),20) |
04:42.41 | FuriousGeorge | but what can it be? i bet its the $(EXTEN), part ;) |
04:45.05 | FuriousGeorge | dont hold back, just shout it out when you know it |
04:51.50 | FuriousGeorge | god this place is quite for a change |
04:52.50 | |Vulture| | ammm I have port 4569 forwarded... I can get IAX2 calls but not dial IAX2 calls.. anyone have any idea? |
05:00.07 | _solstice_ | Anyone here use eyeBeam with Asterisk? |
05:04.45 | ManxPower | |Vulture|: Two Asterisk servers or an asterisk server an IAX phone? |
05:06.25 | |Vulture| | ManxPower: SIP phone connected going to * server behind NAT |
05:06.40 | |Vulture| | SIP phone is behind NAT... but I have 7 of these and all but this one doesn't work |
05:07.54 | ManxPower | |Vulture|: Please put your drink down and step away from the keyboard. |
05:08.05 | ManxPower | SIP does not use port 4569 |
05:09.22 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l03v-14-146.d4.club-internet.fr) |
05:09.25 | |Vulture| | I have 5000-5063 and 10000-20000 forwarded too... Im not that drunk yet |
05:10.05 | |Vulture| | ManxPower: and I can make outbound SIP calls from that line, but not outbound IAX |
05:11.26 | ManxPower | |Vulture|: For the IAX calls, is it two Asterisk servers or an asterisk server an IAX phone? |
05:15.46 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
05:15.46 | *** mode/#asterisk [+o bkw_] by ChanServ |
05:18.03 | *** join/#asterisk wolfson (~hehe@cpe-68-187-189-084.man.nc.charter.com) |
05:22.21 | |Vulture| | ManxPower: its an * server dialing out to POTS over VPC |
05:22.26 | |Vulture| | and i know it works |
05:22.31 | |Vulture| | its just being strange |
05:23.58 | *** join/#asterisk padf00t (WinNT@210.213.76.242) |
05:49.10 | PTG1234 | anyone alive? |
05:50.16 | want561or772did | i just sent an internet fax of my gimped signature to digium |
05:50.47 | want561or772did | maybe they'll gimp my signature onto other documents |
05:51.18 | PTG1234 | http://www.sonystyle.com/is-bin/INTERSHOP.enfinity/eCS/Store/en/-/USD/SY_BrowseCatalog-Start;sid=aT4h7CuDZkwhy2nikDUr52SN-8Puhg203JM=?CategoryName=pa_DigitalMusicPlayers_Network&Dept=audio |
05:51.22 | PTG1234 | what color should i get? :) |
05:51.26 | Moonwick | you didn't need your power of attorney anyway, did you? |
05:51.42 | PTG1234 | its between black and red i am thinking :) |
05:52.04 | want561or772did | nah i have no property or feeding tube dispositions |
05:52.12 | want561or772did | what is it |
05:52.24 | PTG1234 | did you go to url? |
05:52.29 | want561or772did | can't |
05:52.40 | PTG1234 | ah how come |
05:52.42 | want561or772did | it's too long |
05:52.43 | Moonwick | I'd get a white one that said "iPod" |
05:52.59 | PTG1234 | dude this sony is smaller then the ipod :) |
05:53.03 | PTG1234 | and its 30 hours of battery life |
05:53.04 | want561or772did | i found a $100 compact flash mp3 kit |
05:53.05 | PTG1234 | compared to 10 on the ipod |
05:53.15 | Moonwick | but it's sony |
05:53.22 | PTG1234 | sony makes good stuff :) |
05:53.30 | Moonwick | sony completely missed the boat when it comes to music players |
05:53.34 | want561or772did | how much does it cost |
05:53.38 | PTG1234 | they did until this one |
05:53.39 | PTG1234 | supposably :) |
05:53.40 | Moonwick | no AAC support, weird MP3 support... |
05:53.47 | PTG1234 | it actually nativelys upports mp3 now |
05:53.47 | want561or772did | i hate how sirc linewraps urls |
05:54.22 | Moonwick | but doesn't it require you to obfuscate them somehow when you load them? |
05:54.34 | PTG1234 | no |
05:54.36 | PTG1234 | not this one |
05:54.40 | Moonwick | eh |
05:54.41 | PTG1234 | it just came out |
05:55.07 | Moonwick | weird, I thought I read differently |
05:55.58 | PTG1234 | your thinking of the HD1 |
05:56.03 | PTG1234 | this is the brand new HD3 :) |
05:56.15 | Moonwick | ah |
05:56.25 | Moonwick | well, I still dislike sony crap :) |
05:56.26 | want561or772did | there needs to be an ogg-only CF player |
05:56.31 | PTG1234 | suppose to be one of the highest quality mp3 players |
05:56.42 | WilliamK | morning ya'll |
05:56.43 | WilliamK | =) |
05:56.53 | PTG1234 | hah did you just get up? |
05:57.07 | WilliamK | PTG, nope =) |
05:57.22 | PTG1234 | hehe then why the morning :) |
05:57.28 | WilliamK | after 12am |
05:57.29 | WilliamK | =) |
05:57.37 | PTG1234 | heh ah, its 11 here :) |
05:57.39 | PTG1234 | still night |
05:57.42 | PTG1234 | i have had the shittiest day |
05:57.46 | PTG1234 | first i buy an ipod |
05:57.51 | PTG1234 | then when i get home it breaks immediately |
05:57.57 | PTG1234 | then i go to get a sony hd1 |
05:58.01 | PTG1234 | and they only have the floor model |
05:58.02 | PTG1234 | so i buy it |
05:58.06 | PTG1234 | and its missing the docking station |
05:58.10 | PTG1234 | so i can't even use it :) |
05:58.15 | PTG1234 | this is after going to 20,000 places |
05:58.20 | PTG1234 | b/c every store carries different mo3 players |
05:58.27 | PTG1234 | so fuck it i am ordering one :) |
05:58.45 | want561or772did | you need to solder your own |
05:58.57 | PTG1234 | not lik ei can make my own as small as that sony :) |
05:59.27 | want561or772did | yes you can look http://www.walrus.com/~raphael/html/mp3.html |
05:59.48 | want561or772did | but you'll need a shed containing a few tons of plastic mold injector machinery for the shell |
06:00.10 | PTG1234 | yah and it still won't be as small |
06:00.14 | PTG1234 | i don't think you know how small that sony is :) |
06:00.18 | PTG1234 | for a 20gb player |
06:01.05 | want561or772did | yes and if you factor in the battery.. this one uses one AA |
06:01.34 | PTG1234 | dude your nuts :) |
06:01.39 | PTG1234 | plus this one has 30hours of battery life |
06:01.44 | PTG1234 | no AA battery is gonna give you that :) |
06:02.19 | want561or772did | how densely can u glue 2 gig CF cards together |
06:02.26 | want561or772did | probably pretty tight |
06:02.57 | barshad | can any one help me installing OpenH323 with asterisk ? |
06:03.07 | want561or772did | i never did that |
06:05.22 | *** join/#asterisk JerJer (~JerJer@12.173.204.122) |
06:05.43 | newl | PTG1234: should've just bought a pda instead. :) |
06:05.51 | PTG1234 | newl: i got 3 pdas :) |
06:06.04 | PTG1234 | i did jus tbuy a new one 3 days ago |
06:06.40 | PTG1234 | just got an PPC 6600 damn is it sweet :) |
06:07.13 | newl | hah same here, a ipaq h1930, h5450, and a Zaurus 5500 (should upgrade to an 800 series one of these days) |
06:07.32 | PTG1234 | i got a Zaurus, an ipaq and now this PPC 6600 |
06:07.58 | want561or772did | dear god |
06:08.01 | PTG1234 | to be honest with the 6600 my other pdas feel useless |
06:08.12 | PTG1234 | esp once sprint launches EVDO this month |
06:08.18 | newl | heh |
06:08.29 | PTG1234 | 500KB/s.. ... go baby |
06:11.29 | FuriousGeorge | now that ive mastered the connecting sip clients, im working on getting a dialtone out of my fxs |
06:11.51 | FuriousGeorge | i dont have a question yet, but i will, just warning |
06:11.58 | FuriousGeorge | PTG1234: im looking at you |
06:12.01 | FuriousGeorge | ;) |
06:12.42 | CoaxD | furiousgeorge: Not hard, sir. All about zapata.conf :) |
06:13.42 | FuriousGeorge | yeah, im looking at it now, but im having trouble grasping the layout. i dont get what does what, and a loot of settings i dont get. im gonna mess with it for a few minutes |
06:14.31 | *** join/#asterisk gdsm (~gdsm@mk-ns500-1.uk.tiscali.com) |
06:14.59 | PTG1234 | hah |
06:15.43 | CoaxD | FuriousGeorge: I've just DCC sent you my working zapata.conf file |
06:16.19 | CoaxD | FuriousGeorge: If you need the file via some other means (since you have yet to accept the DCC) please, by all means, lemme know |
06:16.37 | CoaxD | ah. there ya go! |
06:16.49 | FuriousGeorge | sorry was afk |
06:16.55 | FuriousGeorge | thanks a bunch |
06:17.25 | PTG1234 | 50 free downloads too if i buy that player |
06:17.27 | PTG1234 | woo hoo :) |
06:18.08 | CoaxD | FuriousGeorge: Sure! |
06:18.47 | CoaxD | FuriousGeorge: Stupidness, honestly. But in reality, zapata.conf aint that tough to figure out if you know the general concept by which it was written. (I had to do a bit of reading before i understood.) |
06:19.02 | CoaxD | FuriousGeorge: Everything works in drop-down fall-thru logic |
06:19.07 | *** join/#asterisk PTA (linux@202.5.131.37) |
06:19.15 | *** join/#asterisk j0 (~dan@S010600105a04ed8d.va.shawcable.net) |
06:19.16 | FuriousGeorge | well yours is what the unix people acll "human readable" |
06:19.25 | CoaxD | FuriousGeorge: Yepper |
06:19.29 | FuriousGeorge | but im sure i will get it eventually, just so much stuff in there |
06:19.39 | j0 | do i need to have a sound card installed to use music on hold? |
06:19.42 | CoaxD | FuriousGeorge: I always write my configs that way; that way, if i get hit by a bus, someone can figure it out |
06:19.44 | CoaxD | j0: No |
06:19.54 | CoaxD | j0: You're not actually accessing /dev/dsp |
06:19.58 | *** join/#asterisk DrPhiljr (~opera@81-170-161-230.bahnhofbredband.net) |
06:20.04 | CoaxD | j0: mpg123 output is piped; not played |
06:20.12 | j0 | hmm.. i'm not getting any music played.. i'm using an x100p |
06:20.26 | CoaxD | j0: Make *SURE* you're using the specified version of mpg123 |
06:20.30 | CoaxD | j0: No others will work |
06:20.35 | j0 | i get some extra static and occasionally a strange "bong" sound... |
06:20.36 | j0 | ok |
06:20.37 | CoaxD | i think its 0.59r but cannot be sure as i dont use it |
06:20.41 | DrPhiljr | hi I am new to asterisk, I was wondering if I can set it up on my pc to receive faxes with my SIP provider |
06:21.07 | JerJer | no |
06:21.12 | CoaxD | DrPhiljr: There's certainly fax capability within asterisk to accept faxes. That said, doing fax over sip certainly isnt the way to go |
06:21.14 | j0 | CoaxD: is 0.59r the asterisk or the mpg123 version? |
06:21.23 | CoaxD | j0: mpg123 |
06:21.40 | CoaxD | j0: You also have to watch out because a lot of distros really use 'mpg321', and symlink it to 'mpg123' |
06:21.43 | CoaxD | j0: This will NOT work |
06:22.03 | j0 | ok.. i'm 0.59q.. in musiconhold.conf should i use something like manual => custom:/var/lib/asterisk/mohmp3,/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
06:22.07 | DrPhiljr | I can receive faxes when I am connected to my ATA box, why not with a soft client |
06:22.10 | j0 | or use the default => quietmp3:.... |
06:22.14 | CoaxD | j0: You need to use 0.59r |
06:22.20 | CoaxD | j0: And *only* that version |
06:22.22 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
06:22.33 | iq{tablet} | hi, anyone using 'app_txfax.so' ? |
06:22.34 | j0 | CoaxD: ok thanks :) i'll try that |
06:22.40 | CoaxD | (apparently, something cool about that one) |
06:22.45 | CoaxD | sure! |
06:22.50 | CoaxD | sweet etc :) |
06:22.59 | FuriousGeorge | coaxd: whats the difference between loopstart and kewlstart |
06:23.06 | iq{tablet} | [app_txfax.so]Ouch ... error while writing audio data: : Broken pipe |
06:23.21 | CoaxD | FuriousGeorge: Read some :) |
06:23.28 | FuriousGeorge | fair enough ;) |
06:23.43 | DrPhiljr | CoaxD:I can receive faxes when I am connected to my ATA box, why not with a soft client |
06:23.47 | *** join/#asterisk sandu (~andu@S0106000476ee2cfe.cg.shawcable.net) |
06:23.48 | shmaltz | FuriousGeorge, how is your asterisk coming along? |
06:24.37 | iq{tablet} | hi, anyone using fax support? |
06:24.42 | FuriousGeorge | im able to set up my sip clients more or less by myself now (that which yuo helpped me with a few weeks back) and i ust got the tdm400s in the main |
06:24.44 | FuriousGeorge | mail |
06:24.49 | FuriousGeorge | so now im working on those |
06:25.06 | FuriousGeorge | once i get that, ill start with voicemail, bells, and whistles |
06:25.15 | sandu | Hi, I'm having a problem with *. DIALSTATUS never gets set. I'm using today's CVS; is there anything that needs to be set (maybe at compile time) to get this working ? |
06:25.18 | *** join/#asterisk aspworld (~aspworld@209.91.159.221) |
06:25.25 | *** part/#asterisk aspworld (~aspworld@209.91.159.221) |
06:26.06 | DrPhiljr | Do I need special HW to use asterisk or can I set it up to just receive faxes with my SIP provider? |
06:26.23 | FuriousGeorge | schmaltz: thanks again for that helping me out by the way, was a big moment actually getting a dialtone |
06:26.33 | JerJer | DrPhiljr: it doesn't work |
06:26.35 | JerJer | move on |
06:26.45 | j0 | CoaxD: wow.. dont know how i would of figured that out on my own :) thanks again |
06:27.17 | DrPhiljr | Jerjer, why does it not work? |
06:28.40 | FuriousGeorge | damn composite extension alwasy seems to make x misbehave after a few huors, but i love me some real x-parency so much i just cant help myself |
06:28.47 | DrPhiljr | I any newbie channel I can join? |
06:28.47 | FuriousGeorge | brb |
06:29.30 | j0 | DrPhiljr: i think newbies are just supposed to read faqs/voip-info.org... otherwise welcome :) |
06:30.25 | DrPhiljr | JerJerI have been told faxing should work, so you are confusing me |
06:30.36 | DrPhiljr | JerJer: I have been told faxing should work, so you are c onfusing me |
06:30.42 | j0 | its my understanding that whiel faxing works.. its not reliable enough to consider using |
06:31.20 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
06:31.22 | DrPhiljr | j0: Thanx. It works fine with my ATA box |
06:31.53 | JerJer | DrPhiljr: not over IP |
06:32.04 | *** join/#asterisk ChristianK (~Christian@p54A3B78E.dip.t-dialin.net) |
06:32.40 | shmaltz | FuriousGeorge, np |
06:32.45 | DrPhiljr | JerJer: I can confirm it works 100% with SIP |
06:33.04 | JerJer | maybe on your LAN with ulaw |
06:33.11 | JerJer | but not reliably over a WAN |
06:34.12 | DrPhiljr | JerJer: I have been receiving faxes over a year from regular landlines with my SIP provider |
06:34.22 | JerJer | not using asterisk |
06:34.27 | JerJer | or why in hell are you asking then? |
06:34.39 | j0 | DrPhiljr: does your sip provider turn them into emails? |
06:34.57 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
06:35.19 | DrPhiljr | JerJer: I am asking because I don't want to connect my fax to a ATA box, instead I want to use my laptop with a soft client to recieve faxes |
06:35.39 | DrPhiljr | j0: No I receive them directly |
06:35.56 | j0 | hm.. are there even any sip clients with fax capabilities? |
06:36.01 | JerJer | because your ATA speaks T.38 |
06:36.08 | JerJer | asterisk does not currently speak T.38 |
06:36.47 | DrPhiljr | j0: no there are no SIP clients with faxing cap, that's why i am want to use asterisk |
06:37.33 | j0 | gotcha :) |
06:37.35 | DrPhiljr | JerJer: mylaw is all it takes to receive faxes, but you need to have fast oing times |
06:38.37 | DrPhiljr | j0: so is it possible? |
06:38.42 | j0 | i wouldn't try it though.. like JerJer suggested |
06:39.03 | DrPhiljr | j0: why? |
06:39.14 | j0 | well i suppose if u got great ping times.. why not :) |
06:39.24 | j0 | give it a try. it may work for u.. but most likely, there will be issues |
06:39.39 | DrPhiljr | j0: I have 4ms ping times |
06:39.52 | sandu | Does it take any config options to have dial() set DIALSTATUS and ANSWEREDTIME on the channel? |
06:40.48 | DrPhiljr | j0: so will asterisk work a soft client or ...? |
06:41.01 | shmaltz | this is a nobrainer, dont know why it took them soooooooooooo long to figure this out: |
06:41.02 | shmaltz | http://story.news.yahoo.com/news?tmpl=story&ncid=1212&e=1&u=/washpost/20050417/tc_washpost/a58268_2005apr16&sid=96168964 |
06:41.24 | j0 | DrPhiljr: you mean a soft client to receive the faxes? |
06:41.58 | DrPhiljr | j0: yes |
06:42.44 | j0 | if u can find a soft client that has fax capabilities (havn't heard of one yet) sure |
06:42.52 | j0 | doesn't * have some sort of fax->email capabilities? |
06:42.56 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:43.17 | DrPhiljr | j0: no, that would be a greta solution, but no |
06:43.21 | j0 | DrPhiljr: http://freshmeat.net/projects/astfax/ |
06:43.31 | j0 | dont know anything about it.. but its probably worth a try |
06:43.57 | DrPhiljr | j0: well, i don't want to send faxes, just receive them |
06:44.49 | j0 | google around.. i'm sure theres an oss solution for u |
06:44.56 | DrPhiljr | j0: waht about SpanDSP? |
06:45.07 | j0 | never heard of it |
06:45.27 | FuriousGeorge | coaxd: when i first started today, even though i didnt have a dialtone, when i picked up the handset i could kinda hear that "handset echo" from the mouth to the ear part, but for the last few hours it has just sounded dead |
06:45.31 | DrPhiljr | j0: http://asteriskathome.sourceforge.net/features.html |
06:45.46 | FuriousGeorge | wondering if thats not part of my lack of dialtone, b/c i think now i have zapata.conf right |
06:45.53 | j0 | i'm not sure, but i think there is a way to get * to send incoming calls to a local 'com port' which could have fax software setup. |
06:46.21 | j0 | DrPhiljr: do you have a working * setup? |
06:46.27 | FuriousGeorge | why not have * send it over an analog line to a fax server or POTS fax machine |
06:46.27 | DrPhiljr | j0: no |
06:46.35 | j0 | DrPhiljr: well start there :) |
06:46.40 | FuriousGeorge | ;) |
06:48.03 | sandu | DrPhiljr: spanDSP would work; it receives the fax and saves it as a tiff so you email it |
06:48.19 | DrPhiljr | FuriuosGeorge: that's a solution i have been using up till now, but that mean my home computer has to be on all the times |
06:49.08 | DrPhiljr | sandu: is it possible to install asterisk with virtual pc on my xp machine? |
06:49.23 | j0 | DrPhiljr: works fine.. i have a test asterisk server in vmware |
06:49.35 | j0 | throw up *@home and ur golden |
06:49.40 | sandu | DrPhiljr: yes you could never tried it |
06:49.51 | sandu | DrPhiljr: also there's a windows port |
06:50.05 | DrPhiljr | there is? |
06:50.11 | sandu | DrPhiljr: but again I have no experience with it |
06:50.13 | j0 | hows that working out for ppl? |
06:50.19 | j0 | i would assume no zaptel drivers too |
06:50.28 | DrPhiljr | where can i get the port? |
06:50.28 | sandu | DrPhiljr: yes no zaptel last I heard |
06:50.42 | DrPhiljr | sandu: what's zaptel |
06:50.54 | iq | hi, anyone know cause of this messgae: Ouch ... error while writing audio data: : Broken pipe |
06:51.13 | sandu | DrPhiljr: http://voip-info.org/tiki-index.php?page=AstWind |
06:51.25 | DrPhiljr | sandu: I read that astrisk @home will format my drive |
06:52.59 | sandu | DrPhiljr: most likely, it will have to install linux but you might be able to do you own manual partitioning |
06:54.22 | DrPhiljr | maybe using virtualpc is safe then? |
06:54.35 | sandu | so I guess nobody has ever had problems with DIALSTATUS/answeredtime not being set :) |
06:54.45 | sandu | DrPhiljr: that or try the windows port |
06:55.22 | sandu | DrPhiljr: but I doubt you'll get spanDSP working on the windows port |
06:55.25 | j0 | DrPhiljr: u just need to get off ur ass and start trying stuff! |
06:55.31 | *** join/#asterisk pgpkeys (~pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
06:55.33 | DrPhiljr | well the windowport won't help me since I need spanDSP |
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07:01.03 | *** part/#asterisk DrPhiljr (~opera@81-170-161-230.bahnhofbredband.net) |
07:01.03 | FuriousGeorge | does anyne know what would cause an analog phone not to even have the "handset echo" where you can hear the mouth part in the ear part |
07:01.06 | jakepdev | iq: most likely a zaptel config issue... though it depends on the errors before theat |
07:01.08 | FuriousGeorge | is taht a hw problem |
07:01.17 | Moonwick | FuriousGeorge: a completely dead line? |
07:01.21 | FuriousGeorge | yeah |
07:01.42 | FuriousGeorge | i havent gotten it setup yet, but i feel like i was closer before |
07:01.54 | iq | jakepdev, I dont have any zaptel hardware. I think I wasn't suppose to compile zaptel support. |
07:02.11 | FuriousGeorge | iq: still useful if you wanna have conferncing |
07:02.13 | jakepdev | you need to check what errors came before that one |
07:02.22 | FuriousGeorge | for timer |
07:02.58 | iq | jakepdev, http://www.pastebin.com/272304 |
07:03.07 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
07:03.16 | iq | FuriousGeorge, will keep it then :) |
07:05.51 | jakepdev | doesn't look like the usual error. but it seems pretty coincidental that it failed right next the the zap line |
07:05.57 | FuriousGeorge | another thing thats been bothering me: i tell my bios to reserve irq 10 (b/c cat /proc/pci shows 10 is free) for pci slot 4, but when i load the driver its on irq 3 |
07:05.59 | FuriousGeorge | what gives? |
07:06.26 | iq | jakepdev, I'm recompiling now |
07:07.08 | FuriousGeorge | iq: you will have to load another driver called ztrtc i think |
07:07.14 | jakepdev | ztdummy |
07:07.25 | FuriousGeorge | right |
07:09.04 | iq | never used it before :) ...i'm setting up this new machine to work with AMP |
07:10.37 | jakepdev | good luck with it iq. gotta get some rest. |
07:10.41 | iq | its working zaptel.conf was missing coz I didn't make sample sample |
07:10.49 | iq | jakepdev, :) |
07:10.50 | jakepdev | great! |
07:11.04 | jakepdev | ok - well good night all |
07:12.42 | FuriousGeorge | later all |
07:18.17 | *** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net) |
07:21.51 | iq | anyone using asterisk@home? Default admin/password works for you - to log on to AMP? |
07:26.58 | |Vulture| | wow... *@home and AMP have brough a whole new crowd... I don't think I like it :P |
07:27.44 | unterx | I tried out *@home, but found I didn't like the mysql interfacing |
07:27.44 | unterx | so I'm back to square one |
07:28.40 | |Vulture| | the beauty of * is being able to configure it yourself, AMP just kills that |
07:29.04 | |Vulture| | because its not a dialplan that anyone that just picked up * is going to even understand |
07:29.29 | unterx | I don't even know what the hell a dialplan is |
07:29.37 | unterx | so I didn't learn anything using *@home |
07:29.46 | |Vulture| | ./etc/asterisk/extensions.conf |
07:30.02 | unterx | well, i know what a dialplan is, but I don't know how to write/customize one |
07:30.06 | |Vulture| | in AMP it is spread over 3 files I think |
07:30.25 | unterx | yeah |
07:30.51 | *** join/#asterisk joaovianna (joao@node-40247a6a.ewr.onnet.us.uu.net) |
07:31.01 | |Vulture| | I mean don't get me wrong.. its a good dialplan but its confusing because its all variables |
07:35.08 | unterx | when you call your asterisk pbx, if all you get is a busy signal, would that be the result of a configuration error or not registering with your provider properly? (over sip) |
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07:46.00 | barshad | can any one help me installing OpenH323 with asterisk ? |
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08:31.55 | unterx | this is strange, extensions behind my asterisk box can place and receive calls, but can not send speach |
08:32.22 | unterx | ie, it can call/receive calls from pstn phones, and can hear people on the pstn end talking/sending speach, but the person on the pstn end can not hear the sip phone |
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08:43.37 | Dovid | hello all |
08:43.57 | Dovid | whats the diffrence between x-lite and the reg. version ? |
08:45.25 | barshad | can any one help me installing OpenH323 with asterisk ? |
08:48.56 | JerJer | barshad: what is so difficult |
08:48.56 | JerJer | ? |
09:08.25 | barshad | JerJer: you there? |
09:08.33 | JerJer | nope |
09:08.37 | barshad | :) |
09:09.02 | cjk | hi, is this normal? voipgate1:~# netstat -lp | grep asterisk | wc |
09:09.02 | cjk | <PROTECTED> |
09:09.18 | barshad | is it possible to compile asterisk-oh323-0.7.1.tar.gz with current * CVS ? |
09:09.31 | JerJer | cjk: depends how many calls are going on |
09:09.43 | JerJer | barshad: cannot help you there |
09:10.16 | cjk | JerJer, 1 single call and my openfiles limit will be hit soon and * will crash. this happens every 3 hours |
09:10.33 | barshad | JerJer: can you recommend me which version of * and openh323 should i use ? |
09:10.55 | JerJer | cjk: then you have problems |
09:11.07 | JerJer | barshad: i can only assist you with my driver |
09:11.12 | JerJer | asterisk-oh323 is not my driver |
09:11.59 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
09:12.13 | barshad | Oh good, how can i compile your driver ? |
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09:12.37 | JerJer | read the README |
09:12.55 | cjk | JerJer, you are right. so now im trying to track down the problem. and maybe someone can tell me if 730 connectiosn is normal |
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09:13.31 | JerJer | netstat -lp | grep asterisk | wc |
09:13.32 | JerJer | <PROTECTED> |
09:13.40 | JerJer | that's with 10 active channels |
09:14.34 | JerJer | *CLI> show uptime |
09:14.34 | JerJer | System uptime: 2 weeks, 1 day, 7 hours, 24 minutes, 55 seconds |
09:14.34 | JerJer | Last reload: 35 minutes, 56 seconds |
09:15.00 | cjk | JerJer, and lsof tells you what? lsof | grep asterisk |
09:15.20 | JerJer | <PROTECTED> |
09:15.21 | JerJer | <PROTECTED> |
09:16.01 | cjk | hmm thats similar here |
09:16.49 | barshad | JerJer: i can not find PWlib 1.3.4, can i use 1.9.0 ? |
09:16.59 | JerJer | huh? |
09:17.07 | JerJer | don't be lame, use cvs -head |
09:23.57 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
09:25.37 | cjk | JerJer, i use quit alot of agi scripts. do you think the problem might come from there |
09:30.22 | cjk | hi guys what do you see on your system witht he following command? lsof | grep asterisk | grep "dev/null" | wc |
09:38.01 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l03v-14-146.d4.club-internet.fr) |
09:44.27 | cjk | netstat -lp | grep asterisk | wc reports me 48, when i do a call it reports me 48+4 and when i finisehd it reports me 48+2, so 50. so after a few hours it reports me more the 1000 and asterisk crashes. any idea? |
09:49.16 | foobos | maybe you have some firewall between and you don't see tcp rst packets |
09:49.25 | foobos | bad firewall that being |
09:51.46 | foobos | you could also try "show channels" if it lists those |
09:56.06 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-215-108.dsl.scarlet.be) |
09:58.26 | unterx | how would one go about making menus with asterisk? |
10:03.25 | pino | unterx: look at how the demo menu is made... |
10:03.33 | wildgoose | So how do people with an asterisk system work in an office with fax machines? Do you wire the fax system to the FXS cards on the asterisk box and accept faxes over FXO input only? If the fax arrives via the analog FXO input then do the asterisk addons for receiving faxes work reliably enough to be used for production? |
10:03.57 | pino | unterx: or read this wonderful example: http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture |
10:05.36 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
10:06.09 | pino | wildgoose: I think that if you want to use traditional fax machines you must either use FXS cards in the * box, or FXO cards and some kind of line simulator (like your old legacy PBX)... |
10:06.27 | pino | then again, who wants traditional fax machines? ;) |
10:17.10 | unterx | thanks pino |
10:17.24 | unterx | hmm, whenever i record via asterisk it sounds quite bad |
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10:39.53 | PoWeRKiLL | Morning :) |
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10:59.13 | cjk | anyone here who knows how far t.38 implementaiton is in asterisk? |
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11:13.42 | PatrickDK | cjk, 0% |
11:13.51 | PatrickDK | unless someone just started it recently |
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11:37.16 | emrah | Hi, I'm sorry to disturb. I'm blind, so I can't see how to change line in my polycom phone. do you have an idea? For example, when I'm online, how can I take an other line to call, where is placed the buton please? |
11:37.30 | emrah | in the left side, the long buton, is it one of them? |
11:37.54 | emrah | When I press the first one, it take the first line, but nothing happend when I press the second or the thirth one |
11:44.07 | tzafrir | any rtfm on checking that a sip extension is usable using sipsak or any other command-line tool? |
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12:23.12 | tzafrir | I'm using minisip as my client of choice for the moment |
12:49.53 | SuPrSluG | emrah:you have to make another extensoin for each in sip.conf and extensions.conf |
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13:14.30 | Blackvel | is it possible to stop an AGI command? |
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13:29.24 | cjk | hi, i sovled my problem. in fact if you register every second on asterisk using sip. your * will go down after 3-5 hours on a non tweaked linux |
13:29.41 | cjk | i had a user registering every second and that caused big problems |
13:29.56 | cjk | maybe one could code an anti-register-flood protection into asterisk |
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13:35.53 | synopia | hi @ all |
13:35.55 | pino | i remember that being in one of Jiri's (of iptel.org fame) presentations about SIP, as one of the problems they faced in developing SER |
13:37.00 | pino | cjk: did that happen because the re-registration happened before each registration was finished, or is there another reason? |
13:45.05 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
13:45.16 | TUplink | where are the soundfiles lacated? |
13:45.20 | TUplink | located* |
13:47.29 | TUplink | n/m /var/lib/asterisk/sounds |
13:47.35 | TUplink | ls |
13:48.18 | TUplink | where dose voicemail get its number sounds from? |
13:53.15 | jontow | hrm |
13:53.24 | jontow | in /var/lib/asterisk/sounds ;) |
13:53.38 | TUplink | the numbers arnt in there |
13:53.39 | mcnobody | TUplink /var/lib/asterisk/sounds/digits/ |
13:53.45 | TUplink | oh... |
13:53.46 | TUplink | ok |
13:54.50 | TUplink | so if i wanted to play it to the user i would playback(digits/num) |
13:54.51 | TUplink | ? |
13:55.18 | *** join/#asterisk ChristianK (~Christian@p54A3BB61.dip.t-dialin.net) |
13:55.22 | mcnobody | SayNumber(123) |
13:55.38 | mcnobody | or sayDigits(123) |
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13:58.23 | TUplink | when i put extention 20002 in to my IVR it recives it as 202 |
14:14.50 | TUplink | how do i set an ultmate timeout? |
14:15.02 | TUplink | and how do i cancle it |
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14:22.47 | ikey1 | hi can any one tell how to do loop back test with cross cable on E1's |
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14:36.28 | HjemmeRoyK | morning, morons |
14:45.36 | cypromis | morning northman |
14:46.12 | pino | are friendly greetings something reserved for Sundays? |
14:46.55 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
14:48.54 | cypromis | hehe |
14:49.01 | cypromis | nah rather reserverd for people you know |
14:49.01 | cypromis | ;) |
14:50.15 | pino | ;) |
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15:00.15 | *** part/#asterisk Om (~om@om.user) |
15:05.38 | TUplink | is there any was someone can let me get a copy of their dialplan? |
15:05.48 | *** join/#asterisk ethogeek (~ethogeek@CPE-24-209-154-94.wi.res.rr.com) |
15:07.45 | pino | you can find many examples on the wiki |
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15:12.36 | HjemmeRoyK | ~dialplan? |
15:12.37 | jbot | [dialplan] something that is planned to be rewritten in VB.net, or no! it's gonna be rewritten in brainf!(k |
15:12.37 | bugbot | dialplan? is assigned nothing and reported nothing. |
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15:32.47 | TUplink | how do i go about making a confrance room? |
15:33.44 | TUplink | will asterisk do video? |
15:34.10 | mcnobody | TUplink I have used video SIP calls. Works perfectly. |
15:34.10 | pino | ~meetme |
15:34.11 | bugbot | meetme is assigned nothing and reported nothing. |
15:34.33 | TUplink | kool |
15:34.48 | TUplink | meetme is my anwser to the conference |
15:34.59 | TUplink | ishte WIKI offline? |
15:35.07 | TUplink | i cant open up voip-info.org |
15:35.33 | mcnobody | TUplink I can access to wiki but it's terribly slow. |
15:35.42 | TUplink | oh... ok |
15:35.48 | TUplink | is it anywhere els? |
15:36.16 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
15:36.33 | mcnobody | TUplink I guess not... |
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15:40.14 | MikeJ[Laptop] | wow, this place is hopping |
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15:47.33 | TUplink | can the meetme room# and the extention to get to it be the same? |
15:48.54 | Weezey | TUplink: I don't see why not. |
15:49.05 | TUplink | ok...cause i cant get in my room |
15:49.24 | Weezey | did you create it in meetme.conf and reload? |
15:49.30 | cypromis | hop hop hop hop |
15:49.33 | TUplink | [conference] |
15:49.33 | TUplink | exten => 21001,1,Meetme(21001,m) |
15:49.33 | TUplink | [localext] |
15:49.33 | TUplink | include => conference |
15:49.48 | TUplink | and all incoming calls are put in localext |
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15:55.50 | dmccollum | Is there anyway to speedup how long it takes asterisk to route an incoming call to the ring group? I have a Vonage router with one PSTN port going to the x100p and the other PSTN port going directly to an analog phone and the analog phone will always ring two times before asterisk will ring my 7960's. |
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15:56.14 | E|nyPRI_ | hi |
16:00.20 | E|nyPRI_ | anyone running sql cdr? |
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16:08.14 | TUplink | <PROTECTED> |
16:08.14 | TUplink | Apr 17 12:08:11 WARNING[2795]: chan_zap.c:763 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
16:08.16 | TUplink | <PROTECTED> |
16:08.29 | jakepdev | dmccollum - you first need to find out where it is getting stuck at - you'll need to use some verbose logging |
16:09.08 | ariel_ | TUplink, you need to have a zap card or Ztdummy installed |
16:09.19 | dmccollum | Jakep: ok. I'll do that and see if I can see anything. |
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16:09.22 | TUplink | well... thats stupid |
16:09.32 | dmccollum | Anything particular I should be looking for? |
16:09.36 | jakepdev | could it be that it's waiting for caller id? |
16:09.37 | TUplink | i have to have a ZAP card to do a meetme |
16:09.52 | TUplink | ? |
16:09.59 | jakepdev | dm - is the call coming from analog? |
16:10.01 | ariel_ | TUplink, either a zap card or ztdummy |
16:10.09 | ariel_ | meetme needs a timing device |
16:10.10 | dmccollum | Jakep: yes |
16:10.12 | jakepdev | yep |
16:10.20 | jakepdev | it's prolly waiting for caller id |
16:10.36 | TUplink | jakedev its comming from COIP |
16:10.36 | jakepdev | that happens between the 1st and 2nd ring |
16:10.38 | TUplink | VOIP* |
16:10.43 | jakepdev | ok |
16:10.49 | jakepdev | then shouldn't be an issue |
16:10.52 | TUplink | so how do i get ztdummy |
16:11.00 | TUplink | but it is |
16:11.00 | dmccollum | Jakep: ok. That makes sense. |
16:11.04 | Qwell | TUplink: Its part of zaptel. Find the instructions on th ewiki |
16:11.17 | TUplink | conf => 21001 |
16:12.02 | jakepdev | dm - for analog origination, you can turn of caller id off in the zapata.conf file and turn immediate=on. that shuld speed it up |
16:12.10 | jakepdev | but no CLID |
16:12.32 | TUplink | Qwell i cant install zaptel |
16:12.35 | *** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com) |
16:12.42 | jakepdev | dm - er. zaptel.conf |
16:12.43 | Qwell | TUplink: Why not? |
16:13.39 | TUplink | um... dont compile on freebsd |
16:13.47 | Qwell | Then you can't use meetme |
16:14.03 | TUplink | im copying freebsd source to do it |
16:14.14 | TUplink | and going to try it from the ports |
16:14.32 | Qwell | ports won't have ztdummy, most likely |
16:14.47 | TUplink | im going to install zaptel |
16:14.55 | TUplink | i orderd 2 fxo's |
16:15.08 | TUplink | will need it anyway |
16:15.21 | jakepdev | i'm trying to find a way to set DNID. SetVar doesn't seem to work |
16:15.35 | Qwell | jakepdev: Whats DNID? |
16:15.45 | jakepdev | dialed number |
16:15.49 | Qwell | oh |
16:16.03 | jakepdev | ANI works fine with SetCallerID |
16:16.11 | jakepdev | but seems there is no couterpart |
16:16.19 | ManxPower | jakepdev: Unless a call is forwarded, DNID is the same as EXTEN |
16:16.31 | jakepdev | hmm |
16:16.40 | jakepdev | ok - i think i know how to do it then |
16:16.45 | dmccollum | Jakep: What would happen if I left usecallerid=yes and immediate=yes? Would it start ringing showing unkown until the callerid came through couple rings later? |
16:16.50 | ManxPower | And you can't set it, it's set FOR you. |
16:17.02 | |Vulture| | Can anyone here with a softclient try connecting to my * server? |
16:17.24 | jakepdev | i need it to go in the FastAGi message as part of the CDR |
16:17.44 | jakepdev | dm - if immediate is on - don't think you can get CLID |
16:17.59 | ManxPower | if immediate=yes MANY things stop working |
16:18.06 | TUplink | where do i get ztdummy |
16:18.14 | |Vulture| | hahahaha |
16:18.16 | dmccollum | jakep: Ok. I'll play around with it and see what happens. |
16:18.19 | jakepdev | ok |
16:18.19 | |Vulture| | ManxPower: I figured out the problem |
16:18.51 | ManxPower | What was the fix |
16:19.10 | |Vulture| | ManxPower: I had put the domain in my /etc/hosts as "127.0.0.1 domainname" and I am using externip=domainname |
16:19.19 | Qwell | I'm ignoring messages today |
16:19.30 | |Vulture| | so when it tried to establish a natted connection it was using 127.0.01 instead of the qualified externip |
16:19.35 | |Vulture| | stupid stupid me |
16:19.52 | TUplink | didnt i read that asteriks ignores /etc/hosts |
16:20.06 | |Vulture| | TUplink: it doesn't |
16:20.11 | |Vulture| | try it |
16:20.18 | TUplink | ok... |
16:20.26 | |Vulture| | just make sure you restart * |
16:20.27 | jakepdev | Manx: Re: " it's set FOR you." - not for me. it comes up blank... here is the incoming stream upon start of call ANI#DNIS# |
16:20.30 | ManxPower | |Vulture|: It's pretty well known that you need networking set up correctly to use Asterisk |
16:20.53 | Qwell | ManxPower: What, asterisk doesn't do it for you? :P |
16:20.59 | ManxPower | jakepdev: I believe DNID is only set for PRI. |
16:21.10 | |Vulture| | ManxPower: yea... I don't think I was totally with it last night |
16:21.20 | TUplink | qwell can i see a copy of your dialplan? |
16:21.30 | |Vulture| | I am gunna blame it on the booze... I was at the greenday concert last night... fun show |
16:21.35 | Qwell | nah, its ugly |
16:21.44 | ManxPower | jakepdev: you can try immediate=yes (this is one of the VERY few cases where it might be a good thing) then have an exten line to catch the ANI#DNIS# then break it out yourself. |
16:21.52 | TUplink | hum.... |
16:21.52 | jakepdev | right :) |
16:21.55 | TUplink | i need one |
16:21.57 | jakepdev | that's what i'm doing |
16:22.03 | TUplink | im doing somthing wrong |
16:22.07 | Qwell | ~wiki |
16:22.08 | bugbot | wiki is assigned nothing and reported M1412, M1666. |
16:22.12 | |Vulture| | haha |
16:22.14 | |Vulture| | I love that |
16:22.17 | jakepdev | manx - but don't know how to set DIND |
16:22.17 | Qwell | erm |
16:22.20 | |Vulture| | the wiki is assigned bugs |
16:22.44 | Qwell | tip: DOn't do that |
16:22.44 | Qwell | jbot yells at you |
16:22.45 | ManxPower | jakepdev: You are not getting DNID from the carrier. |
16:22.45 | *** join/#asterisk Inv_arp (junya@adsl-3-245-209.mia.bellsouth.net) |
16:22.45 | jakepdev | it's another switch |
16:23.00 | Inv_arp | omg xchat got in here autimatically |
16:23.01 | jakepdev | just sends ANI#DNIS# - i do two READ cmds |
16:23.10 | ManxPower | so whre does it send DNID? |
16:23.16 | jakepdev | MYDNIS :) |
16:23.59 | jakepdev | but ideally it would go into the proper variable i.e. DNID - which I think is supposed to be the number called |
16:24.02 | jakepdev | ~DNID |
16:24.04 | bugbot | DNID is assigned nothing and reported nothing. |
16:24.10 | jakepdev | ~DNIS |
16:24.11 | jbot | dnis is probably A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the destination device, which can ... |
16:24.11 | bugbot | DNIS is assigned nothing and reported nothing. |
16:25.19 | *** join/#asterisk reinhardt (~dreinhard@p549442FC.dip.t-dialin.net) |
16:25.34 | TUplink | is the WIKI somewhere els than voip-info.org? |
16:27.43 | Silik0n | no |
16:29.27 | ManxPower | ~docs |
16:29.28 | 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 |
16:29.28 | bugbot | docs is assigned nothing and reported nothing. |
16:29.28 | Silik0n | it would be nice if the wiki guy would let someonemirror that damned thing |
16:29.49 | ManxPower | Silik0n: I don't think anyone is stopping that |
16:30.12 | Silik0n | well you could wget mirror it, but I mean true mirror of the DB and everything |
16:30.24 | Silik0n | I've offered and he refused |
16:30.31 | ManxPower | Why? You would not be able to update the mirror anyway. |
16:30.59 | Silik0n | a read only mirror would be very nice |
16:31.19 | *** part/#asterisk ChristianK (~Christian@p54A3BB61.dip.t-dialin.net) |
16:31.30 | jakepdev | I scond that |
16:32.16 | ManxPower | I don't use the Wiki all that much. Much of hte information is downright wrong. |
16:32.18 | Silik0n | and it is quite often necessary |
16:32.28 | jakepdev | agreed |
16:32.39 | ManxPower | I use the mailing list archives. It has just as much wrong information, but it has more right informatiosn. |
16:32.40 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
16:32.49 | Silik0n | hah |
16:34.38 | file[laptop] | yay Farscape |
16:35.58 | Silik0n | farscape? |
16:36.06 | file[laptop] | yes, Farscape |
16:36.07 | Silik0n | whats thaton? |
16:36.19 | file[laptop] | it's on a Canadian channel called Space: The Imagination Station |
16:36.58 | Silik0n | heh |
16:37.21 | MikeJ[Jayden] | it's on scifi in the states as well |
16:37.25 | Silik0n | yeah |
16:37.27 | file[laptop] | I meant at this moment |
16:38.02 | TUplink | how do i do a modprobe zaptel on freebsd modprobe isntat command |
16:38.07 | Silik0n | file[laptop] did you see where ben browder (sp?) is going to be on SG1 in the next season |
16:38.14 | file[laptop] | Silik0n: yeah |
16:38.19 | Silik0n | TUplink: did you install it fromthe ports tree? |
16:38.40 | TUplink | im still downloading the freebsd src |
16:38.41 | TUplink | ;) |
16:38.50 | Silik0n | modprobe is a linux thing |
16:38.54 | TUplink | thought i would save on space and remove it |
16:39.03 | TUplink | yea... thats what i thiught |
16:39.06 | TUplink | thought |
16:39.22 | TUplink | if i do it in ports will it take care of all of that? |
16:39.51 | Silik0n | dunno never bother trying heh |
16:40.07 | Silik0n | so figure it out and report back heh |
16:40.29 | TUplink | hehe... ok |
16:41.07 | ManxPower | TUplink: You won't find much help here about running Asterisk on *BSD. |
16:41.16 | Qwell | ManxPower++ |
16:41.18 | Silik0n | thats fod damned sure |
16:41.32 | PatrickDK | hmm, I used to run asterisk on fbsd |
16:41.48 | PatrickDK | but too many problems with threading libs and udp packets |
16:42.02 | file[laptop] | oh wow, Tiger is using GCC 4.0 |
16:43.16 | HjemmeRoyK | file[laptop]: is that released yet? |
16:43.26 | *** join/#asterisk frogy (~edmund@cm218-255-137-80.hkcable.com.hk) |
16:43.39 | file[laptop] | not really, 3.4.3 is the latest release |
16:43.46 | file[laptop] | that's why I said, "oh wow" |
16:51.25 | *** join/#asterisk mbranca_home (~matteo@host-84-222-17-174.cust-adsl.tiscali.it) |
16:54.16 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
16:56.01 | shmaltz | anybody here knows why april first is aprils fool? |
16:58.55 | ManxPower | http://www.google.com/search?hl=en&q=+why+april+first+is+aprils+fool&btnG=Google+Search |
16:58.56 | shmaltz | ~aafj |
16:58.57 | jbot | rumour has it, aafj is http://lists.digium.com/pipermail/asterisk-users/2005-April/098601.html |
16:58.57 | bugbot | aafj is assigned nothing and reported nothing. |
16:59.38 | shmaltz | ManxPower, I heard that since Jesus was born on Dec 25th April 1st makes the day of conceiving |
16:59.48 | shmaltz | - 9 months |
16:59.59 | shmaltz | being Jewish it makes sense to me ;) |
17:01.01 | RaYmAn-Bx | except he wasn't conceived really :P...and as far as I remember it's not like he was actually born on 25th..it was just adjusted because it fitted better. |
17:01.47 | shmaltz | RaYmAn-Bx, he wasnt' conceived really? so how was he born? |
17:02.41 | RaYmAn-Bx | shmaltz: well..supposedly he just appeared out of thin air.. :P seriously though, isn't the story that she just became pregnant? |
17:03.14 | shmaltz | well then she conceived |
17:03.21 | *** join/#asterisk loick (~loick@APuteaux-151-1-28-79.w82-124.abo.wanadoo.fr) |
17:03.25 | RaYmAn-Bx | I guess |
17:03.29 | voipheck | r0d3nt |
17:03.32 | voipheck | RaYmAn-Bx |
17:03.54 | ManxPower | Myths usually don't make much sense if you analize them too much. |
17:04.09 | RaYmAn-Bx | indeed |
17:04.24 | shmaltz | every person I asked, if you would leave the coutnry for 6 months then came back to find your wife in her third month, and asked her how, and she answers from god. would you beleive her? they all said no |
17:04.30 | voipheck | anyone here has experience in configuring asterisk with quintum gateway? |
17:04.32 | dmccollum | I don't think Dec 25th is really the day he was born. I've heard it was actually spring time. December 25th was used because it use to be a Pagan holiday that was rolled in when The Romans declared the Catholic church as the official religion. |
17:04.36 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
17:04.50 | ManxPower | ~google site:lists.digium.com quintum |
17:04.50 | bugbot | google site:lists.digium.com quintum is assigned nothing and reported nothing. |
17:05.23 | ManxPower | dmccollum: You'll find most christian holidays are near a pagan holiday. |
17:05.29 | RaYmAn-Bx | I guess another somewhat amusing thing is the alternative meaning of conceive: to invent... |
17:05.49 | dmccollum | Just like the sabbath is really Saturday but the Catholic church moved it to Sunday to ease the transition for pagans. |
17:05.53 | shmaltz | oh my, your twisting my words ;/ |
17:06.11 | shmaltz | dmccollum, St. Peters did that |
17:06.19 | ManxPower | Heck, even chrismas has pagan things. christmas tree, for example is just another name for a Yule tree. |
17:06.28 | *** join/#asterisk iq (~IQ@70-59-160-225.omah.qwest.net) |
17:06.46 | dmccollum | ManxPower: easter, Christmas, Sunday were all pagan holidays. |
17:06.54 | RaYmAn-Bx | ManxPower: it's still called something very similar in nordic languages (juletræ in danish) |
17:06.55 | ManxPower | dmccollum: *nod* |
17:07.04 | shmaltz | St peters was an observant jew, who was sent by the Jewish sages, to make sure that thing don't stay the same, one of the things were Saturday to sunday |
17:07.48 | *** join/#asterisk GMsoft (~gmsoft@gmsoft.developer.gentoo) |
17:07.51 | GMsoft | hi guys |
17:08.53 | dmccollum | Actually it was Constantine that changed the sabbath from Saturday to Sunday. |
17:09.25 | *** join/#asterisk blop (~blop@2001:6f8:204:33:bbbb:bbbb:bbbb:bbbb) |
17:09.28 | *** join/#asterisk delphiuk (~richard@host81-155-71-194.range81-155.btcentralplus.com) |
17:09.31 | blop | :) |
17:09.34 | GMsoft | do you know there is a prog which use the IAX protocol to give free call to several european countries |
17:10.04 | GMsoft | you just create a account and use your asterisk to make phone calls and it works great :) |
17:10.31 | RaYmAn-Bx | GMsoft: ... |
17:10.34 | RaYmAn-Bx | go on :P |
17:10.38 | GMsoft | http://www.voipbuster.com/ |
17:10.45 | GMsoft | have fun:) |
17:11.03 | GMsoft | however it doens't allow you to call gsm unless you pay |
17:11.19 | GMsoft | and you don't have a phone number |
17:11.27 | hohum | why? GSM isn't bound to licensing restrictions like G.729 is |
17:11.28 | GMsoft | but that's still cool :) |
17:11.38 | blop | however, you need the windows software in order to create yourself an account :p |
17:11.39 | GMsoft | hohum: not the codec, the gsm phones :) |
17:12.09 | shmaltz | dmccollum, not according to what we jews beleive |
17:12.18 | hohum | I'm busy setting up a service similar to that |
17:12.19 | shmaltz | or know for that matter |
17:12.25 | hohum | but it will acctually be free |
17:12.32 | hohum | and won't require windows |
17:12.35 | GMsoft | hohum: really ? :) |
17:12.43 | GMsoft | to all countries ?:) |
17:12.53 | blop | lol , it cant be |
17:12.57 | *** join/#asterisk viLeR (~miv@ip-47-252.telesat.com.co) |
17:12.58 | hohum | no |
17:13.19 | hohum | it'll be free to on-net calls |
17:13.24 | GMsoft | ah ok |
17:13.30 | hohum | $19/year for unlimited incoming with a DID |
17:13.34 | *** join/#asterisk fievel (~fievel@210.117-136-217.adsl.skynet.be) |
17:13.35 | *** join/#asterisk Kal_Zakath (~Kal_Zakat@213.219.187.228) |
17:13.37 | GMsoft | but this service allow you to connect to the pstn network |
17:13.38 | Kal_Zakath | hi |
17:13.40 | blop | DID? |
17:13.48 | hohum | and you get charged for usage on outgoing calls |
17:13.55 | hohum | Direct Inward Dial |
17:14.00 | hohum | a phone number in otherwords |
17:14.44 | blop | k, but from which country ? |
17:14.55 | hohum | well |
17:15.27 | hohum | I'll only be able to offer US-based DIDs initally but I hope that will change soon |
17:15.45 | blop | :) |
17:15.49 | hohum | the more termination contracts I sign with DID providers the more coverage I'll have |
17:16.26 | *** join/#asterisk Rick_Hunter (~rhunter@03-178.008.popsite.net) |
17:16.40 | ManxPower | ~docs |
17:16.41 | 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 |
17:16.41 | bugbot | docs is assigned nothing and reported nothing. |
17:16.41 | ManxPower | ~mailinglist |
17:16.42 | jbot | well, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
17:16.42 | bugbot | mailinglist is assigned nothing and reported nothing. |
17:16.58 | hohum | I'm developing a billing system for SER and a voice mail platform based around Asterisk |
17:17.19 | Blackvel | is it possible to interrupt an AGI STREAM FILE? |
17:17.40 | Blackvel | maybe I can send the AGI EXEC DIAL afterwards? |
17:18.36 | *** part/#asterisk fievel (~fievel@210.117-136-217.adsl.skynet.be) |
17:19.43 | file[laptop] | the number of VoIP providers out there is making me sick |
17:20.14 | hohum | file: :) |
17:20.24 | ManxPower | Blackvel: According tot the DOCUMENTATION, you can interrupt STREAM FILE |
17:20.33 | hohum | I'm doing this primarily because I'm developing a billing system |
17:22.35 | ManxPower | http://home.cogeco.ca/~camstuff/agi.html#STREAMFILE |
17:23.22 | easimon | GMsoft: voipbuster works together with asterisk? |
17:23.38 | GMsoft | easimon: yes |
17:23.47 | GMsoft | it uses the IAX protocol |
17:24.47 | easimon | GMsoft: hm, never looked into iax so far... is there any docs on how to set it (voipbuster<->asterisk) up? |
17:25.21 | Blackvel | ManxPower: well, looks like it can be stopped by the client side, but how about the * side? |
17:25.35 | GMsoft | easimon: just call using the iax channel |
17:25.35 | Blackvel | hmm... maybe it is not the command I have to think about. |
17:25.37 | ManxPower | Blackvel: Huh? |
17:25.46 | Blackvel | I need something, where I can send a message and play a music |
17:26.11 | ManxPower | And? |
17:26.23 | Blackvel | and if something happens in the system, that I can control inside the AGI(FastAGI) that the command should be interrupted, just because e.g I want to connect the caller with the EXEC DIAL command |
17:26.23 | ManxPower | You mean make a call, not send a message, right? |
17:26.38 | ManxPower | Why not use MusicOnHold |
17:26.40 | easimon | GMsoft: i'll take a look into that, thx for the info ;) |
17:26.52 | GMsoft | np :) |
17:27.29 | *** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net) |
17:27.57 | Blackvel | ManxPower: wow, very good suggestion. I'll check doc how I can interrupt/stop this |
17:28.04 | Blackvel | probably it is what I am searching about |
17:28.08 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
17:28.09 | Blackvel | but there is even another .... |
17:28.15 | shmaltz | wiki is down? |
17:28.24 | Blackvel | as I have tested here, the dial command returns on busy or on timeout |
17:28.32 | Blackvel | if e.g I have setup a timeout of 30 seconds |
17:28.45 | Blackvel | how would I be able to stop the AGI EXEC DIAL? |
17:28.59 | Blackvel | I guess I can't send just the HANGUP? |
17:29.10 | Blackvel | hehe I know, this sounds very werid |
17:29.11 | Blackvel | weird |
17:29.39 | ManxPower | Blackvel: you can't. |
17:29.56 | ManxPower | Doing a Dial from AGI never does what you want. |
17:30.06 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
17:31.17 | Blackvel | how do you mean that |
17:31.27 | Blackvel | its not good to call exec dial over AGI? |
17:31.32 | Blackvel | or just that I can't interrupt it |
17:33.45 | easimon | GMsoft: but i guess, i have to start voipbuster once to create the account, right? |
17:33.58 | GMsoft | yes |
17:36.35 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
17:36.46 | file[laptop] | yay BoRiS |
17:37.11 | BoRiS | Mr. File on his laptop. :)))) |
17:37.26 | BoRiS | !!!! |
17:37.30 | BoRiS | How are you doing? |
17:37.36 | file[laptop] | how are you? |
17:37.41 | Silik0n | boris the whoris |
17:38.02 | BoRiS | Waking up and just heating up lunch in the microwave |
17:38.36 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-6-210.d4.club-internet.fr) |
17:47.27 | *** join/#asterisk wolfson (~hehe@cpe-68-187-189-084.man.nc.charter.com) |
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17:54.37 | Qwell | kram: afternoon |
17:54.44 | kram | afternoon |
18:00.04 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
18:00.27 | *** join/#asterisk delphiuk (~richard@host81-155-71-194.range81-155.btcentralplus.com) |
18:01.50 | delphiuk | does anyone know why I am getting lot's of "storage size of blah isn't known" when trying to compile zaptel with asterisk 1.07 from cvs stable? |
18:02.31 | MrBelvedr | do you have enough disk space? just a guess? |
18:02.52 | *** part/#asterisk Kal_Zakath (~Kal_Zakat@213.219.187.228) |
18:02.56 | delphiuk | plenty |
18:03.14 | delphiuk | 102 gig to be exact :) |
18:05.24 | Blackvel | manxpower: but show application musiconhold only says that it will play indefinitely, but not how I can stop it in my AGI (e.g I want to execute DIAL) |
18:10.42 | *** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) |
18:13.02 | *** join/#asterisk likwid-- (likwid@nc-65-41-191-111.dyn.sprint-hsd.net) |
18:14.57 | *** join/#asterisk In-Side (~Lowgitek@es-217-129-30-41.netvisao.pt) |
18:15.22 | In-Side | Hi again wazz up people ? |
18:15.43 | In-Side | does anyone has a cenario wich envolves ser+asterisk ? |
18:15.58 | *** join/#asterisk ethogeek (~ethogeek@CPE-24-209-154-94.wi.res.rr.com) |
18:18.06 | *** part/#asterisk orange_haw (~orangehaw@g9129.upc-g.chello.nl) |
18:18.11 | *** join/#asterisk asteriskn00b (asteriskn0@adsl-68-91-7-226.dsl.tulsok.swbell.net) |
18:18.38 | In-Side | I'm with some problems with * using a sip register to pass throgh the calls and I'm looking for some tips regarding to use ser as a nat helpper to asterisk box my configuration is something like this: [UAC -> ser] -> [asterisk] <-> [firewall] <-[all ports fwd]-> [router] -> cable mode - net <.> sip provider |
18:18.59 | In-Side | everything works fine if the provider is a IAX one |
18:19.11 | In-Side | but if i use sip nat refuses to work at all |
18:19.33 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
18:19.42 | In-Side | or better... it works but rtp session refuse to be estabelished |
18:20.23 | In-Side | does anyone has any tip that maybe usefull to read out or something |
18:20.30 | In-Side | I just lost in this mess :S |
18:20.32 | asteriskn00b | afternoon all, I am currently testing eyebeam on my network, calls work properly, however if I try to send a chat message to another user I get "SIP/2.0 405 Method Not Allowed" of course documentation on eyebeam is no existent, anyone have any ideas? |
18:20.39 | MrBelvedr | does ast@home come with gnome or kde installed? |
18:21.00 | In-Side | asteriskDOTbz: seems that the device doesnt suport it |
18:21.12 | In-Side | asteriskDOTbz: are using soft ip phone ? |
18:21.34 | file[laptop] | asteriskn00b: asterisk doesn't support messaging like that |
18:22.28 | asteriskn00b | I thought that the messaging was sip based =/ |
18:22.35 | file[laptop] | it is |
18:22.40 | In-Side | so nobody can help me out with my problem ? |
18:22.42 | file[laptop] | but chan_sip doesn't support it |
18:22.50 | In-Side | asteriskDOTbz: ser supports it fine |
18:22.58 | In-Side | but your devices has to support it too |
18:23.23 | file[laptop] | SER is just a sip proxy so it doesn't have to support it, it can just relay the MESSAGE packet |
18:23.30 | ariel_ | MrBelvedr, no |
18:23.31 | file[laptop] | but as In-Side keeps saying, your device has to support it too |
18:23.35 | In-Side | ok but it relays... |
18:23.42 | In-Side | asterisk don't |
18:23.55 | file[laptop] | correct, asterisk doesn't support it |
18:24.02 | In-Side | anyway I just can't use ser alone |
18:24.12 | In-Side | as I can't register only with ser at my providers.. |
18:24.21 | In-Side | I have to sticky out with * too |
18:24.32 | In-Side | and now I just sticked to IAX providers |
18:24.46 | In-Side | as the damn sip is just hard to play with nat at hell |
18:24.49 | In-Side | :S |
18:25.38 | emrah | Anyone know how to enable the whole three lines in a polycom? Now I can only use the first one. |
18:26.13 | Qwell | So, a friend of mine might be hooking me up with a 79x0(he isn't sure what it is) |
18:26.24 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
18:26.28 | Qwell | Its an "extra" that isn't logged in the company inventory... |
18:26.37 | Qwell | So, he's gonna send it to me. heh |
18:26.40 | file[laptop] | ooh sounds sneaky |
18:26.45 | ariel_ | emrah, I have 7 polycom IP-500 coming in this week. Hope there easy to setup. |
18:26.49 | emrah | Please, there is no one here using a Polycom IP500? |
18:26.49 | Qwell | file[laptop]: very much so, yeah |
18:27.10 | emrah | ariel_: They are strange... |
18:27.10 | PTG1234 | i use polycoms, although i would prefer not to :) |
18:27.12 | In-Side | I just hope to get the damn nat working... that is just what I'm wondering about |
18:27.12 | In-Side | :S |
18:27.13 | PTG1234 | hey file and qwell |
18:27.17 | file[laptop] | hey hey |
18:27.34 | In-Side | If i use my phone with stun it works well if i not use any UA on my entwork |
18:27.36 | PTG1234 | file: wanna here the mp3 player update? |
18:27.57 | ariel_ | PTG1234, why don't you like them? |
18:27.59 | file[laptop] | PTG1234: sure |
18:28.12 | PTG1234 | ariel: they are uncomfortable, and have mutiple issues w/ differen tthings |
18:28.21 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
18:28.25 | PTG1234 | so i went to compusa, to get another player.. and they had this sony HD1.. which was slick |
18:28.29 | TUplink | Apr 17 14:27:15 WARNING[475]: chan_zap.c:763 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
18:28.29 | PTG1234 | but they only had the floor model |
18:28.32 | PTG1234 | so i end up buying that one |
18:28.35 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
18:28.36 | emrah | ariel_: They are easy to setup if you understand the first time. You can setup sevral phones with just one boot server, that's great. |
18:28.39 | PTG1234 | and when i get hom there is no cradle in it :) |
18:28.41 | PTG1234 | so nothing i can do with it |
18:28.48 | PTG1234 | so i ended up ordering the hd3 online |
18:28.52 | |Vulture| | I use IP500s |
18:28.58 | |Vulture| | I have about 60 of them |
18:28.59 | PTG1234 | man that thing is smaller then an ipod mini, 20gb of space, and 30 hours of battery life |
18:29.02 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
18:29.06 | file[laptop] | well that's evil |
18:29.10 | TUplink | how do i get /dev/zap/pseudo |
18:29.13 | In-Side | asteriskDOTbz: regarding to your problem if you want to get WMI in asterisk, please take a look at this: http://www.voip-info.org/wiki-Asterisk+at+large |
18:29.31 | PTG1234 | audio quality is suppose to be awesome too |
18:29.40 | PTG1234 | but now i gotta wait til tuesday to have it |
18:29.40 | PTG1234 | blah |
18:30.02 | easimon | GMsoft: got it working, quality is - as expected - not the best... |
18:30.13 | TUplink | i need meetme to work. but keep getting WARNING[475]: chan_zap.c:763 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
18:30.29 | |Vulture| | TUplink: you don't have a digium card right? |
18:30.32 | easimon | TUplink: install ztdummy |
18:30.35 | In-Side | seems to be impossible but I just can't find a soul that uses asterisk and ser together :S |
18:30.36 | TUplink | rite |
18:30.48 | TUplink | i belive it did install ztdummy |
18:30.52 | |Vulture| | did you make the zaptel drivers? yea what easimon said |
18:30.59 | TUplink | i installed /usr/ports/misc/zaptel |
18:31.01 | file[laptop] | he's using FreeBSD |
18:31.02 | |Vulture| | TUplink: you running 2.6 kernel? |
18:31.05 | |Vulture| | ah |
18:31.08 | TUplink | um... FREEBSD |
18:31.15 | ariel_ | TUplink, look at the Makefile for the ztdummy. comment it out on the zaptel drivers and make clean the make install then make clean and same with asterisk. |
18:31.19 | PTG1234 | FreeBSD rocks! :) |
18:31.21 | In-Side | there is no problem with zaptel and bd |
18:31.24 | In-Side | bsd |
18:31.28 | TUplink | i want both of them |
18:31.34 | emrah | I see some people there is using polycom phones. Please, how can I do to have every lines ok to place 3 calls together, and switching in them? |
18:31.41 | In-Side | you just have to load up the things right |
18:31.48 | TUplink | it wont make from the download have to install it from ports |
18:31.55 | TUplink | rite |
18:32.03 | In-Side | I have a damn clone that just freaks out my bsd box |
18:32.34 | In-Side | damn cheap cards :S |
18:32.39 | TUplink | i did make ; make isntall on zaptel |
18:32.53 | emrah | In-Side: are you speaking with me? |
18:33.00 | In-Side | to all |
18:33.03 | ariel_ | TUplink, did you remove the # from the ztdummy in the makefile |
18:33.09 | In-Side | ya |
18:33.12 | TUplink | um... let me go loook |
18:33.15 | In-Side | you have to uncomment the lines |
18:33.17 | In-Side | at configuration |
18:33.22 | In-Side | I can't say its work |
18:33.34 | In-Side | because the than fxo/fxs card is a peace of shit |
18:33.46 | TUplink | i saw samtingdutting the make bout ztdummy |
18:33.49 | emrah | No one knows hoe to enable all lines with the IP500? |
18:33.55 | In-Side | and after few seconds workign crash the damn box :S |
18:34.05 | emrah | When I press the buton for line 2 or 3, nothing happened |
18:34.09 | PTG1234 | emrah: you put them in the conf file? where is the rock science? |
18:34.15 | In-Side | I never had a hardware that could do this in my bsd box :S |
18:34.21 | In-Side | anyway |
18:34.26 | easimon | TUplink: ztdummy is a kernel module... i guess you have to tell freebsd to load it at boot time... |
18:34.27 | In-Side | can anyone help me with nat ? |
18:34.41 | In-Side | I now it is not a real easy thing ... but.. ok.. |
18:34.44 | In-Side | just some lights |
18:34.45 | In-Side | :S |
18:34.49 | TUplink | anyone know how to load a kernel module in freebsd? |
18:34.56 | In-Side | kldload name_module |
18:35.02 | emrah | PTG1234: But there is no actually a created configuration where I can see an example of that...... |
18:35.04 | ariel_ | In-Side, nat is well a pain sometimes but what are you trying to do? |
18:35.06 | ManxPower | emrah: Each line needs to register as a different user |
18:35.08 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
18:35.14 | emrah | It pure xml files that you have to write yourself |
18:35.17 | PTG1234 | emrah: you know the line1_ entries? |
18:35.17 | In-Side | TUplink: you may load it up at startup |
18:35.23 | PTG1234 | duplicate them with line2_ |
18:35.25 | PTG1234 | and line3_ |
18:35.25 | GMsoft | easimon: yeah quality sux and the time to establish a call is really long |
18:35.36 | In-Side | asteriskDOTbz: regarding to your problem if you want to get WMI in asterisk, please take a look at this: http://www.voip-info.org/wiki-Asterisk+at+large |
18:35.38 | In-Side | ops |
18:35.39 | TUplink | how do i do that kldload zaptel dont work |
18:35.48 | In-Side | I'm with some problems with * using a sip register to pass throgh the calls and I'm looking for some tips regarding to use ser as a nat helpper to asterisk box my configuration is something like this: [UAC -> ser] -> [asterisk] <-> [firewall] <-[all ports fwd]-> [router] -> cable mode - net <.> sip provider |
18:35.53 | emrah | ManxPower: But there is 6 possible registrations! |
18:36.04 | delphiuk | can someone tell me what linking i have to do with 2.4.27 source in order for asterisk to compile? |
18:36.08 | In-Side | TUplink: I can take a look |
18:36.10 | In-Side | wate |
18:36.21 | easimon | GMsoft: in addition to that, it does not hang up correctly ;) |
18:36.47 | emrah | PTG1234: I don't thnig that there is per line configuration |
18:36.48 | In-Side | TUplink: did you install the port ? |
18:36.59 | TUplink | yea |
18:37.01 | PTG1234 | emrah: yes its exactly like i just said |
18:37.08 | TUplink | http://www.voip-info.org/wiki-FreeBSD+zaptel looks like it would be rite but it wont load |
18:37.18 | ManxPower | emrah: Yes. |
18:37.19 | TUplink | what are the odds it is down |
18:37.22 | emrah | PTG1234: What's the configurtion for line one then, where sould I find it? |
18:37.32 | In-Side | did you install this /usr/ports/misc/zaptel/ ? |
18:37.32 | emrah | How is named the first link? phone one? |
18:37.37 | emrah | phone1.xml?` |
18:37.47 | In-Side | TUplink: do you likje me to do a package for you? |
18:37.51 | *** part/#asterisk loick (~loick@APuteaux-151-1-28-79.w82-124.abo.wanadoo.fr) |
18:37.53 | ManxPower | emrah: Configure the phone over the web page. |
18:38.12 | ManxPower | Once you get it working, read the admin guide and configure an ftp server to let the phone upload it's config |
18:38.18 | TUplink | in-side huh? |
18:38.19 | emrah | ManxPower: But there is 6 possiible registration! |
18:38.47 | emrah | Sorry for the ! mark |
18:39.12 | *** join/#asterisk egon_l (~egon@pc-10-19-104-200.cm.vtr.net) |
18:39.30 | emrah | But what I mean is there is 6 registrations possible, with 6 diffrent proxy. So, how is i possible to work with 2 lines? And... I didn't understand how to peak up a call incomming in an other line when I'm speaking with someone else. |
18:40.06 | emrah | tree lines, sorry |
18:40.27 | ManxPower | emrah: The polycom uses the same config files for all models. The phone will ignore configs for lines it doesn't actually have. |
18:40.45 | ManxPower | emrah: dude, you have to read the mailinglist acrhives. |
18:40.57 | ManxPower | polycom phones are not easy to configure, but they are great phones. |
18:41.28 | ManxPower | One of the few issues is that you can't disable call waiting on the phones, you have to keep a second call from coming in via GetGroup, SetGroup in the dialplan |
18:42.13 | emrah | Thanks a lot :) you helped me very much in this Asterisk room. I will do that, but just.. You don't have an indication? Should I register towo times the same proxy, How do I peak up the second incomming call? (I can't see the screen, as you know I think. |
18:42.59 | ManxPower | emrah: correctly set up, the call will come in on a diffferent line on the phone |
18:43.23 | emrah | Do you know where I can find an example of a guide to set is up correctly? |
18:44.08 | *** join/#asterisk fugitivo (~ajf@201.255.101.186) |
18:45.25 | emrah | ManxPower: Thanks a lot, but I have to go somewhere now. If you can just help me a bit more, leave me your essages in a private window. |
18:46.03 | |Vulture| | is the wiki down? |
18:46.04 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
18:46.32 | Sedorox | wow.. 16 hrs to eat |
18:46.38 | Sedorox | musta been a big meal |
18:46.39 | Sedorox | :-p |
18:46.42 | TUplink | |vulture| i think it is |
18:46.46 | docelmo | ya ya ya |
18:46.58 | TUplink | i cant get to it for that last 6 hours |
18:47.16 | niZon | the wiki is slowwwwwwwwwwww |
18:48.14 | ariel_ | I think it's down. I have not been able to get to the wiki since this morning. |
18:48.37 | niZon | yeah seems so.. |
18:48.44 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
18:48.46 | niZon | someone needs to mirror it |
18:48.57 | |Vulture| | wonder how much traffic it gets |
18:49.24 | niZon | a fair bit probablly.. |
18:49.26 | ariel_ | I think people have said that there willing to mirror it. But the person that runs it says no. |
18:49.52 | |Vulture| | well if its under 1 TB a month I could prolly do it it |
18:49.58 | |Vulture| | ah |
18:50.09 | |Vulture| | ariel_: yea then they should find someone else to do it |
18:50.27 | |Vulture| | that information is vital for people learning and those already using * |
18:51.50 | Sedorox | I know I would mirror it... even tho I don't have a big connection |
18:52.18 | joaovianna | . |
18:52.25 | Dovid | an anyone recomend a good wifi voip phone ? |
18:53.19 | niZon | hmm... 2 error pages from google so far today |
18:56.01 | ariel_ | I have some of the pages printer out for my own use. I have it as a manuel. |
18:57.30 | ariel_ | |Vulture|, I will be around your area next week. |
18:57.50 | PTG1234 | how much bandwidth can a bunch of text pages take |
18:58.04 | ariel_ | |Vulture|, going to be staying in the Orlando area for 1 week. I have a time share I need to use. |
19:01.37 | PTG1234 | anyone have an ipod shuffle? |
19:04.29 | file[laptop] | meep I'm back |
19:05.20 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
19:07.18 | *** join/#asterisk want561or772did (~aoiahsdf@68.71.213-36.atlsfl.adelphia.net) |
19:11.56 | Dutts | anyon ehere from the uk? |
19:13.53 | shmaltz | Dutts, I'll be in the UK on Thursday |
19:15.13 | delphiuk | Dutts, yep, i am |
19:16.49 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
19:18.04 | Dutts | delphiuk: I have a proplem sending proper callerid to my analogue phones hanging off my tdm400 card, I see the callerid come in now but I think it's getting sent in the wrong format so my uk phones don't like it..... what settings should I have in my zapata.conf to get this working? |
19:19.39 | delphiuk | Dutts, hang on. I can't even get the thing compiled at the moment. do you want to chat in asterisk-uk? |
19:19.53 | Dutts | yup no probs |
19:19.54 | pino | dutts: you might want to check http://www.voip-info.org/wiki-Asterisk+and+UK+Caller+ID -- if it ever downloads |
19:20.12 | Dutts | piino: ah will try that, yeah I'm having trouble on voip-info atm |
19:21.12 | *** join/#asterisk Moc (~Moc@modemcable165.109-70-69.mc.videotron.ca) |
19:21.59 | |Vulture| | everyone is |
19:22.06 | |Vulture| | the wiki is toasted |
19:22.13 | Sedorox | again |
19:24.54 | Dovid | ah |
19:24.59 | Dovid | thought it was my isp |
19:25.04 | Dovid | yes the wiki seems to be down |
19:25.30 | Dovid | anyone know of a good wifi woip phone ? |
19:26.48 | Dovid | anyone ? |
19:27.34 | Dovid | oh well |
19:28.08 | *** join/#asterisk asteriskn00b (asteriskn0@adsl-68-91-7-226.dsl.tulsok.swbell.net) |
19:28.18 | *** join/#asterisk roamer323 (~sing@HSE-Toronto-ppp131661.sympatico.ca) |
19:30.29 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
19:37.26 | BoRiS | Dovid: Not even worth it yet |
19:40.13 | delphiuk | Dovid, I would recommend a normal DECT with a ATA |
19:40.43 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-6-210.d4.club-internet.fr) |
19:41.48 | GMsoft | voip-info.org down again :-/ |
19:42.04 | *** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net) |
19:43.43 | |Vulture| | is the Cisco one any good? although it has no SIP image |
19:44.05 | *** join/#asterisk _Raptor_ (~RaptorBlu@p5492E052.dip.t-dialin.net) |
19:44.07 | _Raptor_ | hi |
19:44.11 | ManxPower | Cisco what? |
19:44.21 | GMsoft | |Vulture|: which phone ? |
19:44.37 | GMsoft | every cisco phone have a sip image |
19:45.28 | file[laptop] | almost. |
19:45.51 | ManxPower | No. Not all Cisco phones have SIP images. |
19:46.33 | _Raptor_ | can anyone tell me how i can hide my caller id in chan_capi (www.voip-info.org seems to be down) |
19:46.35 | |Vulture| | Ill get the model #s I don't their their Star phone nor their wireless has a SIP |
19:47.31 | |Vulture| | 7920 |
19:47.36 | |Vulture| | http://www.cisco.com/en/US/products/hw/phones/ps379/ps5056/index.html |
19:47.50 | |Vulture| | and the 7936 doesn't have sip either |
19:48.02 | file[laptop] | haha the 7920... |
19:48.10 | ManxPower | The 7095, 7940, and 7960 are the only phones that I know of that have SIP |
19:48.18 | |Vulture| | file[laptop]: whats funny? |
19:48.30 | file[laptop] | |Vulture|: it's a funny phone |
19:48.41 | file[laptop] | it has to see a CCM server on the network before it'll give you a list of the available access points |
19:48.42 | |Vulture| | okay... |
19:48.49 | |Vulture| | oh wow... |
19:48.54 | |Vulture| | tricky tricky |
19:49.07 | *** part/#asterisk RevK ([U2FsdGVkX@excalibur.bracknell.aaisp.net.uk) |
19:49.08 | file[laptop] | malcolmd tried to get it working at VON... haha |
19:49.08 | |Vulture| | so you couldn't even install a SIP image if you didn't have CCM |
19:49.12 | *** join/#asterisk pino (~z@host241-115.pool80116.interbusiness.it) |
19:49.15 | file[laptop] | it has no SIP image to begin with |
19:50.10 | ManxPower | The 7095, 7940, and 7960 are the only phones that I know of that have SIP |
19:50.13 | ManxPower | Let me check |
19:50.47 | |Vulture| | file[laptop]: right, but I am just saying if one was released |
19:51.01 | file[laptop] | needless to say it's a funny phone |
19:51.05 | |Vulture| | you couldn't install it because it need CCM to detect the wireless |
19:51.48 | |Vulture| | polycom should make a wireless solution |
19:51.49 | ManxPower | 7905, 7912, 7940, 7960, are the only phones that can run SIP. |
19:52.32 | Nivex | Does the Sipura SPA-2000 come fairly well recommended around here? |
19:53.10 | file[laptop] | yes. |
19:53.33 | file[laptop] | <monotone> I wholeheartedly recommend the Sipura SPA-2000 and am in no way being influenced by Sipura. |
19:53.43 | Nivex | o.O |
19:54.00 | Nivex | The first one I had was a dud. Wanted to know if it was worth giving them another chance. |
19:54.38 | Sedorox | yes |
19:55.44 | *** join/#asterisk Milligan (~Milligan@wkstn4.gnwd-noc.valuelinx.net) |
19:55.48 | *** join/#asterisk iq (~iq@70-59-160-225.omah.qwest.net) |
19:57.02 | Dutts | can someone take a look at my zapata.conf and tell me why I am not sending caller id properly to my uk pots phones please? |
19:58.08 | Moc | you can't send caller id throught a pots phone line |
19:58.24 | *** join/#asterisk likwid-- (likwid@nc-67-76-41-210.dyn.sprint-hsd.net) |
19:59.36 | Dutts | can you not? I'm talking a standard BT line? I get it into my FXO port from the PSTN but cannot seem to send it out on my FXS port, you saying this is impossible? |
19:59.47 | Dutts | on a TDM400P card |
20:00.32 | Moc | you can't send callerid throught a FXO line. Only receive them |
20:01.15 | Dutts | yup I agree.... but my question is about sending it out on my fxs line |
20:01.26 | delphiuk | Dutts, when it goes to FXO, it is from Asterisk in it's callerId format i think? |
20:01.43 | Dutts | to my regular phone (which when I connect directly to my bt service, withouth * int he way) works fine and displays caller id fine |
20:01.48 | Moc | Dutts, that another story. You said send it to the uk pots |
20:02.13 | Sedorox | .... not sending caller id properly to my uk pots phones please?.... |
20:02.16 | Sedorox | he said his normal phones... |
20:02.27 | pino | M1719 |
20:02.27 | bugbot | M1719 is a feature bug that is closed (markster): [patch] Caller ID support for UK (BT POTS lines). It was filed by TonyHoyle and was last updated on 10-03-04. http://bugs.digium.com/bug_view_page.php?bug_id=1719 |
20:03.01 | Dutts | Moc: sorry mate, yes I see where I have confused the issue there a little! =) no I only said pots to distinguish this phone from an ip phone or anything, analog phone would have been a better way of saying it |
20:03.30 | pino | Dutts: try applying that patch :) (assuming you're using an X100P) |
20:03.51 | Moc | pino, he said he use a tdm400 |
20:04.01 | Moc | I dont know much about how those card work in the UK |
20:04.23 | Dutts | using a TDM400P, I thouight that patch was only to be able to receive caller id from the fxo on an x100p, from what U've heard this is supported in the tdm400p.... |
20:04.24 | pino | Dutts: is your problem getting the CID from the PSTN to * or from * to the phone? |
20:04.57 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-213-149.dsl.scarlet.be) |
20:05.30 | Sedorox | the way I'm unterstanding it.. from * to phones |
20:06.00 | Dutts | pino: * to the phone, I can see in the CLI that my caller id is getting into *, then I do a Dial(Zap/1/${CALLERID}) and it shows correct int he CLI (in that the macro gets expanded corretcly) but my phone just shows CALL so I'm guessin git's int he wrong format or something |
20:06.10 | pino | yes, sorry - i didn't read the lines above! |
20:06.15 | Dutts | I also get Apr 17 19:56:37 WARNING[7632]: chan_zap.c:3613 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. |
20:06.27 | Dutts | <PROTECTED> |
20:10.15 | Dovid | wiki ral slooooooooooooooowwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwwww |
20:10.20 | Dutts | oh it's up now? |
20:14.02 | Dutts | that is the right syntax to send caller id isn't it? Dial(Zap/1/${CALLERID}) |
20:14.06 | Dovid | very sloooooooooooooooooooooooooooooooooooooooooooow |
20:14.17 | file[laptop] | it should go automagically |
20:14.38 | Dutts | so I can try just Zap/1 ? |
20:14.42 | file[laptop] | yes, try it |
20:14.58 | Dutts | actually that's a bloody good point, don't think I've tried that since I got inbound cid working =) one sec |
20:15.03 | |Vulture| | Dutts: if you have channel 1 defined |
20:15.32 | Dutts | Vulture: yeah the call goes out correctly and everything, the phone rings, but caller id isn;'t diaplyed (it normally is able to display caller id when connected to the pstn) |
20:15.45 | file[laptop] | just try and see... |
20:16.20 | *** join/#asterisk asteriskn00b (asteriskn0@adsl-68-91-7-226.dsl.tulsok.swbell.net) |
20:16.29 | Dovid | grr |
20:16.31 | Dovid | its down again |
20:16.48 | file[laptop] | asteriskn00b: what now? |
20:17.09 | Dutts | bah no that didn't work! =( |
20:17.37 | Dutts | is it this though every time it tries to make a call out it says this Apr 17 20:19:29 WARNING[7667]: chan_zap.c:3613 zt_handle_event: Didn't finish Caller-ID spill. Cancelling. |
20:18.44 | asteriskn00b | nothing |
20:18.52 | asteriskn00b | just soking up the knowledge |
20:18.56 | file[laptop] | k |
20:19.00 | asteriskn00b | err soaking |
20:19.31 | Dutts | when you change zapata.conf, is it sufficient to just do a reload in * or do I need to do more? |
20:24.11 | *** join/#asterisk zoa (zoa@82.103.76.147) |
20:25.55 | *** join/#asterisk PBXtech[mobile] (~upirc@wirelessdata-167-245.mycingular.net) |
20:26.33 | Dovid | Dutts: when changing the zapta files you must stop asterisk and restart it |
20:26.41 | Dovid | stop now |
20:26.50 | Dovid | and then asterisk |
20:27.05 | Dovid | damn |
20:27.16 | Dovid | querys on wiki are taking for ever........................................................... |
20:27.28 | asteriskn00b | i cant even get it to load |
20:28.03 | Dovid | u have to be patient |
20:28.16 | Dovid | like 10 minutes for a simple query |
20:28.40 | want561or772did | wiki is taking siesta |
20:30.31 | Dovid | grr |
20:30.38 | Dovid | gona grab some zzzz and try later |
20:30.42 | Dovid | ugh i hate it |
20:31.15 | delphiuk | tutt, anbody would have thought that Asterisk was popular ;) |
20:31.52 | want561or772did | i think the wiki going to sleep is some other problem |
20:32.17 | Dovid | hehe |
20:32.32 | Dovid | RIP WIKI |
20:34.11 | want561or772did | maybe there's an hourly backup going on or distance between the DB and cgi |
20:34.17 | Dovid | hmm |
20:34.26 | Dovid | then it would be like this every week |
20:34.58 | want561or772did | nearly every day i manage to hit wiki during one of its slowdowns |
20:35.18 | asteriskn00b | ok all who has the best prices on ip phones... other than ebay |
20:36.46 | want561or772did | try google cache Dovid |
20:36.50 | *** join/#asterisk Marcel-AS16215 (Marcel-AS1@gic-msg-exc-01.genotec.ch) |
20:36.59 | Dovid | too much of a pain |
20:37.10 | Dovid | anyone know of a good wifi voip phone ? |
20:37.36 | Marcel-AS16215 | hi to all, quick question how i can download from CVS 1.0.7 i download always 1.0.1 with this command -> cvs checkout zaptel libpri asterisk |
20:38.29 | MikeJ[Jayden] | Marcel-AS16215, you are not downloading 1.0.1 with that command, you are getting head |
20:38.54 | Dovid | anyone know a good voip wifi phone ? |
20:38.55 | MikeJ[Jayden] | Marcel-AS16215, look on the wiki under asterisk download |
20:38.58 | *** join/#asterisk mog_home (~mog_home@146.229.178.196) |
20:38.59 | want561or772did | hmmmmm |
20:39.43 | MikeJ[Jayden] | Dovid, none of them are really good |
20:40.02 | Dovid | hehe |
20:40.08 | Dovid | well i need one for a client |
20:40.20 | Dovid | the cisco one looks tempting but the price isnt |
20:40.25 | want561or772did | get a portable phone & fxs |
20:40.29 | Dovid | cant |
20:40.40 | Dovid | he wants to put em on hold, call tranfer etc. |
20:40.49 | want561or772did | ah |
20:41.02 | Dovid | i can prob get him a pda with wifi |
20:41.11 | Dovid | hmm |
20:41.14 | Dovid | not a bad idea |
20:41.33 | Dovid | get cheap pda's wit wifi and put a soft phone on it |
20:43.55 | *** join/#asterisk rowter (~Drake@201.133.210.80) |
20:44.03 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-198-62.dsl.scarlet.be) |
20:44.41 | niZon | PDA's with wifi aren't cheap :\ |
20:45.35 | want561or772did | fxs ports should allow you to flash over to a pbx menu, choose options like forward, then flash back to the line you put on hold |
20:51.13 | Dovid | niZon: they are on ebay |
20:51.27 | Dovid | cheper than a cisco at least |
20:51.28 | Dovid | http://search-completed.ebay.com/pda-wifi_W0QQcatrefZC6QQcoactionZcompareQQcoentrypageZsearchQQcopagenumZ1QQfisZ2QQfromZR10QQfrtsZ50QQfsooZ1QQfsopZ3QQga10244Z10425QQsacatZQ2d1QQsbrsrtZl |
20:53.32 | Dutts | anyone know what'd going on with the wiki? |
20:53.53 | Sedorox | it went p00f |
20:54.00 | niZon | m00f |
20:54.02 | smiley- | http://www.asterisk-support.de/mirror/odbc/app_dbodbc.c is that the version to go with if I want to use the odbc-functions with asterisk? |
20:54.07 | poli | Dutts, I am trying to get in for 3 hours now, offline. :( |
20:54.15 | smiley- | who needs the wiki then there is google cache? ;) |
20:54.19 | Dutts | poli: heheheh ok I think I'll give up for tonight |
20:54.21 | smiley- | s/then/when/ |
20:54.35 | poli | Dutts, ;) |
20:57.35 | *** join/#asterisk drbraun (~reb@d128118.adsl.hansenet.de) |
20:57.41 | drbraun | Evening all |
20:58.13 | Marcel-AS16215 | Do Asterisk Conferncecall works on FreeBSD with the ZapTel Dummy Driver |
20:58.20 | Sedorox | yes |
20:58.27 | Sedorox | MeetMe |
20:58.34 | Blackvel | bye for today :) |
21:00.46 | *** join/#asterisk starme (~starme@pcp07223314pcs.dalect01.va.comcast.net) |
21:00.57 | starme | hello all |
21:01.38 | starme | anyone awake? |
21:02.27 | Weezey | to call with SIP, what's the correct Dial string? Dial(SIP/user:pass@host/number?) |
21:02.40 | *** join/#asterisk Rob- (~robbie@haylott.plus.com) |
21:04.30 | drbraun | hi starme |
21:04.36 | starme | weezey: dial(SIP/SIP_CHANNEL/number, length,r) |
21:04.52 | starme | drbraun: hows it going? |
21:04.59 | Weezey | starme: it has to be in sip.conf to work then? |
21:05.16 | drbraun | starme, shite, to be honest. fricking chan_capi ..... |
21:05.17 | starme | weezey: what are you trying to do? |
21:05.29 | Weezey | starme: just dialing out via sip |
21:05.45 | starme | as a part of an application? |
21:06.10 | Weezey | no, just as part of extensions.cofn |
21:06.12 | Weezey | conf |
21:06.12 | starme | starme: honesty is always good |
21:06.24 | Weezey | talking to yourself? |
21:06.25 | drbraun | starme: ;) |
21:06.31 | starme | so from a phone or softphone |
21:06.35 | drbraun | starme: what about you? |
21:06.39 | starme | drbraun: long day ;) |
21:06.49 | starme | dr: working on my asterisk config |
21:06.50 | Weezey | starme: to a phone relay |
21:07.15 | starme | trying to understand the whole context thing, I have a bunch of lines, but I want to restrict the directory and outbound lines to specific things |
21:07.35 | drbraun | starme: spent 11h so far today on this issue. got no incoming audio from chan_capi when making an outgoing call. other side can hear me fine. Otherway round works with no probs ... |
21:07.49 | starme | weezey: thats that I have in my extensions.conf to handle an outbound call for a find me line |
21:08.06 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
21:08.06 | *** mode/#asterisk [+o anthm] by ChanServ |
21:08.21 | starme | drbraun: you set nat? |
21:08.33 | *** part/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
21:09.09 | drbraun | starme: making a sip call to * on the same network, i.e. no NAT. Same issue when connecting via IAX2 |
21:09.23 | starme | ah |
21:09.25 | starme | weird |
21:09.44 | starme | what is chan_capi? |
21:10.00 | drbraun | starme: in deed :( And the voice quality of the call is great. If it would only be both ways :) |
21:10.23 | drbraun | starme: capi channel for .... CAPI devs :) |
21:10.36 | starme | ah |
21:10.42 | starme | forgive me, I am still learning |
21:10.43 | starme | :) |
21:10.48 | drbraun | starme: me too |
21:11.00 | drbraun | nothing to forgive ;) |
21:11.21 | starme | I have a bunch of polycoms, and a couple SIPURA 1001 atas |
21:12.20 | drbraun | starme: cannot complain about toys. If the fricking capi chan would work, I'd be a happy soldier |
21:12.37 | starme | what exactly is a capi device |
21:14.15 | drbraun | starme: an ISDN card which can be handled by The Common ISDN API (CAPI). AVM and the likes fall under this. |
21:15.18 | drbraun | starme: you must be in North America or some other non-European part of the world, right? |
21:15.24 | starme | US |
21:16.08 | drbraun | starme: 'You don't do' ISDN very much over there, do you? |
21:16.25 | Dovid | wiki still dead ? |
21:16.32 | drbraun | wiki is up again |
21:16.33 | starme | rarely, sometimes in a data center |
21:16.36 | starme | but not often |
21:16.56 | drbraun | upppssss, may be it is down again :) |
21:17.07 | Dovid | all images are down |
21:17.23 | starme | can anyone help with this: I have an * install that has multiple SIP and a single ZAP line coming into it |
21:17.23 | Dovid | site is down |
21:17.26 | Dovid | !wiki |
21:17.31 | Dovid | !~wiki |
21:17.34 | Dovid | !wiki |
21:17.38 | Dovid | ~wiki |
21:17.39 | bugbot | wiki is assigned nothing and reported M1412, M1666. |
21:17.39 | starme | I want to split up my extensions for director and allow outbound calling through a specific trunk |
21:17.52 | Dovid | ~list |
21:17.54 | jbot | one warez list being sent |
21:17.54 | bugbot | list is assigned nothing and reported M2430. |
21:17.55 | Dovid | ~ |
21:18.16 | starme | so all my "200" extensions have one directory and go outbound on the ZAP channels |
21:18.36 | starme | my "400" extensions have access to their directory and go outbound on a specific SIP channel |
21:18.44 | drbraun | anybody out there that is running chan_capi with an AVM controller? Prefrebly in Germany? |
21:19.17 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
21:20.12 | easimon | drbraun: yup, here... |
21:20.21 | *** join/#asterisk tessier (~treed@210.245.38.37) |
21:20.45 | drbraun | easimon: Have you got a min? Mind if I msg you directly? |
21:21.48 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
21:22.38 | rabelais | what is the name of the online directory where I can register my name and iax/sip address?? it was a .org, and it had a w and a 3 in it I think, maybe an h as well...but I can't find it anymore |
21:23.39 | Qwell | e138? something like that |
21:23.41 | Qwell | ~enum |
21:23.42 | bugbot | enum is assigned nothing and reported nothing. |
21:23.50 | Qwell | ~e138 |
21:23.51 | bugbot | e138 is assigned nothing and reported nothing. |
21:23.59 | NewSole | e164.org |
21:24.03 | Qwell | Thats the one :P |
21:24.17 | NewSole | and yes you can register |
21:24.27 | rabelais | ah, ya |
21:24.27 | rabelais | that's it |
21:24.30 | rabelais | thank you |
21:25.26 | NewSole | keep getting "No Auth Found" |
21:30.18 | rabelais | I'm at the signup page for e164.org...if a user is trying to contact someone with an e164.org account, will they ever need to know/type in the account username? or only the number that they assign me later |
21:31.36 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
21:34.00 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
21:34.06 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
21:34.48 | want561or772did | i thought e164 was for registering ip presence for lines you also own a DID for |
21:35.27 | want561or772did | iaxtel, though, would assign you a 700XXXXXXX number which they forward to |
21:37.02 | Dovid | WIKI DOWN |
21:37.05 | Dovid | GRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRRR |
21:38.43 | zoa | haha |
21:42.23 | want561or772did | still :( |
21:42.41 | smiley- | use google cache :) |
21:43.12 | want561or772did | anyone know the correct format for callerid that cellphones want? NXXNXXXXXX alone makes them want to put a + in front of the number |
21:43.16 | want561or772did | like it's international |
21:43.39 | PatrickDK | heh, iaxtel doesn't assign did's |
21:43.55 | want561or772did | i know, i didn't mean they gave you a did |
21:44.24 | want561or772did | just a new id which didn't require an existing did |
21:46.39 | *** join/#asterisk Strom_TM (~Strom_TM@office4.tmcs.net) |
21:52.09 | smiley- | <PROTECTED> |
21:52.10 | smiley- | <PROTECTED> |
21:52.10 | smiley- | Segmentation fault |
21:52.13 | smiley- | yes \o/ |
21:52.56 | Luhiwu | anyone knows who can sell bulk cheap DIDs? |
21:53.09 | Luhiwu | any country, any areacode, just cheap :) |
21:54.06 | Dovid | ok |
21:54.09 | Dovid | <PROTECTED> |
21:54.23 | Dovid | whats the url that u use for google chache ? |
21:54.59 | niZon | cache:url? |
21:56.05 | Dovid | yes |
21:56.18 | niZon | try that :P |
21:56.47 | Dovid | huh ? |
21:58.06 | smiley- | search for the thing with google.. and then press the Cached link |
21:59.00 | Dovid | eh |
21:59.05 | niZon | in the search bar enter cache:<url> |
21:59.06 | Dovid | wiki isnt in google cache |
21:59.11 | Dovid | goto head to work |
21:59.14 | Dovid | be back later |
21:59.19 | niZon | ~google cache:www.voip-info.org/tiki-index.php |
21:59.19 | bugbot | google cache:www.voip-info.org/tiki-index.php is assigned nothing and reported nothing. |
21:59.32 | niZon | interesting |
22:00.49 | smiley- | is there any "real version" of ODBCput and that stuff in asterisk now? the app_dbodbc seems to segfault all the time |
22:02.48 | *** join/#asterisk jeffik (jefik@69.158.30.24) |
22:03.28 | *** join/#asterisk notat (~ariel@telconinjas.net) |
22:15.03 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
22:22.11 | *** join/#asterisk pbxjunkie (~Stormtroo@ppp14-adsl-159.ath.forthnet.gr) |
22:23.16 | gambolputty | Is there a limit to the number of simultaneous calls * can handle in one box? |
22:23.36 | file[laptop] | I'll pretend I didn't see that |
22:23.45 | Strom_TM | only limited by the amount of hardware you want to throw at it |
22:23.48 | |Vulture| | lol |
22:23.50 | tzanger | gambolputty: no, asterisk utilizes superconductors; once you get up to running speed the resistance loss in the conductors vanishes |
22:23.59 | file[laptop] | many factors attribute to the number |
22:24.09 | tzanger | it's kind of like zero-point energy but without the quacks |
22:24.10 | file[laptop] | including the rotation of the earth |
22:24.14 | |Vulture| | codec for one |
22:24.16 | Strom_TM | and by "hardware" I mean hammers, jigsaws. screwdrivers, etc |
22:24.25 | file[laptop] | indeed, hammers |
22:24.30 | |Vulture| | jigsaws! |
22:24.32 | tzanger | I find that when the earth spins counterclockwise I can't handle nearly the same number of calls |
22:24.42 | |Vulture| | tzanger: ROFL |
22:24.46 | file[laptop] | tzanger: it's true! |
22:24.46 | Strom_TM | hahaha |
22:25.01 | tzanger | now whenever that happens I just sacrifice a virgin and my call volumes come back to normal |
22:25.02 | Strom_TM | also, your call capacity increases by about 15% during a solar eclipse |
22:25.06 | |Vulture| | tzanger: but when the earth stops its a bitch on server load |
22:25.10 | tzanger | it's getting awfully hard to convince people to come over these days |
22:25.11 | file[laptop] | those virgins, can always count on them |
22:25.43 | |Vulture| | tzanger: if you need any my roomates are available... but I think your looking for female virgins |
22:25.57 | tzanger | |Vulture|: yeah male virgins aren't rare enough |
22:26.27 | |Vulture| | tzanger: true... is there an age limit on the female virgins? like 18+? |
22:26.42 | |Vulture| | cause I think it might be pretty hard to find a 21 year old virgin |
22:26.46 | file[laptop] | like a virgin, OOOOOOOOH |
22:27.02 | |Vulture| | :P |
22:28.20 | Strom_TM | oh god, now my head is full of the sounds of bad digital mastering |
22:28.38 | file[laptop] | it's nice to have you back in the channels! |
22:28.50 | Strom_TM | hehe yes, i need to not go away for so long :) |
22:29.15 | file[laptop] | indeed |
22:29.52 | |Vulture| | whats the best over the counter drug for neck pain? |
22:30.00 | file[laptop] | a two by four |
22:30.04 | file[laptop] | over the head |
22:30.06 | |Vulture| | lol |
22:30.10 | tzanger | file[laptop]: no no no that's for generating neck pain |
22:30.12 | |Vulture| | Im serious |
22:30.12 | Strom_TM | what kind of neck pain? |
22:30.28 | tzanger | Strom_TM: he swallowed his viagra too quickly and he's got a stiff neck now |
22:30.33 | Strom_TM | hah |
22:30.43 | |Vulture| | well I fell climbing a few weeks ago and was like... SHIT then I stood up and I was fine... |
22:30.59 | |Vulture| | but then this morning I wake up and arch my back and my neck goes snap |
22:31.04 | |Vulture| | and now its tense as hell |
22:31.12 | |Vulture| | don't have much motion in it |
22:31.23 | file[laptop] | go for a massage! |
22:31.32 | |Vulture| | yea I think I might |
22:31.42 | Strom_TM | try ibuprofen - it's an anti-inflammatory analgesic |
22:31.45 | |Vulture| | they have them cheap here.. for the students learning |
22:32.05 | |Vulture| | Strom_TM: took like 3 at noon and 2 more a min ago |
22:32.13 | Strom_TM | actually, go for a dose of ibuprofen and a dose of acetaminophen and that should do the trick |
22:32.27 | |Vulture| | acetaminophen? |
22:32.30 | Strom_TM | yes |
22:32.54 | |Vulture| | tylenol... thanx google :) |
22:32.59 | |Vulture| | brb getting some |
22:37.12 | Moc | |Vulture|, yea, IP600 are nice phone |
22:38.27 | |Vulture| | I have a buncha IP500/7960/7940s and were slowly phasing out our 60/40s |
22:38.39 | |Vulture| | but I still want to try a 600 for personal use |
22:38.58 | Strom_TM | my only terribly minor gripe about the 7960 is that when on speakerphone, the SIP firmware plays the touchtones. whereas the SCCP firmware plays beeps. I want the SIP firmware to beep, dammit :) |
22:39.11 | *** join/#asterisk chamba (~chamba@64.119.36.42) |
22:39.17 | Strom_TM | |Vulture|, how much do you want for the 60s? :) |
22:39.34 | MrBelvedr | how can i install a pacage from the CENTOS cd-rom |
22:39.35 | Moonwick | Strom_TM: I can empathize :) |
22:39.51 | |Vulture| | Strom_TM: duno we haven't gotten to that phase yet, I will be selling 60s and FXO modules torwards mid summer |
22:47.12 | NewSole | keep getting "No Auth Found" |
22:50.00 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02m-6-210.d4.club-internet.fr) |
22:50.43 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
22:53.32 | NewSole | anyone alive |
22:54.10 | Strom_TM | no |
22:54.19 | Strom_TM | we're all dying in fearful agony |
22:55.07 | NewSole | so I am not alone... lol |
22:56.07 | TomL | ~seen manxpower |
22:56.12 | jbot | manxpower is currently on #asterisk (1d 19h 41m 20s). Has said a total of 71 messages. Is idling for 3h 4m 23s |
22:56.12 | bugbot | seen manxpower is assigned nothing and reported nothing. |
22:56.19 | TomL | damn you manx :P |
22:56.25 | jakepdev | <MrBelvedr>: Centos is an RedHat variant - so i'd assume to install a package - you's use RPM |
22:56.28 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
22:56.54 | file[laptop] | gah packet loss |
22:56.56 | file[laptop] | silly ISP |
22:57.00 | dmccollum | Yep, CentOS is built from the Redhat source, so it should be almost identical. |
22:58.14 | *** join/#asterisk voiper (~none@pcp0010158348pcs.eatntn01.nj.comcast.net) |
22:58.42 | *** join/#asterisk sudoer (~toy@denali.ccs.neu.edu) |
22:58.47 | sudoer | voip-info is down |
22:58.55 | jakepdev | sure is |
22:59.22 | dmccollum | I hate when great sites like that go down. ipcops.net was down for a couple months. |
23:01.32 | sudoer | i take it cluecon is fake? |
23:02.07 | file[laptop] | ...sure it is |
23:02.13 | sudoer | nm |
23:02.28 | sudoer | is there a website for it? |
23:02.35 | file[laptop] | not yet |
23:06.12 | sudoer | id like to go |
23:06.34 | *** join/#asterisk Slainte (~Slainte@66.55.113.102.ppp.northrock.bm) |
23:08.07 | Romik | hello - somebody can advice what this a problem? Apr 18 00:17:45 WARNING[24533]: chan_iax2.c:5553 socket_read: Call rejected by 199.203.98.117: Unable to negotiate codec |
23:08.22 | file[laptop] | it was unable to negotiate the codec |
23:08.28 | Romik | what it mean? |
23:08.29 | Slainte | I think its a codec problem :) |
23:08.40 | cypromis | people who can read have an advantage I;d sau |
23:08.42 | cypromis | I'd say |
23:08.44 | cypromis | args |
23:08.57 | cypromis | it means your allow/disallow line son both sides do not match |
23:09.14 | Romik | it should make convertion in this case? |
23:09.33 | cypromis | where ? |
23:09.41 | cypromis | if you disallow it to connect with a compatible codec |
23:09.52 | cypromis | is the stream between the two endpoints supposed to do the translation ? |
23:10.01 | Romik | no! |
23:10.06 | Romik | but this possible |
23:10.20 | file[laptop] | Romik: you can't talk to another box using a codec if they don't support it |
23:10.55 | cypromis | asterisk will translate between two different connections |
23:11.03 | cypromis | but not on a conncetion |
23:11.04 | file[laptop] | exactly. |
23:11.07 | cypromis | that is not possible |
23:13.01 | Romik | this is related to [general] section of iax conf? |
23:13.16 | Romik | yes! this is fixed it |
23:13.21 | Romik | thank to everybody |
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23:15.57 | Strom_TM | wheeeee |
23:15.57 | want561or772did | i think my change fell out |
23:15.57 | Romik | this can be ignored: Apr 18 00:23:34 NOTICE[24533]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/livevoip/5 of format speex since our native format has changed to ulaw |
23:15.57 | Romik | <PROTECTED> |
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23:57.09 | NewSole | anyone alive |
23:57.11 | Juxt | lol me |
23:57.29 | shmaltz | wiki is down? |
23:57.30 | NewSole | :P |
23:57.32 | Juxt | yeah |
23:57.33 | Romik | somebody can advice about this message "Apr 18 00:23:34 NOTICE[24533]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/livevoip/5 of format speex since our native format has changed to ulaw"? |
23:57.34 | Juxt | that too |
23:57.36 | shmaltz | NewSole, just born |
23:58.27 | NewSole | no... newsole because we made a decision and we are going to suply DID's and Voip to Asterisk users |
23:58.58 | Juxt | NewSole, whcih company are you with? |
23:59.53 | NewSole | lets just say... clearity and and dropping is not a problem.... and we support g729 and g723 |