00:00.34 | Un1x | anyone here can please help me with a simple dailplan for my extensions.conf |
00:04.03 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
00:05.02 | crochat | Hello ! |
00:05.17 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
00:05.27 | SpaceBass | Un1x, have you checked the handbook? |
00:05.41 | Qwell[] | Un1x: There are at LEAST 3 different ways to call Digium |
00:05.51 | SpaceBass | Un1x, not being dismissive at all...just that I read it once and went cross-eyed then read it again and it started to make sens |
00:06.08 | crochat | I have a little problem with Asterisk... actually, I think the problem isn't Asterisk, it's the network... but I don't understand :-( I need help ! |
00:06.23 | SpaceBass | crochat, tcpdump is your friend :) |
00:06.24 | Qwell[] | Un1x: http://www.digium.com/en/company/contact.php |
00:06.34 | CrashHD | crochat: you will get further by asking your question than you will by talking about it |
00:06.40 | crochat | SpaceBass: Sure, it's already done |
00:07.08 | hads|home | Un1x: The free after sales support from Digium doesn't include setting up a dialplan AFAIK |
00:08.02 | crochat | So now, I have two computers with exactly the same Asterisk configuration, and those computers are connected directly on the Net |
00:08.14 | *** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com) |
00:08.30 | watchy2 | should i beable to call through my tivo using a ATA to asterisk? |
00:09.02 | crochat | I tried the tcpdump command : "tcpdump -i eth0 dst host 66.234.138.73" (66.234.138.73 is my SIP provider) on both computers |
00:09.04 | CrashHD | watchy2: data is same as fax...you will have unreliable results |
00:09.15 | watchy2 | yea thats what i thought |
00:09.23 | watchy2 | i wonder how the shit im suppose to update this tivo then |
00:09.37 | CrashHD | network connection |
00:09.38 | crochat | This tcpdump command shows nothing on my server, but shows that on the second computer : |
00:09.45 | crochat | 01:42:13.070013 IP 84-74-153-239.dclient.hispeed.ch.sip > 66.234.138.73.sip: SIP, length: 399 |
00:09.45 | crochat | 01:42:13.241658 IP 84-74-153-239.dclient.hispeed.ch.sip > 66.234.138.73.sip: SIP, length: 614 |
00:09.48 | CrashHD | unless it's a direct tv tivo then you need to have a phone line |
00:09.57 | watchy2 | it is directv tivo homie |
00:10.02 | CrashHD | your screwed |
00:10.06 | watchy2 | but the bitch is complainin about a phone call |
00:10.17 | CrashHD | mine use to do that every 6 months |
00:10.21 | CrashHD | I would just take it to a friends |
00:10.25 | Un1x | anyone wanna help me build a simple dail plan? |
00:10.29 | watchy2 | yea ill probably just take it to work |
00:10.32 | CrashHD | start the connection, and leave it over night |
00:10.41 | watchy2 | so data is that unreliable over freakin voip huh? |
00:10.46 | watchy2 | i kinda imagined it was |
00:10.52 | watchy2 | but never inquired up on it |
00:10.57 | CrashHD | there are some standards to allow data |
00:11.06 | CrashHD | but not fully supported here, from what I'm told |
00:12.21 | Un1x | anyone wanna help me build a simple dail plan? please? |
00:13.12 | *** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com) |
00:13.21 | watchy2 | i guess i need a new freakin cable provider to |
00:13.44 | watchy2 | so i guess im just screwed. ill take my tivo to work and do it there i guess |
00:15.08 | *** join/#asterisk Dr-Linux (n=ubuntu@202.59.73.131) |
00:15.15 | *** join/#asterisk prepaid (n=nikhil@ip68-229-76-11.ri.ri.cox.net) |
00:15.34 | prepaid | I'm trying to user the Record cmd, but was wondering if anyone knows a way to make it beep 10 seconds before the max recording time? |
00:15.37 | Dr-Linux | hi guys |
00:16.49 | Dr-Linux | prepaid: put some wait after the beep sound? |
00:17.41 | prepaid | Dr-Linux I guess, but I'm a little confused... I execute Record with max time = 50 seconds, and want it to beep at 10 seconds before 50, what exactly are you proposing?, playback a 40 second blank file with a beep at the end? |
00:19.01 | Un1x | shuit man |
00:19.36 | Dr-Linux | no, i mean after the beep put Wait(10) |
00:20.10 | Dr-Linux | aww |
00:20.23 | Dr-Linux | beep is a part of record application? |
00:20.23 | prepaid | huh |
00:21.01 | prepaid | it can be for the beep on record startup |
00:21.07 | prepaid | or you can just play teh 'beep.gsm' file |
00:21.34 | Dr-Linux | yeah |
00:21.44 | prepaid | but i don't get how to play a beep 40 seconds into a record |
00:22.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
00:25.19 | dlynes_laptop | No, there's an indication in indications.conf for the beep |
00:25.27 | dlynes_laptop | Not quite sure where it's used in asterisk, though |
00:25.50 | prepaid | is there any idea on how to accomplish what i'm trying to do? something that seems like a rather simple request? |
00:28.39 | dlynes_laptop | prepaid, i don't see anywhere in the code where it's used for anything you'd need it for, either |
00:28.41 | dlynes_laptop | prepaid, gimme a sec |
00:29.05 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
00:30.39 | caio1982 | tzafrir_laptop: ping |
00:33.18 | dlynes_laptop | prepaid, have a look in apps/app_dial.c |
00:33.36 | dlynes_laptop | prepaid, then do a search for ast_get_indication_tone |
00:33.50 | dlynes_laptop | prepaid, you'll find some code around there on how to utilitize an indication tone |
00:34.31 | Un1x | dlynes man i been waiting for you |
00:34.34 | dlynes_laptop | prepaid, you'd probably need to modify some code in the apps/app_monitor.c code to make use of it |
00:34.45 | Un1x | i need help bro, i got asterisk libpri and zaptel installed on slackware |
00:34.54 | dlynes_laptop | prepaid, i'm assuming you're using Monitor(), and not Record(), anyways |
00:34.54 | Un1x | but problem comes here that i need some help with configuration |
00:34.59 | dlynes_laptop | Un1x, ? |
00:35.09 | Un1x | extensions.conf need a dailplan i think |
00:35.18 | tzafrir_laptop | caio1982, here |
00:36.01 | dlynes_laptop | Un1x, [incoming] exten => s,1,Dial(SIP/100) ; [outgoing] exten => _X.,1,Dial(SIP/terminator/${EXTEN}) |
00:36.07 | caio1982 | tzafrir_laptop: are you working on pkg-voip these days or are you a little bit away as your coworker said? i'd ask you about my last email... that new unicall stuff |
00:36.47 | tzafrir_laptop | I'm a little bit away |
00:36.47 | dlynes_laptop | eh? I thought it was coppice that was doing unicall? |
00:37.08 | caio1982 | dlynes_laptop: debian packages, not upstream code |
00:37.09 | tzafrir_laptop | I'm now back for a while, till Monday or so |
00:37.18 | dlynes_laptop | caio1982, ah |
00:37.35 | caio1982 | tzafrir_laptop: ah, ok |
00:38.08 | tzafrir_laptop | I have a laptop there and ton of free time. But no internet connection |
00:38.37 | *** join/#asterisk detien (i=detien@unaffiliated/detienn) |
00:38.45 | dlynes_laptop | tzafrir_laptop, sounds like me...except the ton of free time part |
00:38.45 | tzafrir_laptop | I end up playing Wesnoth too long |
00:39.01 | *** part/#asterisk detien (i=detien@unaffiliated/detienn) |
00:41.21 | caio1982 | there where? |
00:41.27 | caio1982 | :P |
00:43.18 | *** join/#asterisk engineeer (i=1001@adsl-68-94-5-68.dsl.rcsntx.swbell.net) |
00:47.40 | prepaid | dlynes_laptop: sorry about the delay in responding... i'm actually using record, not monitor... i just want the user to know we're recording a 50 second clip, then warn them when there are 10 seconds up |
00:49.06 | dlynes_laptop | prepaid, there's an easier way to do it, you know? |
00:49.21 | engineeer | is there g729 support for the lastest svn branch out there somewhere or am I missing it somewhere |
00:49.41 | hads|home | engineeer: Not at this stage aparantly. |
00:49.54 | prepaid | dlynes_laptop: nope no idea, fill me in :) |
00:50.08 | dlynes_laptop | prepaid, and you don't even have to patch any C code :) |
00:50.18 | prepaid | dlynes_laptop: i'm liking where this is going... |
00:50.19 | engineeer | ok thanks about every TA I have uses g729 |
00:50.22 | dlynes_laptop | prepaid, Use the dial command to dial into a Local/extension |
00:51.03 | dlynes_laptop | prepaid, so when they call into the extension, they'll get Dialed() into a Record() application |
00:51.07 | hads|home | engineeer: I think they are going to release it sometime around when 1.4 goes into beta |
00:51.14 | Un1x | dylnes_laptop, msg please |
00:51.25 | engineeer | ok thanks for the info |
00:51.25 | dlynes_laptop | Un1x, hold your horses |
00:51.28 | Un1x | kk |
00:51.34 | dlynes_laptop | Un1x, you're not the only person i'm talking to |
00:51.39 | Un1x | sry |
00:51.53 | prepaid | dlynes_laptop: okay... but how does that help with the beep 10 seconds before end.. or is that what you're getting at now |
00:52.19 | caio1982 | hads|home: do you have URL with more info about it or something? |
00:52.24 | dlynes_laptop | prepaid, If you read the documentation for the dial command, you'll see there's some options to warn the user x number of seconds before the end of their call |
00:53.00 | dlynes_laptop | prepaid, It's a hack, but it should work :) |
00:53.11 | hads|home | caio1982: No sorry, I just picked it up somewhere, dev list or here or something. |
00:53.38 | dlynes_laptop | prepaid, linux/unix is the swiss army knife of computing; asterisk is the swiss army knife of telecommunications :) |
00:53.44 | hads|home | caio1982: So I could well be off, but that's what I heard. |
00:54.16 | prepaid | dlynes_laptop: ahhh very true.. was not aware of that feature in Dial L(x[:y][:z]) :) |
00:54.33 | caio1982 | hads|home: no problem, i'll try google... although i'm skeptical about it |
00:54.46 | dlynes_laptop | prepaid, exactly...you found the exact option i was talking about |
00:54.54 | dlynes_laptop | prepaid, just couldn't remember what the option was |
00:55.09 | prepaid | dlynes_laptop: it's okay... but i appreciate it, i never knew Dial had that option... |
00:55.13 | dlynes_laptop | prepaid, it's in there originally i think for calling card applications |
00:55.50 | SwK | anyone know how to hard factory reset Hitachi WIP5000 ? |
00:56.36 | prepaid | dlynes_laptop: yah that'd make sense... well i do appreciate the insight and the new idea, thanks! |
00:57.06 | dlynes_laptop | prepaid, well, i'm a programmer, so I always look for the easy way out :) |
00:57.15 | dlynes_laptop | prepaid, no point reinventing the wheel |
00:57.23 | prepaid | dlyes_laptop: hehehe doesn't everyone but much thanks yet again |
00:58.20 | prepaid | dlyes_laptop: so in the AGI i dial them to a record extention, but is there any easy way to get them back to the original calling AGI? |
00:58.28 | prepaid | dlyes_laptop: yah i'm obviously missing something simple within my head |
00:58.45 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
00:59.08 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
01:00.07 | BugKham | the mpg123 processes occupy 100% of my cpu |
01:00.19 | BugKham | anyone had this problem before? |
01:03.56 | Zodiacal | anyone know why parkandannounce() doesn't speak the parked #? heres the wiki for parkandannounce() http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce |
01:04.15 | Zodiacal | it does park for me tho |
01:05.12 | dlynes_laptop | prepaid, hrm....good question |
01:05.27 | engineeer | SwK http://www.abptech.com/mainpages/support/hitachi_downloads.html |
01:05.42 | prepaid | dlynes_laptop: heh yah i thought maybe i was just overlooking something stupid |
01:05.45 | dlynes_laptop | prepaid, write it as a macro? and pass a value to the macro, where the value is the extension it was dropped into, from? |
01:06.37 | dlynes_laptop | I don't know if that's possible though...I've never actually written a macro, so i'm not terribly knowledgable as to what their capabilities are |
01:07.03 | prepaid | dlynes_laptop: well hmm maybe, it's originally going to get called from an AGI since it's interacting with a database to see what the max record time can be. |
01:07.36 | dlynes_laptop | prepaid, and that's something i know even less about |
01:07.43 | dlynes_laptop | prepaid, i'm going to have to learn agi soon |
01:07.48 | dlynes_laptop | prepaid, but not just yet :) |
01:08.10 | prepaid | dlynes_laptop: well it's really simple... i appreciate your hack idea, i'll see if i can figure out a way to incorporate it, or maybe i just have to hack the source of the Record command |
01:08.27 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
01:08.47 | dlynes_laptop | prepaid, well, you've got access to the dial(L(::)) command from agi though, don't you? |
01:09.09 | prepaid | dlynes_laptop: yes i do |
01:09.21 | xbmodder_lappy | Does Digium still have its "Digium certified Expert" |
01:09.35 | ghento | Hi folks. I am having a bit of a problem with asterisk. Basically when someone calls from a cell phone, the extensions.conf works perfectly. HOwever, I just had someone call from a landline, and it wasn't working at all. It appears that Read() wasn't reading their input. Has anyone experienced this before? |
01:09.51 | dlynes_laptop | xbmodder_lappy, if you can read chan_sip.c and full understand it, I'll certify you :) |
01:10.16 | xbmodder_lappy | lol |
01:10.24 | xbmodder_lappy | dlynes_laptop, do you work for digium? |
01:10.31 | dlynes_laptop | nope :) |
01:10.36 | xbmodder_lappy | because I've always wanted to kill someone who works at digium |
01:10.43 | Un1x | lol |
01:10.59 | dlynes_laptop | xbmodder_lappy, #asterisk-dev is dominated by digium employees |
01:11.09 | dlynes_laptop | xbmodder_lappy, and a good number of them hang out in this channel, too |
01:11.27 | *** join/#asterisk andymul (n=iCallAnd@cpe-69-203-217-237.nyc.res.rr.com) |
01:11.30 | dlynes_laptop | right, russellb ? |
01:12.07 | andymul | Anyone interested in some PHP/Asteirsk programming work please PM me |
01:25.11 | *** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com) |
01:25.11 | *** mode/#asterisk [+o anthm] by ChanServ |
01:28.57 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
01:30.16 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
01:33.41 | JunK-Y | some1 has been able to send the correct event to a gxp-2000, which makes it rebooting via a sip notify? |
01:36.39 | Un1x | by the way anyone spoofed caller id with asterisk before i was told how to do it it was setcallerid command, but forgot where and how... |
01:37.25 | engineeer | there is an agi script to do that on demand |
01:37.27 | *** join/#asterisk watchy2 (n=wiit@h236.176.255.206.cable.cmdn.cablelynx.com) |
01:37.32 | watchy2 | anyone ever update a tivo or voip |
01:38.23 | Un1x | engineeer, i dont wanna use a agi script better to enter it into a conf or lift up the phone press somrthing like 11 and it ask u to enter the number u want to appear on the id |
01:38.56 | engineeer | the script does all that for U from an ext but U have to use a providerthat will support it like voipjet |
01:39.09 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
01:39.15 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
01:39.15 | Un1x | or nufone |
01:39.43 | engineeer | either or I use voipjet for my "stuff" |
01:39.53 | Un1x | engineeer where can i get the script? |
01:40.09 | engineeer | I have it on my server but I think its on the wiki somewhere |
01:40.35 | dlynes_laptop | Un1x, are you using all analog lines? |
01:40.41 | dlynes_laptop | Un1x, if so, you can't spoof the caller id |
01:40.56 | dlynes_laptop | Un1x, you can only do that if you're using a pri |
01:41.13 | dlynes_laptop | Un1x, or a voip line, and most voip providers don't support it |
01:41.39 | Un1x | dlynes analog line? |
01:41.45 | Un1x | no im using VOIP to make outgoing calls... |
01:41.48 | dlynes_laptop | Un1x, pots/pstn/... |
01:41.54 | Un1x | nopes im not using those lines... |
01:42.00 | Un1x | im using voip only on my asterisk :) |
01:42.20 | Un1x | nufone publicy supports it and i know some others ;) |
01:42.38 | dlynes_laptop | well, you told me you were using a tdm400p, so i thought you might be using one or two analog lines |
01:43.28 | Un1x | no not at all |
01:43.34 | watchy2 | data over voip is impossilbe isnt it |
01:44.00 | engineeer | the file I have is called cidspoof.agi but I dont remember if I renamed it |
01:44.09 | *** join/#asterisk ComputerWarm (n=donc@209.29.157.109) |
01:44.12 | Un1x | dlynes i started asterisk but i didn't get a tone... |
01:44.25 | Un1x | engineer wanna send? |
01:44.27 | ComputerWarm | HEllo all |
01:45.04 | dlynes_laptop | didn't get a tone? |
01:45.05 | engineeer | I can do that give me a sec to find it in the dir |
01:45.08 | dlynes_laptop | what's a tone? |
01:45.25 | dlynes_laptop | watchy2, moip? |
01:45.41 | ComputerWarm | any agi programmers in just a quick question if there is. can anyone see a reason this wouldn`t work write ("SET CONTEXT callback"); write ("EXEC GoTO s|100"); |
01:45.50 | dlynes_laptop | watchy2, not impossible...just unreliable |
01:45.50 | Un1x | dlynes dailtone... |
01:46.05 | dlynes_laptop | Un1x, you need to configure your tdm400p |
01:46.07 | dlynes_laptop | Un1x, that's why |
01:46.10 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
01:46.17 | dlynes_laptop | Un1x, and i'm not the person to talk to about that...never used one |
01:46.21 | Un1x | damn |
01:46.31 | hads|home | Why do you have a TDM400 if you aren't using analog lines? |
01:46.43 | dlynes_laptop | hads|home, he's using analog handsets |
01:46.48 | watchy2 | dlynes_laptop: i was trying to update my tivo over voip using a sipura ata it dont seem to work |
01:47.02 | dlynes_laptop | watchy2, heh..good luck on that |
01:47.03 | JunK-Y | ComputerWarm: you should stop ur agi, then start ur stuff at callback|s|100 |
01:47.12 | dlynes_laptop | watchy2, but force it to use ulaw |
01:47.14 | hads|home | Ah. |
01:47.21 | watchy2 | dl: hrm that may help? |
01:47.25 | dlynes_laptop | watchy2, you can't use any kind of compressed codec |
01:47.29 | Un1x | hads|home can ya help? |
01:47.34 | dlynes_laptop | watchy2, or you'll guarantee data loss |
01:48.03 | watchy2 | dl: it says in the info its using g711u right now |
01:48.06 | watchy2 | what should i pick |
01:48.21 | dlynes_laptop | watchy2, you definitely need to use ulaw, and make sure echo cancel isn't enabled, and make sure silence suppression isn't enabled |
01:48.44 | asterisk-dud | I have a tdm405p card for fxo ports and channel banks for fxs ports and asterisk keeps hanging up calls after about ten minutes when the come in for the fxo port and are routed to a fxs channel, can anyone help me? |
01:48.45 | watchy2 | FAX CED Detect Enable how about that |
01:48.58 | watchy2 | i dunno what that is |
01:48.59 | dlynes_laptop | watchy2, also are you trying to download this tivo over voip stuff from outside your lan? |
01:49.02 | ComputerWarm | JunK-Y: ok but if i execute the exit command before sending the caller back to the extensions config would i not lose them? |
01:49.05 | dlynes_laptop | watchy2, don't worry about that |
01:49.19 | watchy2 | dl: haha yea im trying to do it over nufone |
01:49.28 | watchy2 | i got a feeling im fighting a losing battle? |
01:49.28 | dlynes_laptop | watchy2, hah...good luck with that |
01:49.42 | watchy2 | id have better luck going through a tdm400p? |
01:49.45 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
01:49.54 | watchy2 | or something connected to the pstn? |
01:49.55 | dlynes_laptop | watchy2, you might have better luck with a premium voip provider, not a budget voip provider |
01:50.05 | dlynes_laptop | watchy2, but even then, good luck |
01:50.06 | watchy2 | well i use it for testing not production |
01:50.13 | watchy2 | would i have better luck with pstn? |
01:50.18 | dlynes_laptop | watchy2, and you'll need to use a really low baud rate, too |
01:50.36 | ComputerWarm | ? |
01:50.45 | watchy2 | yea |
01:50.54 | watchy2 | fuck it ill just quit being lazy and take it to my parents |
01:50.56 | dlynes_laptop | watchy2, you shouldn't have any problem with pstn, but again, if you're going to go pstn->ulaw->ata->modem, you'll need to use 9600bps or lower |
01:51.15 | watchy2 | yea i dont think i can force my tivo to go that low |
01:51.18 | watchy2 | so its impossible |
01:51.29 | watchy2 | how do folks do pstn to ata to fax? |
01:51.39 | dlynes_laptop | 9600 baud or lower |
01:51.46 | watchy2 | ah |
01:51.46 | dlynes_laptop | and it's not 100% reliable, either |
01:51.54 | watchy2 | so most people had dedicated fax lines i take it/ |
01:52.04 | dlynes_laptop | watchy2, pretty much, yeah |
01:52.20 | JunK-Y | u can save it in db? |
01:52.43 | watchy2 | dlynes_laptop: ok well ill jus take it to my parents when the upgrades out |
01:52.46 | dlynes_laptop | watchy2, i've seen people have good luck using fax2email -> internet -> email2fax using hylafax for the fax servers |
01:53.02 | watchy2 | ah |
01:53.10 | watchy2 | but no one sends fax through asterisk |
01:53.18 | watchy2 | not in a production enviroment |
01:53.19 | dlynes_laptop | watchy2, yes they do |
01:53.27 | dlynes_laptop | like i said..hylafax |
01:53.29 | watchy2 | oh |
01:53.29 | engineeer | un1x or Unix my screen cleared |
01:53.37 | engineeer | Un1x? |
01:53.46 | dlynes_laptop | they use iaxmodem's virtual modems in combination with hylafax |
01:53.53 | dlynes_laptop | they seem to get pretty good reliability with that |
01:53.59 | watchy2 | oh |
01:54.16 | dlynes_laptop | other people get good reliability with app_txfax/app_rxfax |
01:54.24 | dlynes_laptop | however, i've had 0% success rate with that |
01:54.35 | *** join/#asterisk kb3nnj (n=theblue@c-69-140-159-42.hsd1.md.comcast.net) |
01:54.40 | watchy2 | so like this co i deployed a voip box to has been having major issues |
01:54.47 | watchy2 | so i do some checking using that zttest shit |
01:54.53 | watchy2 | there irq shit is REALLY bad % |
01:54.55 | andymul | Anyone interested in some PHP/Asterisk programming work please PM me |
01:55.04 | dlynes_laptop | watchy2, yeah...i ran into that, too |
01:55.06 | watchy2 | i ran fucking hdparm -t and got 12mb/s |
01:55.13 | dlynes_laptop | watchy2, couldn't get a zttest score over 88% |
01:55.16 | watchy2 | i ran it on another box and got 150 hahah |
01:55.24 | watchy2 | dl: replacing the box fix alot of issues/ |
01:55.33 | dlynes_laptop | watchy2, so i said forget it, tossed the digium card, went sangoma, never had a problem since |
01:55.39 | watchy2 | haha |
01:55.43 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
01:55.52 | watchy2 | im having major issues with a tdm400p with echos on 8 lines |
01:56.00 | watchy2 | i want to stab digium in the fucking throat for this bs |
01:56.11 | dlynes_laptop | i've got a sangoma a200 with hwec, and i've got 0 echo |
01:56.14 | ComputerWarm | anyone here willing to give me a head with this issue with the agi script? |
01:56.18 | watchy2 | but im gonna replace the emachine they gave me to be a server |
01:56.20 | *** part/#asterisk kb3nnj (n=theblue@c-69-140-159-42.hsd1.md.comcast.net) |
01:56.24 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
01:56.24 | dlynes_laptop | it sounds better than a nortel system |
01:56.26 | watchy2 | and replace it with another box |
01:56.42 | CrashHD | dlynes can you recommend a good premium voip provider ? |
01:56.44 | watchy2 | if replacing it with another box dont work im gonna call digium and tlel them shove the cards up there ass |
01:56.51 | dlynes_laptop | CrashHD, ME! |
01:56.53 | dlynes_laptop | heh |
01:56.58 | dlynes_laptop | j/k |
01:57.04 | Jason99 | hello, i'm just wondering if someone can explain to me how the MWI works... What triggers Asterisk to tell the phone there is a VM and how does Asterisk know which phones to tell? I could not find any documentation about this.. |
01:57.14 | CrashHD | OEM you're the one I need to talk to I've trying to remember who I talked to on IR see about Felipe service |
01:57.15 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
01:57.15 | dlynes_laptop | we don't do voip to residential customers |
01:57.30 | dlynes_laptop | only commercial |
01:57.38 | CrashHD | and testing out speech to text so you'll need to excuse my language |
01:57.45 | watchy2 | do you use myspace dlynes_laptop |
01:57.50 | dlynes_laptop | nope |
01:57.54 | CrashHD | My space is the devil |
01:58.01 | watchy2 | you should i want to add u to my friends list |
01:58.04 | dlynes_laptop | it's just a blogging site, isn't it? |
01:58.05 | watchy2 | and we can be emo together |
01:58.10 | dlynes_laptop | emo? |
01:58.13 | watchy2 | dont you want to be emo with me |
01:58.14 | dlynes_laptop | sounds kinky |
01:58.19 | CrashHD | My space is like a virus |
01:58.22 | watchy2 | we can wear makeup and lipstick and cut on each other |
01:58.24 | watchy2 | itll be great |
01:58.59 | Jason99 | anyone know anything about message waiting indicator ? |
01:59.05 | dlynes_laptop | CrashHD, I think watchy2 might be your neighbor on myspace |
01:59.18 | dlynes_laptop | Jason99, just ask your questio |
01:59.24 | dlynes_laptop | ~suggestions |
01:59.26 | jbot | rumour has it, suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite ... |
01:59.27 | Jason99 | I did above |
01:59.28 | CrashHD | lol |
01:59.35 | Jason99 | What triggers Asterisk to tell the phone there is a VM and how does Asterisk know which phones to tell? I could not find any documentation about this.. |
01:59.35 | Zodiacal | anyone know of a way to saydigits when a blind transfer gets initated to the user that initated the blind transfer? |
01:59.43 | CrashHD | myspace is just myspace |
01:59.44 | Zodiacal | or is a blind transfer final and cuts off the connection |
01:59.45 | CrashHD | lame |
01:59.47 | dlynes_laptop | Jason99, the phone subscribes to mwi indication |
01:59.56 | watchy2 | hehe |
02:00.05 | watchy2 | i want to get high and stab kittens |
02:00.07 | *** part/#asterisk Amilcar_ (n=amilcar@201.34.202.17) |
02:00.20 | dlynes_laptop | Jason99, each time it checks mwi status, asterisk will let it know how many messages there is |
02:00.24 | rene- | wtf watchy2 |
02:00.32 | watchy2 | rene: im emo dude |
02:00.33 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
02:00.33 | *** mode/#asterisk [+o mog_home] by ChanServ |
02:00.38 | Jason99 | dlynes_laptop: ok, I must be doing something wrong then.. ok thanks, let me take a look |
02:00.39 | dlynes_laptop | ~emo |
02:00.52 | rene- | ahhh |
02:00.59 | dlynes_laptop | ~wiki emo |
02:01.10 | Jason99 | dlynes_laptop: Asterisk knows by the mailbox field in the sip.conf? |
02:01.21 | dlynes_laptop | Jason99, correct |
02:01.29 | dlynes_laptop | Jason99, mailbox=101 |
02:01.33 | dlynes_laptop | Jason99, mailbox=101@default |
02:01.44 | dlynes_laptop | Jason99, mailbox=101@mycontext |
02:01.54 | dlynes_laptop | default is the default context for voicemail |
02:02.00 | *** join/#asterisk engineeer (i=1001@adsl-68-94-5-68.dsl.rcsntx.swbell.net) |
02:02.04 | dlynes_laptop | so if you don't specify @context, it'll assume default |
02:02.21 | Jason99 | dlynes_laptop: but which context would I want to put? I don't get that |
02:02.42 | watchy2 | whatever context your set your mailboxes under homie |
02:02.56 | dlynes_laptop | Jason99, are all of your mailboxes defined in the 'default' context in your voicemail.conf file? |
02:03.01 | rene- | watchy2 i'd tought that emo dudes would be very much against stabbing kittens |
02:03.13 | Jason99 | dlynes_laptop: ah, yes I get it |
02:03.19 | watchy2 | rene: well i dunno im not really emo, i just play a emo fag on irc? |
02:04.04 | dlynes_laptop | wtf is emo music? |
02:04.09 | rene- | somebody told me emo really meant gay |
02:04.26 | dlynes_laptop | and i thought a screeching weasel was some kinda alcoholic drink? |
02:04.31 | watchy2 | apparently emo music is dead |
02:04.37 | watchy2 | it died like 10 years ago |
02:04.41 | watchy2 | seriously |
02:04.44 | rene- | AIDS? |
02:04.54 | watchy2 | maybe rene, it did go out with the 80s i think |
02:05.05 | engineeer | Un1x did U get you file? |
02:05.23 | *** join/#asterisk AJaymn (n=FreePBX8@156-77.dsl.scc.net) |
02:07.10 | rene- | off |
02:08.52 | watchy2 | how the fuck do you copy in putty? |
02:08.55 | watchy2 | just highlight right |
02:10.47 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
02:14.41 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
02:15.17 | watchy2 | i want to meet a hotchick who shows me how to use asterisk and marrys me |
02:15.34 | Assid | watchy2: good luck |
02:15.59 | watchy2 | i can barely find girls who arent fat |
02:16.00 | watchy2 | :( |
02:16.55 | *** join/#asterisk BugKham (i=CKGLOB@221.128.110.41) [NETSPLIT VICTIM] |
02:16.55 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) [NETSPLIT VICTIM] |
02:30.31 | dlynes_laptop | watchy2, stay out of mcdonald's then |
02:30.50 | dlynes_laptop | watchy2, and the bar scene |
02:31.05 | dlynes_laptop | watchy2, take up felching |
02:31.07 | watchy2 | haha |
02:31.54 | rob0 | watchy2 meets these fat girls, and to be polite, says, "For a fat girl you sure don't sweat too much!" |
02:32.06 | watchy2 | haha |
02:32.09 | watchy2 | im fat to |
02:32.25 | dlynes_laptop | all girls are the same, fat or not |
02:32.32 | dlynes_laptop | just roll them in flour and find the wet spot |
02:32.35 | rob0 | (Gomer Pyle, USMC ... an obscure reference to an old sitcom.) |
02:34.01 | ComputerWarm | guys question or maybe point me to some reading if you will i am trying to figure out how to update a mysql table of how long the caller was online and delete that amount from his account |
02:34.23 | ComputerWarm | i need to do this after he hangs up. and everytime i try the script terminates when the carrier hangs up |
02:34.52 | CrashHD | I would do agi |
02:35.34 | ComputerWarm | CrashHD: thats how i am attempting to do it but when the caller hangs up it doesn`t update the database i can`t figure out how to get the script to continue until its completed the update |
02:38.00 | CrashHD | to continue? or to not continue until the agi call is done? |
02:38.38 | ComputerWarm | continue sorry |
02:39.25 | CrashHD | that was a question |
02:39.54 | CrashHD | you do not want to continue until the AGI has completed the update correct? |
02:40.01 | CrashHD | ohh |
02:40.09 | CrashHD | I see |
02:40.10 | CrashHD | hmm |
02:40.19 | CrashHD | soo this is an incoming call? |
02:40.30 | ComputerWarm | actually out going call |
02:40.32 | CrashHD | or outgoing? |
02:40.44 | CrashHD | using the continue after disconnect flag for the dial command? |
02:40.57 | ComputerWarm | didn`t know there was one |
02:41.06 | CrashHD | g? I think |
02:41.07 | CrashHD | let me look |
02:41.31 | CrashHD | "g: When the called party hangs up, exit to execute more commands in the current context. " |
02:41.51 | CrashHD | I think that will fix your problem |
02:41.58 | ComputerWarm | ok thank you |
02:42.36 | ComputerWarm | once this is fixed its just a matter of figuring out the math of the billing increments and then test this thing |
02:43.00 | *** part/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
02:43.02 | CrashHD | sounds like a whole ton of fun |
02:43.05 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
02:43.56 | ComputerWarm | CrashHD: ya not really.. it was until i got stock on this part... but now i think with what you recommended it should work |
02:43.58 | *** join/#asterisk racter (n=racter@c-67-167-206-229.hsd1.il.comcast.net) |
02:44.19 | CrashHD | walls are meant to be climbed |
02:44.38 | racter | hey folks -- i'm looking for a really simple IVR solution and asterisk just looks so complicated; do i really need to buy all this hardware just to set up a one-line touchtone-navigable system? |
02:45.27 | tuck3r | a few days ago i upgraded to kernel 2.6.17 and zaptel 1.2.7, now in /var/log/messages i'm getting lots of "rtc: lost some interrupts at 1024Hz." not sure what caused it, any ideas? |
02:45.32 | *** join/#asterisk sp0n9e (n=sp0n9e@69.12.216.48) |
02:45.40 | tuck3r | racter: just get a cheap ATA |
02:47.17 | racter | tuck3r: if i buy an ata, asterisk will recognize it and i can use asterisk to program a simple IVR? |
02:47.48 | tuck3r | "recognize", no you have to set it up but yeah |
02:48.01 | racter | cool |
02:48.13 | watchy2 | hey whats it take to configure an fxs port on a tdm400p? besides /etc/zaptel.conf and zapata.conf |
02:48.23 | watchy2 | ive never confed a fxo port on a tdm |
02:48.52 | watchy2 | i mean an fxs |
02:49.25 | dlynes_laptop | watchy2, that's it |
02:49.37 | dlynes_laptop | watchy2, maybe adjust your gains and that kinda thing, too |
02:49.37 | watchy2 | and you can pick up the phone and dial? |
02:49.51 | watchy2 | it doesnt need to authenticate or anything? |
02:49.51 | hads|home | Depending on your dialplan, yes. |
02:49.58 | dlynes_laptop | watchy2, well, obviously you need to define a context for it, and write that context in your dial plan |
02:50.05 | watchy2 | yea |
02:50.06 | dlynes_laptop | watchy2, and set up an extension for it |
02:50.11 | racter | thx for the tip, tuck3r |
02:50.23 | watchy2 | but if its setup in /etc/zaptel.conf zapata.conf should it get a dialtone after that? |
02:50.47 | hads|home | Once you have loaded the module and run ztcfg, yes. |
02:50.53 | watchy2 | after an asterisk restart that it |
02:51.10 | hads|home | Yes. |
02:51.46 | watchy2 | hrn i steped un1x through it and hes got asterisk back but hes not getting a dialtone for some reason |
02:51.57 | watchy2 | <Un1x> 1 default en default |
02:52.08 | watchy2 | port 1 and 2 are fxs ports on a tdm400p |
02:53.59 | watchy2 | http://pastebin.ca/104673 |
02:54.07 | watchy2 | does that look correct for a fxsport on a tdm400p? |
02:55.03 | hads|home | looks alright at a glance |
02:56.09 | ComputerWarm | CrashHD: at times i am thinking its just to much work but oh well :-) |
02:56.17 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
02:56.29 | sp0n9e | how hard would it be for an average sysadmin to create a pbx with asterisk that has 10 incoming lines and connects 16 phones and provides a queue and an autoattendant? |
02:56.40 | watchy2 | you know linux? |
02:56.43 | ComputerWarm | not hard at all |
02:56.47 | sp0n9e | pretty well |
02:56.47 | ComputerWarm | take a few minutes to read |
02:56.55 | watchy2 | then a few hours time and shit |
02:56.58 | sp0n9e | i've been flipping through a draft manual |
02:57.09 | watchy2 | its pretty cool. i did it with 40 phones sp0n9e |
02:57.14 | watchy2 | and im a newbie to asterisk |
02:57.23 | watchy2 | i got like automated directory and shit up its pretty cool |
02:57.35 | sp0n9e | okay, i feel comfortable editing conf files, etc...just new to a lot of the phone terminology |
02:57.46 | watchy2 | yea get the asterisk book |
02:57.51 | sp0n9e | place of employment is moving and everything is getting upgraded |
02:57.53 | watchy2 | itll explain alot and its good to read at home |
02:58.00 | sp0n9e | recommendations? i'll get my boss to buy one |
02:58.09 | watchy2 | its a bright yellow book i dunno what its called |
02:58.47 | sp0n9e | how much proc/memory would i need for 10 incoming lines and 16ish phones? |
02:58.53 | hads|home | ~thebook |
02:58.55 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
02:59.13 | sp0n9e | oooooOOOOooooo |
02:59.14 | watchy2 | sp0n9e: a decent box i wouldnt use some gay amd shit |
02:59.34 | hads|home | riiiight. |
02:59.36 | tuck3r | amd shit? |
02:59.43 | watchy2 | im not a fan of amd |
02:59.54 | watchy2 | intels new proccessor has raped them |
02:59.57 | tuck3r | nobody cares really |
02:59.59 | hads|home | Obviously. Although that doesn't make them shit. |
03:00.10 | sp0n9e | the budget for this is probably around $1500 |
03:00.25 | watchy2 | you can easily get a box to handle all that on $1500 |
03:00.32 | sp0n9e | i would prefer for everything to fit in the rack :) |
03:00.34 | watchy2 | now phones is a different story |
03:01.31 | sp0n9e | what am i looking at for phones? |
03:01.34 | watchy2 | id recommend polycoms, ciscos are nice but i liked my polys better |
03:01.50 | sp0n9e | what kinds of protocols should i look for? |
03:01.53 | watchy2 | sip |
03:02.04 | watchy2 | you gonna have a secretary ? |
03:02.12 | sp0n9e | not i |
03:02.12 | watchy2 | watching all lines and doing transfers and stuff? |
03:02.14 | *** join/#asterisk EvilDeshi (i=evildesh@oxford-bb-occam3-ws-100.dsl.maqs.net) |
03:02.21 | sp0n9e | we may...eventually |
03:02.22 | watchy2 | is there gonna be 1 chick answering all the phones? |
03:02.25 | watchy2 | ah |
03:02.38 | watchy2 | then just get some 301s i think or 501s, look at em both and compare |
03:02.38 | sp0n9e | the ebay guy will probably do it |
03:02.52 | *** part/#asterisk racter (n=racter@c-67-167-206-229.hsd1.il.comcast.net) |
03:04.01 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
03:04.06 | watchy2 | well it shouldnbt be to hard to do |
03:04.13 | watchy2 | plus folks in this channel are always willing to help |
03:04.34 | sp0n9e | yeah, the channel looks big enough that there's help here 24/7 :) |
03:04.44 | tuck3r | just don't take processor advise from them |
03:04.45 | watchy2 | yea |
03:05.00 | watchy2 | bookmark voip-info.org to |
03:05.04 | watchy2 | that websites great |
03:05.41 | hads|home | ~docs |
03:05.42 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
03:05.51 | sp0n9e | wow |
03:05.54 | engineeer | polys have the best speaker phones as well |
03:08.22 | sp0n9e | i'm getting 10 lines from a cox "combined services" fiber line...it's going to run through some type of cisco device (forgot the name) what type of hardware will i need? |
03:08.32 | watchy2 | hrm |
03:08.39 | watchy2 | is it coming out to be normal pstn lines? |
03:09.14 | watchy2 | im guessing cox is going from digital cable to standard analog |
03:09.18 | sp0n9e | i'm looking through some of the information i have access to |
03:09.23 | sp0n9e | well, it's off of fiber |
03:09.23 | watchy2 | youll need analog cards |
03:09.34 | sp0n9e | but i think it may come out analog |
03:09.54 | watchy2 | if its coming out analog get some of those sangora analog cards with built in hardware echo |
03:10.07 | watchy2 | i wont ever buy a card without built in hw echo again |
03:13.04 | sp0n9e | i'll do the rest of the research on-the-clock :) |
03:13.07 | sp0n9e | thanks for the feedback |
03:13.25 | *** part/#asterisk sp0n9e (n=sp0n9e@69.12.216.48) |
03:15.25 | watchy2 | we got dewds fxs workin |
03:15.35 | watchy2 | i think he just had it pluged into the wrong hole |
03:17.11 | watchy2 | can something be in more then 1 context in *? |
03:17.27 | *** part/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
03:18.47 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-8.getnet.net) |
03:19.12 | watchy2 | i need some more kidbop cds |
03:28.10 | Assid | woohoo |
03:28.11 | Assid | wassup |
03:29.28 | Jason99 | One I have set a variable, how would I clear it? |
03:29.58 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
03:30.24 | Un1x | ok |
03:30.31 | Un1x | anyone here know someone who uses splitfinity |
03:35.15 | engineeer | did U get your file? |
03:36.31 | engineeer | that was for Un1x |
03:42.48 | *** join/#asterisk Clausian (i=reginald@203-206-65-20.dyn.iinet.net.au) |
03:43.18 | Clausian | anyone here got FWD working with *? |
03:47.55 | *** join/#asterisk bmg505 (n=leon@dsl-165-156-57.telkomadsl.co.za) |
03:52.09 | Un1x | engineeer yes i got the agi script |
03:52.12 | Un1x | for cidspoofing :) |
03:52.58 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
03:54.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
04:04.10 | *** join/#asterisk xanderp (n=pphillip@74.133.18.245) |
04:05.21 | xanderp | could someone please give me the $.02 explanation as to the major difference between astlinux and trixbox? |
04:06.10 | Un1x | ALOT! |
04:06.20 | Un1x | features support flexibility |
04:06.22 | Un1x | many things my freind |
04:07.05 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
04:09.14 | xanderp | i've not built a pbx before (* or any other) but am fair at linux. i want to tinker with setting up a few phones connected via sip to an box. my end goal is to setup extensions for each of the bedrooms/kitchen/basement/etc... and have voicemail, etc, and have them able to call out my x100p over my vonage account. any suggestions as to a good distro to start with to learn? |
04:09.59 | xanderp | oops.. the asterisk's changed my text to bold :) |
04:10.20 | hads|home | xanderp: Trixbox is based on CentOS, it's large and includes FreePBX, FOP and many other things. It is designed to be run on a standard PC. Astlinux is a custom built distro specifically for running Asterisk on embedded systems such as a Soekris or Wrap and such. |
04:10.56 | hads|home | xanderp: IMHO the best way to start is to take the distro that you feel most comfortable with and then download and install Asterisk from source. |
04:11.32 | CrashHD | agree'd |
04:11.39 | xanderp | ok.. i'm going to install on a dual p3 550mhz box with 512meg ram. it's got a 9gig hd. it's an older machine i don't use for anything, so i thought i'd give this a go on it. |
04:12.07 | hads|home | That box will be plenty for your purpose. |
04:12.30 | xanderp | gentoo is my distro of choice, but to compile everything on this old tank would take forever... ;) |
04:12.40 | hads|home | You will probably run into issues with the X100P though, most people do. |
04:12.56 | xanderp | echo? |
04:13.06 | hads|home | Yes. |
04:13.12 | xanderp | (grrrrr) |
04:13.20 | xanderp | i knew it was cheap for a reason |
04:13.37 | benjk | X100P or more precisely Ambient MD3200 softmodems are good for providing a Zaptel timing source |
04:13.43 | EyeCue | It's been fun on freebsd, far less of a heacache than i expected :) |
04:13.49 | benjk | they are not good for anything else |
04:14.11 | mog_home | x100p arent that bad |
04:14.35 | benjk | if they were manufactured a few years ago, they might be acceptable |
04:14.46 | hads|home | xanderp: I'm not in the US, but Vonage is a VoIP provider right? There's not much point going from VoIP -> analog -> VoIP |
04:14.52 | benjk | but if they are manufactured more recently, they are garbage |
04:14.52 | mog_home | yeah i have true ones purchased by digium ^_^ |
04:14.58 | mog_home | i didnt know that |
04:15.07 | mog_home | i knew that intel stopped making the chip, or so i heard |
04:15.15 | benjk | that's because the chipset is not being manufactured anymore |
04:15.21 | symlink | mog_home: you just grab everything you can eh? |
04:15.32 | mog_home | whats that file |
04:15.36 | xanderp | hads|home: this is just to get my head around asterisk, and then possibly replace the vonage for long term. |
04:15.37 | benjk | first Intel acquired Ambient, then they stopped making the chip |
04:15.52 | symlink | mog_home: x100ps... *cough*usb adapters*cough* |
04:15.52 | hads|home | xanderp: Cool. |
04:15.55 | benjk | now the Chinese manufacturers use refurbished chips |
04:16.05 | mog_home | s100us sucj |
04:16.08 | mog_home | er suck |
04:16.15 | symlink | they don't exist. |
04:16.20 | mog_home | tricky chinese theives |
04:16.21 | benjk | and even worse, some use left over chips that didn't pass quality control |
04:16.49 | benjk | but they are good as Zaptel timing sources |
04:17.01 | benjk | just don't use them as FXO interfaces |
04:17.04 | xanderp | at that point i would replace the x100p with straight voip to a provider, and possibly hardware ip phones, but for now, it's just alpha testing |
04:17.29 | benjk | you can always buy a Sipura3000 SIP-FXO adapter |
04:17.39 | benjk | its about 70 USD or so |
04:17.58 | mog_home | or tdm400p |
04:18.00 | benjk | works reasonably well, certainly much better than those junk modems |
04:18.03 | mog_home | i think thats 00 or so |
04:18.04 | hads|home | xanderp: Fair enough, there's nothing wrong with what you have for testing. You may find it works fine for what you need. |
04:18.08 | mog_home | er 100 |
04:18.21 | benjk | TDM400 is not that economical if all you need is a single port |
04:18.33 | mog_home | true |
04:19.08 | benjk | also, Sipura has support for many more caller ID schemes than Zaptel drivers |
04:19.10 | xanderp | i was looking at the sipra 2002 linksys box that is locked to earthlink, but didn't know if i could unlock it easily. they were only 60 bucks. |
04:19.28 | hads|home | benjk: Yes but the TDM400 can do things like distinctive ring and such that the SPA3102 can't. |
04:19.30 | benjk | locked ones are probably not such a good idea |
04:19.38 | *** join/#asterisk tempest1 (n=asf@adsl-144-60-181.chs.bellsouth.net) |
04:19.45 | Un1x | i need some help |
04:19.48 | Un1x | when somone calls me |
04:19.51 | Un1x | they get, Congrats thing |
04:19.58 | benjk | only makes sense if it does detect caller ID properly though |
04:20.04 | Un1x | from asterisk on how they successfuly installed asterisk |
04:20.08 | mog_home | Un1x: its your default context stuff |
04:20.12 | mog_home | its still active |
04:20.19 | xanderp | are there ANY inexpensive adapters that can do like 4 lines of analog phones fair enough for home use? |
04:20.22 | mog_home | you need to edit extensions.conf |
04:20.32 | mog_home | tdm400p |
04:20.48 | xanderp | mog_home: that for me? |
04:20.51 | Un1x | yea imm do that in a bit |
04:20.53 | benjk | so if you happen to be in a place which uses a caller ID scheme that Zaptel doesn't support, then your disctintive ring feature isn;t going to do you any good |
04:20.53 | mog_home | yes |
04:21.05 | mog_home | lol true benjk |
04:21.05 | xanderp | thanks will check on it. what should i expect to $? |
04:21.16 | mog_home | you want 4 lines in or out |
04:21.25 | mog_home | i think its around 300 for that |
04:21.45 | xanderp | 4 lines in the house that can call each other, the outside world, or be called by the outside world. |
04:21.53 | benjk | its generally about 70-80 USD per port no matter where you go |
04:22.01 | hads|home | benjk: Huh? callerid is seperate from distinctive ring. |
04:22.08 | xanderp | thanks, nice to have a good rule of thumb |
04:22.08 | mog_home | till you add a lot of ports that is |
04:22.22 | benjk | it is separate, but you do distinctive ring to indicate who is calling |
04:22.25 | mog_home | http://forum.gbadev.org/viewforum.php?f=20 |
04:22.35 | benjk | so if you don't know who is calling, then what good is distinctive ring? |
04:22.36 | mog_home | oops other thing im playing with |
04:23.15 | xanderp | would i be ok using 2 sipra 2002's with 2 ports each? or is that overly complicated for what i am trying to do? |
04:23.29 | benjk | sipura 2000 is only for telephones |
04:23.35 | benjk | not for telephone *lines* |
04:23.38 | hads|home | benjk, maybe it's different over here. Distinctive ring is used on POTS lines as a way of adding an extra number. |
04:23.57 | hads|home | benjk: So you can differentiate between faxes or whatever. |
04:24.09 | xanderp | benjk: so they won't sip 2 phones to the pbx as 2 different extensions? |
04:24.11 | benjk | that's inbound distinctive ring yes |
04:24.18 | *** join/#asterisk s0lid (n=jlq@210.213.240.222) |
04:24.20 | benjk | but that too, depends on which caller ID scheme is used |
04:24.46 | benjk | so if your Zaptel device or driver doesn't support the caller ID scheme your telco uses than that feature will not be available to you |
04:25.38 | benjk | xanderp, there are two kinds of analog ports |
04:25.48 | benjk | ones for connecting to the telco, called FXO ports |
04:26.07 | benjk | and ones for connecting a telephone, called FXS ports |
04:26.27 | [TK]D-Fender | benjk : He's got his head on straight.... you need to follow.... |
04:26.31 | benjk | if you want to hook up two analog phones, then the sipura 2000 is fine, cause it has 2 FXS ports |
04:37.03 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
04:37.03 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
04:37.06 | xanderp | so to tinker and learn would trixbox be a good place to get my feet wet and have it at least hold my hand a little in the installation |
04:37.12 | Clausian | can you try ringing 789140? |
04:37.20 | [TK]D-Fender | benjk : Though I'd love to know what you class as "nice" at $95 |
04:37.21 | benjk | trixbox isn't for tinkering |
04:37.31 | benjk | ACT 104 |
04:37.47 | [TK]D-Fender | benjk : .jp available model? |
04:37.57 | xanderp | isn't it the next rev of the @home distro? |
04:38.01 | [TK]D-Fender | benjk : never heard of it over here. |
04:38.08 | benjk | yes, they also have JP display capable firmware |
04:38.16 | hads|home | xanderp: IMHO the best way to start is to take the distro that you feel most comfortable with and then download and install Asterisk from source. |
04:38.20 | [TK]D-Fender | xanderp : you won't learn anything with Trixbox. and it is the new name for A@H |
04:38.35 | benjk | its a Taiwanese manufacturer, called Advanced Century Telecommunications (or Technology?) |
04:38.37 | xanderp | ok, i'll roll my own... |
04:38.40 | [TK]D-Fender | benjk : Got a link... would love to see something cheap AND good... |
04:38.57 | benjk | let me try to find it |
04:38.58 | quid2478 | I'm using TB... but to be honest, I'm doing most of my work with the config files using vi. :) |
04:39.08 | xanderp | (3 days worth of gentoo compiling is not something i look forward to, but i do so love gentoo!) |
04:39.29 | Un1x | i hate vi |
04:39.32 | Un1x | it's terrible |
04:39.38 | [TK]D-Fender | file not found!!!!! |
04:39.43 | benjk | http://www.act-tel.com.tw/_pg/products/productItemR.ASP?ContentsManageID=11&UnitsManageName=IP%20Phone |
04:39.47 | quid2478 | well that and nano, though I admit... i'm used to vi |
04:39.47 | symlink | I know, it's sad :( |
04:39.56 | benjk | and they do have IAX2 firmware |
04:40.03 | xanderp | nano is my savior... i never can remember vi commands |
04:40.17 | [TK]D-Fender | benjk : I've seen that exact frame remarked under several names including GNET |
04:40.18 | benjk | but you need to bother them a lot to get it |
04:40.23 | Clausian | can someone ring fwd 789145 please? |
04:40.27 | quid2478 | xanderg: the thing that is drawing more to nano, is that it's the same editor off of old PC BBS software I used to use |
04:40.28 | benjk | yes, they are more in the OEM business |
04:40.40 | quid2478 | Clausian: Use the "ring me" feature of FWD |
04:40.41 | benjk | many companies buy them in bulk and sell them under their own brand |
04:40.46 | Clausian | it never works |
04:40.50 | Clausian | :X |
04:41.04 | Clausian | my softphone is registered fine though |
04:41.06 | [TK]D-Fender | benjk : Believeable, but I strongly suspect the quality wouldn't fall under my idea of "good" |
04:41.09 | benjk | Here in Japan, Fujistu rebrand those with a nicer enclosure and sell them against NECs rebranded Cisco 79xx |
04:41.23 | quid2478 | I trashed FWD... who needs it |
04:41.31 | benjk | they are good phones |
04:41.58 | benjk | have you ever used any of those? |
04:42.26 | [TK]D-Fender | benjk : No, afer seeing GNET sell the ATCOM PA1688 stuff it made me shudder.... |
04:42.28 | benjk | the display is a little small |
04:42.44 | benjk | its not an ATCOM PA1688 phone |
04:42.46 | [TK]D-Fender | benjk : Looks too 1980 Radio-Shack for em... |
04:43.05 | benjk | the industrial design of the enclosure isn't all too hip, true |
04:43.09 | [TK]D-Fender | benjk : jsut iguring any company selling it wouldn't be selling anything worthwhile :) |
04:43.16 | benjk | but the phone is solid quality |
04:43.19 | xanderp | well, it's late here, thanks for all the info everyone! |
04:43.35 | benjk | also, it looks better in reality than it does on the photo |
04:43.40 | *** part/#asterisk xanderp (n=pphillip@74.133.18.245) |
04:43.57 | [TK]D-Fender | benjk : Yeah, its kinda... BLEH. But if you its FUNCTIONAL, then it has a place. Then again, what is your take on Aastra? Very classic look, basic usability, good price IMO. |
04:43.58 | coppice | people only judge phones by the case. how it works is largely irrelevant :-) |
04:44.03 | benjk | the buttons are better than anything I have seen in the range up to 200 USD |
04:44.29 | benjk | yeah Aastra is a nice phone too |
04:44.36 | benjk | looks more stylish |
04:44.39 | coppice | the buttons on a $1 analogue phone are fine, but on $50 IP phones they are nasty. something is definitely wrong there |
04:44.48 | benjk | Taiwanese aren't that good at industrial design |
04:45.03 | benjk | coppice, :D |
04:45.51 | [TK]D-Fender | coppice : Glad I pay over $100 for mine :) |
04:46.10 | [TK]D-Fender | coppice : So I can get that incredible 1$ analog phone button feel! |
04:46.13 | benjk | in my opinion, the ACT 104s is the most solid IP phone in the price range up to 200 USD |
04:46.30 | hads|home | That's fairly bold |
04:46.32 | coppice | you're glad to pay >$100 for something that costs $15? weird |
04:47.04 | Un1x | why is zaptel doing this |
04:47.04 | Un1x | <PROTECTED> |
04:47.04 | Un1x | <PROTECTED> |
04:47.04 | Un1x | <PROTECTED> |
04:47.04 | Un1x | <PROTECTED> |
04:47.06 | benjk | the looks are of course a different thing |
04:47.10 | [TK]D-Fender | benjk : I cannot for a second believe that it competes with a Polycom IP 501....... |
04:47.13 | Un1x | i pick up the phone and after i press like 4t5h digit to dail |
04:47.17 | Un1x | i get busy signalish |
04:47.24 | Un1x | zaptel hanging up on me why is that |
04:47.29 | Clausian | i have asterisk registered on fwd, and a softphone registered on another account. whenever i dial asterisk's fwd number, the softphone tells me its busy. why is asterisk doing this? |
04:47.39 | benjk | well, my customers clearly like the ACT better than the Polycoms |
04:47.54 | [TK]D-Fender | coppice : Find me a $15 source for this kind of quality then :) |
04:48.08 | benjk | nobody is exactly thrilled by its looks, but functionality wise its there |
04:48.31 | coppice | benjk: the taiwanese are good at industrial design. the ODMs there just know the case will get reworked to suit a big customer, so the original case the make isn't that important. you see this with lots of products. The final case will also be a taiwan design, but will look good |
04:48.44 | [TK]D-Fender | benjk : if you say so.. I can't fathom it. 2 line LCD vs pixel, high end speakerphone, nice solid feel, PoE options, . |
04:48.54 | benjk | coppice, fair enough |
04:49.23 | Un1x | Comon someoen help please... |
04:49.26 | benjk | as I said, Fujitsu is selling the P104s here in Japan with their own (more stylish) enclosure and sell it against NECs rebranded Cisco 79xx |
04:49.38 | benjk | for 400 USD |
04:49.46 | benjk | :) |
04:50.36 | benjk | they do have customised firmware though |
04:51.37 | Un1x | anyone know why |
04:51.38 | Un1x | zaptel fires up the channel and kills it |
04:52.30 | De_mon | Un1x you broke it |
04:52.32 | benjk | the P123 doesn't look too bad |
04:52.34 | benjk | http://www.act-tel.com.tw/_pg/products/productItemR.ASP?ContentsManageID=69&UnitsManageName=IP%20Phone |
04:52.46 | benjk | bit of a Cisco knock off |
04:52.50 | symlink | Un1x: does the number you're dialing exist in the context that it will be searching? |
04:53.01 | benjk | I will have to enquire about those |
04:53.13 | Un1x | what do you mean symlink... |
04:53.23 | Un1x | im dailing a regular number... |
04:53.30 | [TK]D-Fender | benjk : Only the color scheme... |
04:53.48 | coppice | benjk: well the ciscos are made in taiwan too :-) |
04:53.50 | benjk | the general look |
04:53.50 | symlink | Un1x: you have to tell Asterisk how to handle that stuff, it's not psychic - you tell it "if I dial this number, do these things" "if I dial a number matching this, do these things" |
04:53.53 | [TK]D-Fender | benjk : then again, I've seen that combo around all over anyways.... not much of a strech |
04:54.02 | benjk | the display is still kinda smallish though |
04:54.10 | benjk | could be a little larger |
04:54.14 | Un1x | oh yes thats in my extensions.conf |
04:54.21 | Un1x | it was working a minute ago, but after that it stopped working... |
04:54.39 | coppice | a bigger display really pushes up the BOM. It could go as high as $20 :-) |
04:55.43 | Un1x | weird if i dail long distance works perfectly |
04:55.52 | Un1x | but anything within North America and it hangs up |
04:56.26 | symlink | check your dialplan logic... it's probably not matching any extension in the context it is searching so it gives up |
04:57.04 | Un1x | symlink would you like me to paste my extensions.conf on pastebin? |
04:57.19 | symlink | you can, but it's 2AM here and I'm slightly tired |
04:57.44 | *** join/#asterisk Azrael (n=Azrael@orion.negativeblue.com) |
04:59.24 | Un1x | http://pastebin.ca/104733 |
04:59.28 | Un1x | symlink; http://pastebin.ca/104733 |
05:00.04 | Clausian | i have asterisk registered on fwd, and a softphone registered on another account. whenever i dial asterisk's fwd number, the softphone tells me its busy. why is asterisk doing this? |
05:00.06 | symlink | and what are you dialing? and that's really insecure... someone could make calls out your box right now |
05:00.25 | Un1x | symlink how, so and how can i secure it? |
05:00.35 | symlink | you really need to learn how the dialplan works |
05:00.42 | Un1x | im new to this man |
05:00.52 | Un1x | ive gotten pretty far knowing nothing less then 5 hours ago |
05:00.54 | hads|home | ~thebook |
05:00.57 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
05:00.57 | symlink | okay, but there's lots of docs out there - and a book |
05:00.57 | Un1x | and getting this runnign and stuff |
05:01.02 | Un1x | yea i got the book with my card |
05:01.08 | Un1x | i read it all twice, it dont explain too much |
05:01.13 | Un1x | so now im going to read more docs |
05:01.19 | Un1x | try to fix it if i can and then go sleep |
05:01.37 | symlink | so on IAX2 there's a user entry called guest, with no password... it goes to the context default |
05:01.49 | symlink | if someone were to use that entry on your box right now, they could call any long distance or international number |
05:02.34 | Un1x | i see |
05:03.13 | Un1x | ok well symlink wouldn't it be better to tell me how to temporarioly disable it or secure it rather then tell everyone how to get into my box lmao |
05:03.46 | symlink | get rid of the entry in iax.conf for the guest user, or split up your contexts, or rename the default context |
05:03.52 | benjk | just kidding |
05:04.20 | Un1x | symlink; just comment this |
05:04.21 | Un1x | [guest] |
05:04.21 | Un1x | type=user |
05:04.21 | Un1x | context=default |
05:04.21 | Un1x | callerid="Guest IAX User" |
05:04.46 | symlink | I really just suggest learning about contexts, and splitting it up and getting it out of your default context |
05:05.11 | symlink | having your outbound dialing in default is just... not good |
05:05.20 | Un1x | well i will learn more but i doubt i'll be able to learn all and go sleep in the next hour it's already 1 am here, so please help me out, and just tell me can i just comment that out, and feel safe and go sleep :P? |
05:05.34 | benjk | deafult context should be like ... |
05:05.35 | benjk | [default] |
05:05.35 | benjk | ; |
05:05.36 | benjk | ; do not accept any calls to s@default |
05:05.36 | benjk | exten => s,1,NoOp(incoming connection attempt from ${CALLERID} to s@default) |
05:05.36 | benjk | exten => s,2,SetVar(PRI_CAUSE=21) ; Call reject |
05:05.37 | benjk | exten => s,3,Hangup |
05:05.45 | hads|home | Pull the plug and you will be safe |
05:05.45 | symlink | just don't have Asterisk running... |
05:05.51 | Un1x | :| |
05:05.56 | Un1x | what a great idea thanks |
05:06.11 | symlink | learn this stuff in the morning, try to read about dialplans again and ask logical questions |
05:06.26 | benjk | PRI_CAUSE is only there if you use a PRI (or BRIstuff) |
05:07.04 | benjk | in any event, you want to hangup in default, but you also want a record of the attempt, thus the NoOp() |
05:07.10 | Un1x | heh i commented out iax.conf :p |
05:07.14 | *** join/#asterisk oej (n=oej@65.197.203.67) |
05:07.36 | symlink | SIP has default as the default context too... |
05:08.02 | Clausian | Jan 1 13:03:17 NOTICE[5427]: chan_sip.c:9683 handle_response_invite: Failed to authenticate on INVITE to '"unknown" <sip:789140@fwd.pulver.com>;tag=as6b79c6fe |
05:08.05 | Clausian | why it do this? |
05:08.07 | benjk | and mgcp and skinny and h323 |
05:08.18 | [TK]D-Fender | ok, I'm fried.... later all... |
05:08.27 | symlink | fried chicken! |
05:09.13 | benjk | Clausian, because you "Failed to authenticate on INVITE" |
05:09.30 | benjk | your password is wrong, or not there |
05:09.32 | *** join/#asterisk sumasuma (n=sumase@cm222.omega183.maxonline.com.sg) |
05:09.34 | Clausian | what does that mean though? all the passwords etc are right |
05:09.43 | Clausian | asterisk is registered with it succesfully :X |
05:09.45 | benjk | obviously not |
05:09.51 | sumasuma | which software is best to use with text to voice conversion? |
05:09.52 | benjk | that's INVITE |
05:09.55 | benjk | not REGISTER |
05:09.57 | sumasuma | along with asterisk |
05:10.03 | benjk | you may have the right password for REGISTER |
05:10.07 | benjk | but not for INVITEs |
05:16.17 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
05:16.17 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
05:20.02 | dlynes_laptop | Clausian, because your register => ... line is correct and your [sipcontext] username=... ; secret=... is not correct |
05:21.14 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
05:24.06 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
05:24.50 | Clausian | yes i fixed it |
05:25.58 | Clausian | but now when i call my FWD number that asterisk is registered under on my softphone, it says 'Call Rejected: 486 Busy here' |
05:34.32 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
05:42.26 | *** join/#asterisk Clausian (i=reginald@203-206-65-20.dyn.iinet.net.au) |
05:42.34 | Clausian | how can i tell asterisk to use a nat proxy? |
05:48.34 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
05:49.07 | Kerry_G | are there any tools available to monitor trunks and alert if one goes down? |
05:49.26 | sevard | TOOORRNANDDOOO |
05:49.42 | sevard | COMEON FUCKAR I GOT A LIL CAPTIN IN ME |
05:54.54 | *** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
05:57.38 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
06:01.28 | *** join/#asterisk dlynes_laptop (n=dlynes@24.83.215.16) |
06:13.10 | *** join/#asterisk s0lid (n=jlq@210.213.245.59) |
06:13.47 | *** part/#asterisk Azrael (n=Azrael@orion.negativeblue.com) |
06:33.37 | L|NUX | any one used chan_jabber application ? |
06:35.04 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
06:38.27 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
06:44.01 | *** join/#asterisk RaHaiL (n=rahail1@209-19-88-238.detroit.mi.D-Conn.net) |
06:44.06 | RaHaiL | hi there |
06:44.17 | RaHaiL | is there any one here can make me interface |
06:45.47 | sevard | there is an interface |
06:46.09 | RaHaiL | for billing purpose and usage |
06:46.18 | sevard | i can make you interface |
06:46.22 | sevard | with a large rock |
06:46.27 | RaHaiL | :) |
06:46.38 | sevard | for your issue |
06:46.40 | sevard | check this out |
06:46.43 | sevard | google perl |
06:46.48 | sevard | that'll give you the tool you need |
06:46.50 | sevard | plus |
06:46.55 | RaHaiL | I am not a coder |
06:47.00 | RaHaiL | dont know nothing |
06:47.03 | sevard | invent a system to take alcohol out of one's body |
06:47.14 | sevard | you suck at english, that's for sure |
06:47.26 | RaHaiL | :) nop suck at gramer |
06:47.29 | RaHaiL | you got it wrong |
06:47.31 | RaHaiL | :) |
06:47.32 | sevard | and spelling. |
06:47.35 | RaHaiL | yeah |
06:47.38 | RaHaiL | you got it right |
06:47.39 | sevard | and sentence syntax |
06:47.40 | sevard | thus |
06:47.44 | RaHaiL | yeap |
06:47.44 | sevard | english |
06:48.04 | sevard | i'm drunk and i can still tell, dude that's bad. |
06:48.12 | sevard | e |
06:48.21 | RaHaiL | hehe I know I am trying my best to get rid of this bad habbit |
06:48.30 | sevard | m-w.com |
06:48.35 | sevard | spellcheck.com |
06:49.20 | RaHaiL | eh |
06:49.30 | RaHaiL | do you think you can make me one interface |
06:50.55 | L|NUX | can some one help me with asterisk + jabber |
06:50.57 | sevard | for |
06:51.10 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.141) |
06:51.14 | sevard | jabber is homogay |
06:51.56 | RaHaiL | heheh |
06:51.58 | L|NUX | sevard: hummmm |
06:52.35 | sevard | RaHaiL: i'm not going to build you an 'interface' but i'll help you build one yourself |
06:52.45 | sevard | you ought to check out prebuilt bulling platforms though |
06:52.54 | sevard | lots listed on voip-info.org |
06:53.03 | L|NUX | RaHaiL : check this out http://www.voiceone.it |
06:53.55 | sevard | hahahaha |
06:53.56 | sevard | http://www.photosbydolph.com/SouthernModels/NerdFinalWebDCLogo.jpg' |
06:54.02 | sevard | http://www.photosbydolph.com/SouthernModels/NerdFinalWebDCLogo.jpg |
06:54.44 | cy3o3 | I have an inquery. What is the best reccommended (as far as cheap and CID spoofing goes) voip provider? |
06:56.21 | RaHaiL | no seriously |
06:57.07 | *** join/#asterisk tengulre (n=tengulre@219.144.140.10) |
06:57.25 | tengulre | Hi,all |
06:58.01 | RaHaiL | L|NUX |
06:58.39 | tengulre | anybody know why not registry when I running commmand on CLI> iax2 show registry? |
06:59.27 | tengulre | I have dual asterisk and distruble two defferent place. |
07:00.32 | sevard | cy3o3: most voip providers will allow you to pass whatever CID you send down the line, not CNAM, or course, shellshark, teliax... whatever. |
07:01.10 | cy3o3 | Just kind of looking for reccommendations for a good company to go with .. pricing, support, uptime, etc.. |
07:01.16 | cy3o3 | what do you use sevard? |
07:01.18 | sevard | i gave you two. |
07:01.33 | cy3o3 | Okay, well I have a huge list. That's why I am torn between them all :/ |
07:01.56 | cy3o3 | I'll peep those two.. thx sevard |
07:01.57 | sevard | shellshark is pretty good. teliax is okay but more expensive |
07:02.04 | cy3o3 | coo |
07:02.07 | tengulre | anybody can help me? pls! |
07:02.27 | RaHaiL | trt telasip |
07:02.35 | RaHaiL | try* |
07:03.16 | tengulre | How can i connect two asterisks with IAX2 protocol? |
07:05.20 | Juggie | rtfm |
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07:07.15 | RaHaiL | <PROTECTED> |
07:07.24 | RaHaiL | Any php coder here |
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07:16.11 | cy3o3 | aight went with shellshark I guess |
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07:24.39 | tainted_ | core dump core dump core dump YAY! |
07:25.04 | RaHaiL | if i know how |
07:25.16 | tainted_ | just make calls in asterisk |
07:27.24 | tengulre | anybody know how to connect two asteirsk? |
07:28.36 | RaHaiL | tainted_ |
07:28.45 | RaHaiL | do you want help me with that small project |
07:31.48 | RaHaiL | i need some one help to get a interface for user where they can login and see there usage |
07:32.09 | docelmo | hehe |
07:32.23 | docelmo | thats simple enoguh if your MySQL CDR Logging |
07:32.47 | RaHaiL | I am using * home |
07:32.51 | docelmo | ACK! |
07:32.54 | docelmo | nevermind |
07:32.56 | docelmo | your own your own |
07:33.02 | RaHaiL | oh man |
07:33.18 | docelmo | check #freepbx |
07:33.52 | pnlarsson | tengulre: http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
07:34.22 | tengulre | pnlarsson, thanks for answer! :) |
07:36.27 | RaHaiL | eh no on |
07:36.33 | RaHaiL | anway i guess no luck for me |
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07:51.52 | tengulre | pnlarsson: I can not registriy the iax2 to remote server, do u know why? |
07:53.51 | tengulre | I got : IAX2/219.233.118.37:4569-3 is circuit-busy |
07:54.28 | tengulre | anybod know why? |
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08:03.53 | L|NUX | any one used chan_jabber application ? |
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09:12.22 | L|NUX | jbot : !!! |
09:12.23 | jbot | \"Multiple exclamation marks,\" he went on, shaking his head, \"are a sure sign of a diseased mind.\" - Terry Pratchett, Eric |
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09:19.43 | L|NUX | any one worked on asterisk + gtalk ? |
09:20.02 | Qwell | L|NUX: mog |
09:20.12 | Qwell | he is teh man, when it comes to jabber |
09:20.25 | *** join/#asterisk hack1 (i=1076@203.199.110.93) |
09:20.26 | L|NUX | Qwell : i have little issue |
09:20.31 | *** part/#asterisk hack1 (i=1076@203.199.110.93) |
09:20.33 | L|NUX | might be some one can help me out :) |
09:21.19 | *** join/#asterisk SanketMedhi (n=sanket@202.63.175.78) |
09:21.35 | L|NUX | Qwell : mog is at home :) |
09:21.44 | Qwell | sleeping, hopefully |
09:21.45 | L|NUX | Qwell : might be he will not come on from home :) |
09:21.47 | L|NUX | yupz |
09:22.00 | Qwell | well, what's the problem, specifically? |
09:22.06 | docelmo | say qwell got a sec? |
09:22.22 | L|NUX | well i am using svn |
09:22.31 | Qwell | docelmo: sure |
09:22.50 | L|NUX | i have provided user credentials for my gtalk user |
09:22.59 | L|NUX | but when i try to call its not working |
09:23.36 | Qwell | L|NUX: yeah, you'll wanna ask mog tomorrow |
09:24.33 | docelmo | ok.. this is a dev q.. When a call is setup in asterisk its in the dialplan. Whats the easiest way to find out the channel name? For instance.. I need to know the channel name for something I am doing inside of chan_sip. It keeps kicking pbx_helper_setvar to global when I need it set to the specific channel |
09:24.37 | docelmo | Any ideas? |
09:25.02 | Qwell | It's like 2:30am |
09:25.11 | docelmo | its 5:30 here.. I know |
09:25.14 | docelmo | tell me bout it. |
09:25.16 | Qwell | there is a var though for channel name |
09:25.21 | Qwell | ${CHANNEL}? |
09:25.24 | Strom_C | docelmo: you wave the appropriate dead chicken |
09:25.37 | Qwell | no, CHANNEL is a var...umm |
09:25.45 | Qwell | channelvariables.txt |
09:25.54 | Qwell | or doc/README.variables on 1.2 |
09:26.14 | Qwell | I'm just gonna give up and go to bed |
09:27.02 | docelmo | well no.. I mean for instance.. When I setup a struct ast_channel *chan; How when I set that can I manually push the current A leg of the channel into it? |
09:27.28 | L|NUX | Qwell : will ask him for sure |
09:27.32 | SanketMedhi | how do I check if my Mysql Realtime is working? When I use asterisk -vvvvvvvvvvvvdddddddddddd I see that MySql is registered properly, but the SIP phone gets an authentication failure. I have used this page http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip for setting up SIP Realtime |
09:27.37 | Qwell | docelmo: ...tomorrow |
09:27.53 | L|NUX | Qwell : i have only find two people who have worked on it one is developer which is mog and one another person cparty |
09:27.53 | docelmo | :~( |
09:27.54 | L|NUX | :) |
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09:29.19 | SanketMedhi | anyone? |
09:29.58 | Assid | SanketMedhi: have oyu added the sip credentials to the database? |
09:30.13 | SanketMedhi | credentials? |
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09:30.42 | SanketMedhi | I have added one record for the phone I have |
09:31.36 | Assid | okay and what does verbose show with respects to mysql |
09:34.08 | Qwell | bed |
09:35.27 | Assid | nini Qwell |
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11:00.32 | Clausian | how would i go about being able to listen in on calls being made through my asterisk box? |
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11:03.05 | svenna_ | hi all, im living in germany and am using chan_capi as my outgoing device. when someone rings me, for him it sounds like calling england - i cant remember where to configure the tone. does someone? |
11:04.13 | benjk_ | that's amazing |
11:05.09 | jhiver | indications.conf |
11:06.07 | benjk_ | you mean you are speaking German and he's getting English? |
11:07.41 | Nugget | heh |
11:10.01 | Clausian | asterisk supports realtime babelfish now? |
11:12.09 | benjk_ | yeah, sounds like it |
11:12.45 | benjk_ | maybe its a new feature of capi though |
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11:18.40 | svenna_ | naaaaa you know what i mean :-) |
11:23.11 | svenna_ | it was indications.conf:- thx |
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12:05.19 | e-ddie | nå ja, eg gidde'sje sitta å nørda her |
12:05.23 | e-ddie | heheh |
12:05.26 | e-ddie | wrong channel :D |
12:12.07 | *** join/#asterisk potsboy (n=chrisg@dsl-145-210-106.telkomadsl.co.za) |
12:13.56 | potsboy | goodday all, have started playing with *.. what is the alarmreciever.conf used for?? precisous little docs in this regard |
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12:49.23 | vlt | Hello. How can I activate logging to /var/log/asterisk/event-log? This file is empty now. |
12:50.04 | potsboy | vlt: waht do you want to log there? |
12:50.34 | CyberMad | sorry for OOT: how many cent is 1 euro? |
12:50.42 | CyberMad | is that 100 cents? |
12:51.27 | hypnox | lol, yeah |
12:51.30 | hypnox | hence cent |
12:52.20 | CyberMad | ^^ thanks.. |
12:54.39 | vlt | potsboy: INcoming/outgoing calls, numbers, everything ;-) |
12:55.14 | potsboy | rather have a look at /var/log/asterisk/cdr-csv/ ..is that not what you want? |
12:56.27 | vlt | potsboy: Ehem, ok. Yes, that's what I was looking for ... ;-) |
12:57.18 | potsboy | k, may want to check /etc/asterisk/cdr_custom to change the format :) |
12:58.58 | vlt | Thank you. |
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13:12.49 | *** join/#asterisk crocz (n=crocco@cc961619-a.groni1.gr.home.nl) |
13:12.58 | crocz | Hello everyone, |
13:14.42 | crocz | I am having a little trouble with Asterisk. Well it all works, calling and recieving calls to and from voip-voip / voip-phone / phone-voip. But there is a little problem. when I am calling to another user (coip to voip) that person cannot hear me, whilst I can hear them very well. |
13:15.12 | crocz | At first I thaught it was my firewall, but when I call from voip to phone I can talk without any problems. |
13:15.29 | pnlarsson | sip? |
13:16.06 | crocz | Anyone has a clue what the reason for this might be? PS: If other people are calling from other places(location / other networks ..) it simply works even for voip to voip |
13:16.07 | crocz | Yes |
13:17.09 | pnlarsson | the other user is connected to your *? |
13:17.09 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com) |
13:17.16 | pnlarsson | is he behind nat? |
13:17.39 | crocz | Yes |
13:18.13 | pnlarsson | yes on both? |
13:18.33 | pnlarsson | Are your * behind nat? |
13:19.37 | crocz | No |
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13:20.19 | pnlarsson | it's a nat prob, go and check the voip-info section for nat. |
13:20.22 | crocz | The only difference between me and the other users is that they are forwarding ports 5000 to 5100 to their machine |
13:20.44 | potsboy | rtp runns from 10000 -> 20000 by default |
13:21.06 | potsboy | if he has a ast box set rtp tp 500 -> 5100 |
13:21.51 | crocz | potsboy, the documentation for "ekiga" said to forward that range (5000 to 5100) |
13:22.56 | pnlarsson | ekiga? |
13:23.23 | pnlarsson | potsboy: Thats incomming to * |
13:24.00 | potsboy | ekiga = ubuntu |
13:24.01 | pnlarsson | Do you have nat=yes for this user? |
13:24.40 | potsboy | i would run a tcpdump on the asterisk side you will proly notice their is no rtp comming from ekiga |
13:24.58 | pnlarsson | Aha, so the other user is running * and he is forwarding 5000-5100, has he changed the setting in * to reflect that? |
13:25.02 | potsboy | brb |
13:26.43 | crocz | pnlarsson, Hmmm let me check mate |
13:27.09 | crocz | pnlarsson, I do. |
13:29.52 | potsboy | crocz: change the rtp range to 5000 -> 5100 on yourside they both have to match up as per sip invites, had the same problem with a nasty cisco |
13:31.42 | crocz | potsboy, The problem here is that I cannot change this place's router's settings. But what's really troubling me is, when I call phones, the other party CAN hear me. Whilst when I call another SIP id, they cannot |
13:32.54 | crocz | Furthermore, I have tried NOT to forward that range at a friend's, and not being a DMZ. He still was able to make calls, hear people and others could hear him. It's not really making sense to me :( |
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13:33.36 | pnlarsson | if you only have one device behind that nat, and set nat=yes it's normally just works... |
13:34.27 | potsboy | yep .. natting issue..sip sucks iax rules :) |
13:34.59 | pnlarsson | But if the other user is running *, why not use iax instead of sip? |
13:35.52 | potsboy | crocz: the best thing is to make a call and run either tethereal or tcpdump and send it to pastebin else we are just guessing |
13:38.57 | crocz | potsboy, hold on. |
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14:04.35 | *** join/#asterisk pb1dft (n=pb1dft@pb1unx.com) |
14:05.19 | n9urk | I was poking around the openpbx site. Is there anything significant going on in openpbx? Has anyone used it before? (I do know it is/was a fork of * but that is about it) |
14:05.46 | eKo1 | I've remained faithfull to *. |
14:06.55 | crocz | Ok, now THIS is weird! I found out that if I am the caller, the other person will be able to hear me. But if I am being called, that very same person won't. |
14:07.08 | n9urk | eKo1: I don't see any reason to switch, I am just curious as to if anything is going on. I see about 5 os projects a week where it looks like they are going to save the world and end global warming but there is nothing going on in reality - Someone had/has an idea and knows how to put a site together and then has no follow through |
14:07.43 | crocz | n9urk, I am using FreePBX for administrating Asterisk |
14:09.27 | n9urk | crocz: cool, I am looking at the site now. Is it an additional layer to asterisk or does it include some version of asterisk as well? |
14:09.48 | crocz | in my case, additional. |
14:10.31 | n9urk | crocz: Ok I see "launched the freePBX (formerly Asterisk Management Portal) " I know of AMP |
14:10.40 | n9urk | I didnt know the name changed |
14:11.44 | crocz | pnlarsson, potsboy, did you see the last thing I said about my problem? |
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14:19.24 | potsboy | crocz i stand by with the previous rtp range issue.. the fact that you dont have control over the router doesnt help, do a "tethereal -v port 5060 and host <ip of remote host>" and you can compare the "working" call against the other this should sort it out |
14:20.10 | potsboy | another thought maybe that the remote router does not support udp stream's that do not origante from the internal lan on the other side |
14:20.19 | potsboy | l8r all |
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14:48.42 | Cherebrum | Anyone do load balancing with Asterisk? |
14:49.52 | Cherebrum | I have two geographicly located asterisk servers and a SRV record designed to allow the UAs to failover to ASterisk #2 in the case taht #1 fails.. however.. I need a way to make Asterisk #2 register to the sip proxy and start accepting calls on the trunk side when Asterisk #1 fails |
14:51.28 | Cherebrum | I've also got to keep the asterisk configs synced up so I think I'm just going to have to run rsync in a cron job and wost case they loose some voicemail |
14:53.06 | pnlarsson | You need to sense that *1 goes down... Is a ping enough? |
14:53.43 | Cherebrum | well... I was thinking maybe I could use my nagios server and have it execute a script as an escillation procedure |
14:53.59 | Cherebrum | I was hoping there was a more elegant way |
14:54.00 | Cherebrum | :p |
14:54.37 | pnlarsson | nagios does a good job :) |
14:54.37 | crocz | Hmmm |
14:54.41 | crocz | may be heartbeat |
14:54.50 | Cherebrum | over a wan? |
14:55.03 | Cherebrum | one is in Florida and one in Illinois |
14:55.35 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
14:56.38 | Cherebrum | Sales guy sold it and just assumed we could make it work. ;) |
14:57.56 | Cherebrum | oh! I know what I can do... |
14:58.12 | Cherebrum | if my OpenSER proxy can do SRV lookup I can just point it to the SRV record |
14:58.27 | Cherebrum | and it will work just like the phones do |
14:58.44 | Cherebrum | it will send the call to the asterisk that ACKs the call |
15:00.19 | eKo1 | that seems like a good idea |
15:00.31 | Cherebrum | I just wont use registrations |
15:01.25 | Cherebrum | http://www.voipuser.org/forum_topic_3624.html |
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15:51.52 | mrtass111222 | hi |
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15:52.32 | mrtass111222 | what's the best calling card platform for use with asterisk |
15:53.14 | eKo1 | asterisk is the calling card platform |
15:53.18 | eKo1 | so that question makes no sense |
15:53.48 | mrtass111222 | ok which billing is the best to use with asterisk |
15:53.58 | crocz | He probably wants * to function as a gateway for some other voip provider |
15:54.14 | crocz | oh... I was wrong :) |
15:54.31 | mrtass111222 | i need to build a calling cards platform so far i have installed asterisk on fedora |
15:54.37 | mrtass111222 | what's next |
15:55.28 | crocz | http://www.voip-info.org/tiki-index.php?page=Asterisk+billing |
15:55.32 | eKo1 | mrtass111222: search voip-info.org for calling cards and start reading/investigating |
15:55.39 | crocz | first google result |
15:56.21 | tzafrir_laptop | mrtass111222, I would seriously consider the support lifetime of Fedora |
15:57.19 | mrtass111222 | can anyone knows how these stuff works i'll pay for any help |
15:58.47 | folder | Can anyone see why this call failed? I tried again a few seconds later and it worked. http://pastebin.ca/105335 |
16:00.12 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
16:00.24 | folder | here's the sucessful call: http://pastebin.ca/105337 |
16:00.32 | eKo1 | mrtass111222: contact your nearest * consultant near you. |
16:10.37 | mrtass111222 | can anyone help me configure my asterisk ? |
16:11.53 | *** join/#asterisk crparr (n=crparr@213.129.243.185) |
16:13.12 | crparr | Hi! I'm really new to asterisk and have one question: is it possible to access one phone and one fax using one sipura/linksys adapter with extensions? |
16:13.27 | *** join/#asterisk EvilDeshi (i=evildesh@oxford-bb-occam3-ws-100.dsl.maqs.net) |
16:13.39 | eKo1 | Define 'access'. |
16:14.14 | crparr | eg. phone: ext 10, fax ext 20 |
16:14.37 | crparr | but both connectzed to one sipura adapter |
16:15.15 | crparr | what other hardware is needed when using the adapter? |
16:15.18 | crparr | Is it possible to run asterisk behind a firewall? |
16:15.28 | mrtass111222 | u need configure sipura channels |
16:15.48 | mrtass111222 | how many lines the sipura support |
16:15.52 | mrtass111222 | 2 ? |
16:15.55 | crparr | 2 |
16:16.09 | crparr | at least it hass 2 phone connectors |
16:16.24 | mrtass111222 | go to the web interface for the sipura |
16:16.35 | mrtass111222 | and configure dialpeers there |
16:16.50 | mrtass111222 | 10 for line 1 |
16:16.54 | mrtass111222 | 20 for line 2 |
16:17.02 | crparr | thanks. |
16:17.18 | crparr | but - I'm only gathering information at the moment. |
16:17.34 | mrtass111222 | i worked with sipura |
16:17.37 | crparr | what hardware is needed in the pc? one lan card? |
16:18.10 | mrtass111222 | u need connect the sipura to the network |
16:18.18 | crparr | ok. |
16:18.24 | crparr | thats no prob then. |
16:18.55 | crparr | one last question: Is it possible to run an asterisk pbx behind a firewall? |
16:19.04 | mrtass111222 | there should be connectivity between the asterisk and sipura and u need register te sipura onthe asterisk |
16:19.17 | mrtass111222 | i dont know buddy abt asterisk |
16:19.22 | mrtass111222 | i'm trying get help here |
16:19.27 | crparr | oh sory |
16:20.58 | tzafrir_laptop | mrtass111222, do you have sip users defined for those? |
16:21.14 | mrtass111222 | xlite1/xlite1 |
16:21.46 | tzafrir_laptop | Can you register to them with any other sip client? |
16:21.48 | mrtass111222 | but i can't see the sip port 5060 opened when i scan the ASTERISK SErver |
16:21.56 | *** join/#asterisk potsboy (n=chrisg@dsl-145-210-106.telkomadsl.co.za) |
16:22.09 | tzafrir_laptop | mrtass111222, port 5060 UDP, not TCP |
16:22.29 | tzafrir_laptop | 'netstat -lnup' on the Asterisk server |
16:22.41 | tzafrir_laptop | should show you listening UDP ports |
16:23.30 | mrtass111222 | not showing 5060 |
16:23.42 | tzafrir_laptop | Is asterisk runinng? Can you connect to it with 'asterisk -r' from the asterisk host? |
16:23.53 | mrtass111222 | in sip.conf i set bind address to 0.0.0.0 as they say |
16:24.03 | mrtass111222 | and restarted the service still nothing |
16:24.20 | mrtass111222 | yes i can't connect to it |
16:24.27 | mrtass111222 | with asterisk -r |
16:24.29 | potsboy | hey all, what is the purpose of dnsmgr.conf? |
16:24.30 | tzafrir_laptop | can or can't? |
16:24.33 | *** join/#asterisk oej (n=oej@65.197.203.67) |
16:24.44 | mrtass111222 | i can sorry |
16:25.57 | tzafrir_laptop | next: let's see of the sip module is loaded. from the asterisk CLI (asterisk -r) write 'sip' and press TAB twice. Does it complete some commands? |
16:27.22 | mrtass111222 | IS-0665*CLI> sip |
16:27.23 | mrtass111222 | debug history no notify prune reload show |
16:27.32 | mrtass111222 | it does |
16:28.02 | tzafrir_laptop | So chan_sip.so is loaded. |
16:28.20 | mrtass111222 | maybe it's firewall issue on fedora |
16:28.27 | mrtass111222 | how can i turn off the firewall |
16:28.42 | tzafrir_laptop | There is no process that listens on UDP port 5060? (from the output of netstat) |
16:28.46 | *** join/#asterisk SanketMedhi (n=sanket@221.135.148.49) |
16:28.57 | mrtass111222 | no |
16:29.03 | tzafrir_laptop | mrtass111222, netstat's output ignores the firewall settings. |
16:29.07 | mrtass111222 | i also tried nmap localhost |
16:29.21 | mrtass111222 | no 5060 |
16:29.37 | tzafrir_laptop | nmap from localhost laso ignores most firewall settings as you normally won't filter anythin from localhost |
16:29.54 | tzafrir_laptop | try these: |
16:29.59 | tzafrir_laptop | set verbose 5 |
16:30.06 | tzafrir_laptop | sip reload |
16:30.13 | *** join/#asterisk s0lid (n=jlq@203.177.12.98) |
16:30.34 | tzafrir_laptop | (should that attempt to re-bind the UDP port?) |
16:31.01 | mrtass111222 | i can see 5060 with netstat |
16:31.03 | mrtass111222 | but not with nmap |
16:31.19 | potsboy | telnet? |
16:31.19 | Nugget | telnet is eeeeeeevil! |
16:31.31 | potsboy | mmkay :( |
16:31.37 | eKo1 | telnet telnet telnet! |
16:31.56 | tzafrir_laptop | mrtass111222, this is now more of a firewall issue... |
16:32.08 | mrtass111222 | yes i beleive so |
16:32.41 | tzafrir_laptop | I'm not familiar with FEdora's firewall mechanism. Any Fedora user here? Which version of Fedora is it, BTW? |
16:33.03 | mrtass111222 | core 5 |
16:33.10 | eKo1 | Just enter 'service iptables stop' in your shell |
16:34.06 | mrtass111222 | i did |
16:36.27 | potsboy | mrtass explain your setup, is the system remote /local router etc |
16:37.56 | *** part/#asterisk SanketMedhi (n=sanket@221.135.148.49) |
16:37.56 | eKo1 | mrtass111222: so the firewall is off then |
16:38.10 | mrtass111222 | yes i can register on the asterisk now |
16:38.15 | mrtass111222 | thx |
16:39.12 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.161) |
16:39.42 | mrtass111222 | 'service iptables stop' will stop the firewall but will it load again on the next reboot ? |
16:40.46 | potsboy | chkconfig --level 2345 iptables off.. will stop the firewall its best to edit /etc/sysconfig/iptables and add 5060 in the input chain |
16:41.29 | *** join/#asterisk pengyong (n=lala@218.93.158.200) |
16:42.02 | eKo1 | mrtass111222: probably |
16:47.23 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.74.28) |
16:47.57 | tzafrir_laptop | "5060" there refers to a UDP port or a TCP port? |
16:49.55 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.217) |
16:50.16 | ManxPower | UDP, I assume you would specify the protocol |
16:50.24 | ManxPower | don't forget to allow the RTP ports as well. |
16:51.33 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
16:51.54 | asterisk-dud | how can i get asterisk to alert the user that they have a message in thier inbox? |
16:52.14 | *** join/#asterisk apardo (n=apardo@eu85-87-2-82.clientes.euskaltel.es) |
16:53.26 | *** join/#asterisk tamp4x (n=tampon@ip68-110-205-128.ri.ri.cox.net) |
16:53.31 | potsboy | dud, send a message to their email |
16:53.48 | tamp4x | is there away to use array variables? |
16:53.51 | tamp4x | a way |
16:55.12 | tamp4x | hmm i guees i can use concat then parse that |
16:57.27 | JunK-Y | caster*CLI> show function ARRAY |
16:59.20 | ManxPower | Is there actually an ARRAY function now? |
16:59.39 | MikeJ | *CLI> show function ARRAY |
16:59.40 | MikeJ | :P |
17:00.29 | ManxPower | I would have to SSH over a connection with 900ms - 2000ms latency to connect to an Asterisk box. Not worth the pain. |
17:01.05 | clyrrad | Hey MikeJ - did you see my solution I PM'ed you yesterday? |
17:02.55 | MikeJ | clyrrad, no |
17:03.17 | clyrrad | remember we were pondering over the Local or LOCAL |
17:03.24 | clyrrad | to see if it was case sensitive? |
17:04.07 | MikeJ | I remeber the discussion.. but you said it wasn't working.. |
17:04.15 | clyrrad | the problem was that in queues.conf your member = Local/ext@context has to bee all one string - you have have a "-" or "_" in the context name |
17:04.44 | clyrrad | I had Local/ext@context-likethis .... and you cant do that |
17:04.57 | MikeJ | oh.. you had a space... hehe.. yeah.. that won't work |
17:05.02 | clyrrad | you can use "-" or "_" in extensions.conf - but not with local in queues.conf :) |
17:05.28 | clyrrad | hahah yea.... that was driving me nuts |
17:05.28 | clyrrad | LOL |
17:05.35 | MikeJ | don't use spaces in context names... it shouldn't even allow that.. not sure why it does. |
17:06.15 | clyrrad | no not spaces..... I was using a "-" |
17:06.19 | shido6 | [Space The_finalFront ier] |
17:06.27 | clyrrad | like nameed-context |
17:06.31 | clyrrad | and thats does not work..... |
17:06.37 | MikeJ | you can't use them at all ? |
17:06.41 | clyrrad | Nope |
17:06.42 | MikeJ | that's even more broken |
17:06.46 | clyrrad | in the dial plan you can |
17:06.50 | clyrrad | but not in queues.conf |
17:06.54 | MikeJ | sigh.. oh well.. |
17:07.03 | clyrrad | hahaha took a bit of playing with to find that out |
17:07.34 | clyrrad | Do you know of a Dial flag or option that will only allow Dial to make outgoing callsed based on what patterns are set in outbound? |
17:07.36 | MikeJ | so, did you fix it and submit your patch to the bug monster |
17:07.49 | clyrrad | No - I was not sure if it was a bug.... |
17:07.53 | clyrrad | but it seems that way... |
17:08.00 | MikeJ | seems broken to me |
17:08.11 | clyrrad | yea - its not consistant thats forsure |
17:08.17 | MikeJ | patterns set in outbound? |
17:08.24 | clyrrad | yea.... |
17:08.43 | MikeJ | as in, ... what does that say.. that makes no sense to me |
17:08.45 | clyrrad | but I want to use a Dial statment in a macro - but only want to allow dial patterns as defined in outbound |
17:08.58 | MikeJ | what is outbound |
17:09.37 | clyrrad | well in the dial plan I have a bunch of pattern match stamtnets that allow calls out over IAX.... and not others.... |
17:09.48 | clyrrad | to restrict international calls for example |
17:10.06 | MikeJ | I know what dial patterns are.. I still don't know what outbound is |
17:10.25 | clyrrad | im using "oubound" as a word to say what I am trying to do |
17:10.29 | clyrrad | its not a context or antying |
17:10.35 | clyrrad | its a file i made called outbound.inc |
17:10.36 | MikeJ | ummm |
17:10.39 | clyrrad | and it has all those dial patterns in it |
17:10.50 | MikeJ | so make it a context, and send those calls to that context. |
17:10.52 | clyrrad | then I just included that file in the phones context |
17:10.57 | MikeJ | and it will only work if it's in there |
17:11.16 | clyrrad | yea - thats how it works now when you dial out from a phone..... |
17:11.25 | MikeJ | so what's the problem? |
17:11.30 | clyrrad | but if you dial out though the macro I am creating it lets you dial whatever you want |
17:11.35 | clyrrad | including international... |
17:11.51 | MikeJ | no... it's based on what is in the context |
17:12.09 | MikeJ | in a macro, you are still in the calling context |
17:12.12 | clyrrad | strange.... becase it allows international to be made.... |
17:12.16 | clyrrad | yea thats what I thought |
17:12.17 | MikeJ | set your includes right |
17:12.38 | clyrrad | yep - it works everywhere just not in this macro lol |
17:12.54 | MikeJ | ummmmm |
17:13.18 | MikeJ | I have no idea what your saying.. but the good news is... I am going to go get a bite to eat... |
17:13.22 | MikeJ | YUM! |
17:13.30 | clyrrad | hahahaha |
17:14.23 | *** join/#asterisk leejohn (n=leejohn@210.213.199.72) |
17:21.18 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
17:22.20 | *** join/#asterisk cr0n (i=d@dsl-146-248-230.telkomadsl.co.za) |
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17:23.42 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.217) |
17:23.46 | *** join/#asterisk potsboy (n=chrisg@dsl-145-210-106.telkomadsl.co.za) |
17:26.00 | *** join/#asterisk NoName|R (i=noname@c-71-232-59-52.hsd1.ma.comcast.net) |
17:27.08 | NoName|R | hi all can some on give me a hand i can get zap installed i am using centos 4.3 and Linux pbx.cantella.com 2.6.9-39.0.2.EL #1 Thu Jul 13 04:53:11 CDT 2006 x86_64 x86_64 x86_64 GNU/Linux |
17:28.02 | potsboy | whats the prob noname |
17:28.05 | NoName|R | i doesnt seem to be making /dev/zap and doest seem to be loading the zap module |
17:28.10 | leejohn | NoName|R: if you can't get zaptell install look for spinlock.h bug |
17:28.17 | leejohn | zaptel* |
17:28.32 | *** part/#asterisk rata (n=rodrigo@princed/developer/rata) |
17:28.44 | NoName|R | leejohn, i looked at that and doesnt seen to be an issue with this ver |
17:29.09 | potsboy | noname its definately in 4.3 |
17:29.11 | NoName|R | do u have to rebuild the kern after installing the zaptel? |
17:29.20 | leejohn | NoName|R: no |
17:30.38 | NoName|R | potsboy, i looked and it didnt see the type in there |
17:30.55 | potsboy | oop only saw no your running x86_64.. my bad |
17:31.04 | leejohn | NoName|R: pastebin the last output |
17:31.23 | NoName|R | but i am willing to try anything as i am not getting anywhere |
17:31.55 | NoName|R | leejohn, what output u want? |
17:32.15 | leejohn | NoName|R: the line where broke your installation |
17:32.31 | NoName|R | i am only seeing 1 line in logs with zap in them |
17:33.18 | leejohn | NoName|R: could you please be specific? can you explain more? |
17:33.23 | NoName|R | it seems to install fine doesnt through any errors just never makes /dev/zap or loads module |
17:34.04 | NoName|R | sorry for spelling and slowness i broker my elbow so i am forced to type 1 handed |
17:34.21 | rob0 | ~centosbug |
17:34.23 | jbot | centosbug is probably a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
17:34.26 | rob0 | maybe? |
17:34.52 | NoName|R | boot.log:Jul 29 12:07:33 pbx modprobe: FATAL: Module zaptel not found. |
17:34.52 | NoName|R | boot.log:Jul 29 12:07:33 pbx zaptel: Loading zaptel framework: failed |
17:35.24 | leejohn | NoName|R: no problem :) do you think it's a udev issue? |
17:35.38 | leejohn | NoName|R: modprobe -l | grep zaptel what's d output? |
17:36.16 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
17:38.46 | NoName|R | leejohn, nothing :/ |
17:39.21 | NoName|R | hmm |
17:39.22 | leejohn | NoName|R: it means that the compilation of zaptel wasn't successful |
17:39.27 | NoName|R | hold on' |
17:39.30 | leejohn | kk |
17:39.32 | leejohn | Ok |
17:40.11 | NoName|R | seems like kernel-2.6.9-39.0.2.EL and kernel-smp-devel-2.6.9-39.0.2.EL are installed |
17:40.58 | leejohn | could you try jbot suggestion? |
17:40.59 | NoName|R | but [root@pbx linux]$ uname -r |
17:40.59 | NoName|R | 2.6.9-39.0.2.EL and /usr/srn/kern is smp |
17:41.19 | leejohn | ouch |
17:41.39 | NoName|R | sould i unistall non smp and reinstall snp dev |
17:41.41 | leejohn | kernel-devel package from your current running kernel |
17:41.54 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
17:42.08 | leejohn | then try to redo the installation of zaptel |
17:42.32 | NoName|R | if that is the prob i will kill my self so simple but i missed it |
17:42.40 | leejohn | :) |
17:42.48 | *** join/#asterisk mitcheloc (n=mitchelo@c-24-23-37-212.hsd1.ca.comcast.net) |
17:43.40 | folder | Can anyone see why this call failed? I tried again a few seconds later and it worked. http://pastebin.ca/105335 |
17:43.42 | folder | here's the sucessful call: http://pastebin.ca/105337 |
17:44.12 | leejohn | folder: wait let me check |
17:44.19 | folder | okie |
17:46.34 | DaveHope | Hello all. At present, My dialplan entry for dialing normal users goes something like this: _2XX,1,Dial(${EXTEN}) However, that only allows me 99 real people. Say I want user extensions to be 200-300, would I do the following: _[2,3]XX,1,Dial(${EXTEN}) ? |
17:49.16 | *** join/#asterisk Frogdude (n=FroggerD@c-24-16-72-159.hsd1.wa.comcast.net) |
17:52.15 | leejohn | folder: what do you try to accomplish? i can't imagine the logs i'm really tired =( |
17:56.34 | folder | leejohn: calls come in from sipgate.co.uk , asterisk calls my cellphone via the sip gsm gateway on 192.168.253.200, and connects the two. It seems that the final part - the gsm bit, isn't always working. It is sometimes but not others. |
17:57.29 | leejohn | folder: are you using voiceblue? or something |
17:57.41 | folder | I have been reading the Asterisk TFOT dialplan chapters the last few nights, and I think I am ready to ditch Trixbox and D.I.M(yself) now though, which might help debugging. |
17:58.37 | folder | leejohn: similar. Portech MV-370 from http://www.portech.com.tw/eweb/MV370/mv370.htm |
17:59.31 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-50-226.cybersurf.com) |
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18:07.58 | tamp4x | if i have a variable of size 20 characters, and i do Dial(SIP/${variable:0:10}&SIP/${variable:10:20}&SIP/${variable:20:30}) will the 3rd sip chan not be dialed or will asterisk crash? |
18:08.43 | tamp4x | or will it complain |
18:09.47 | *** join/#asterisk DaveHope (n=Dave@internal.davehope.co.uk) |
18:10.52 | *** part/#asterisk leejohn (n=leejohn@210.213.199.72) |
18:12.41 | folder | DaveHope: I beleive it should actually be _[23]XX,1,Dial(${EXTEN}) |
18:12.56 | folder | DaveHope (no comma) |
18:13.20 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-181.sw.biz.rr.com) |
18:13.35 | ctooley | is there a way to make "SayDigits" interruptible? |
18:14.45 | folder | DaveHope: That would match 200 - 399. Not the 200 - 300 that you asked for though. I presume that's what you're after. |
18:15.00 | DaveHope | Hrm. Thanks. |
18:15.46 | folder | DaveHope: I'm only going off what it says in "Asterisk - The Future of Telephony". |
18:16.01 | folder | DaveHope: perhaps /the/ book is wrong ? [shrug] |
18:16.46 | folder | DaveHope: actually, that link says the same. It doesn't say to use a comma there does it? |
18:17.10 | DaveHope | Nope :) |
18:17.17 | folder | ah :) |
18:17.31 | DaveHope | I was doing what the comment said, rarher than the example of what it'd achieve :) |
18:17.32 | DaveHope | Thanks. |
18:17.37 | folder | did you confuse the comments bit? or am I barking up the wrong tree? |
18:17.38 | folder | ahh. |
18:17.42 | folder | lol |
18:18.54 | folder | brb, poo. |
18:22.22 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
18:23.00 | mrtass111222 | asterisk died with code 1 |
18:23.14 | mrtass111222 | what's wrong ??? |
18:24.20 | mrtass111222 | guys can anyone help have a proper installation of asterisk from scratch |
18:24.33 | mrtass111222 | i spent all night reading the manual and now it's dying with code 1 |
18:24.40 | hypnox | look at the log |
18:25.35 | *** join/#asterisk the-chaos5 (n=sendme@ACB3F7FB.ipt.aol.com) |
18:26.29 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
18:26.35 | the-chaos5 | where can i find an instalation documentation for the open source asterisk ? |
18:26.49 | mrtass111222 | asterisk.org |
18:27.06 | mrtass111222 | support |
18:27.37 | the-chaos5 | theres only one for the Asterisk Business Edition |
18:28.56 | hypnox | http://www.google.com/search?q=installing+asterisk |
18:30.35 | mrtass111222 | http://www.digium.com/en/supportcenter/documentation/ |
18:31.22 | the-chaos5 | jes but theres only one for the Business Edition |
18:31.51 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
18:33.12 | mrtass111222 | if u have ur system on public i'll help u set it up |
18:33.12 | mrtass111222 | but not more then initial setup :S |
18:33.53 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-106.columbus.res.rr.com) |
18:35.10 | tzafrir_laptop | mrtass111222, where did you get the message that asterisk died with code 1? |
18:35.43 | mrtass111222 | after i rebooted the server |
18:35.53 | mrtass111222 | i was trying to lunkch the asterisk service |
18:35.59 | tzafrir_laptop | How do you start asterisk? |
18:36.01 | mrtass111222 | then i start getting that message |
18:36.14 | mrtass111222 | from /etc/init.d |
18:36.17 | mrtass111222 | ./asterisk start |
18:36.28 | tzafrir_laptop | Do you run asterisk as a user or as root? |
18:36.43 | mrtass111222 | root |
18:37.07 | tzafrir_laptop | Try running it manually to see if it can start: asterisk -cvvvvvvvvvv |
18:37.21 | mrtass111222 | k |
18:37.40 | tzafrir_laptop | If it gets to 'Asterisk Ready' it is probaly OK. Exit it. |
18:38.19 | mrtass111222 | [root@IS-0665 ~]# asterisk -cvvvvvvvvvv |
18:38.20 | mrtass111222 | <PROTECTED> |
18:38.20 | mrtass111222 | <PROTECTED> |
18:38.20 | mrtass111222 | <PROTECTED> |
18:38.20 | mrtass111222 | <PROTECTED> |
18:38.27 | tzafrir_laptop | ~pb |
18:38.29 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
18:38.50 | tzafrir_laptop | too late |
18:38.53 | sevard | the internet is boring today. |
18:40.10 | potsboy | mrtass: i gues asterisk died cuase zaptel was not loaded |
18:40.16 | *** join/#asterisk mrtass111222 (n=Tassi@83.229.72.14) |
18:40.27 | potsboy | mrtass: i gues asterisk died cuase zaptel was not loaded |
18:40.31 | the-chaos5 | wb |
18:41.29 | mrtass111222 | after i installed asterisk i installed mysql |
18:42.10 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
18:43.10 | mrtass111222 | now it's calling for module "res_config_mysql.so" |
18:47.33 | *** join/#asterisk riddlebox (n=mythtv@24-171-10-102.dhcp.stls.mo.charter.com) |
18:47.48 | ph|ber | im getting this on my incoming.. Subclass: REJECT CAUSE : No such context/extension. my iax connections has context=from-pstn.. my [from-pstn] extention has exten => _XXXX,1,Goto(Sip/2000,1) |
18:47.53 | ph|ber | any ideas? |
18:48.12 | [TK]D-Fender | ph|ber : Your GOTO is bad. |
18:48.27 | ph|ber | goto is bad? |
18:48.31 | mrtass111222 | case sensitive ? |
18:48.36 | [TK]D-Fender | ph|ber : Sip/200 is NOT a context |
18:48.51 | [TK]D-Fender | ph|ber : OR and exten. |
18:48.51 | ph|ber | ahh.. hrm... k |
18:49.11 | ph|ber | exten => _XXXX,1,Goto(internal,${EXTEN},1) |
18:50.05 | [TK]D-Fender | ph|ber : Looks a lot better.... |
18:55.39 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
18:57.20 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
19:05.29 | mrtass111222 | can anybody provide direct help |
19:05.46 | mrtass111222 | i need some love... |
19:06.32 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
19:06.49 | Jason99 | does anyone know why music on hold would play in slow motion? |
19:07.10 | Jason99 | I'm using the default moh mp3 that come with asterisk |
19:08.36 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
19:08.49 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
19:09.13 | tzafrir_laptop | Jason99, there's plenty of classical music availble (that is in the public domain) |
19:09.45 | Un1x | shit man when someone dails my did, on incomng they get this bullshit that you have successfully installed asterisk |
19:09.49 | tzafrir_laptop | sineapps.com has some samples |
19:09.50 | Un1x | and my phone dont ring :S |
19:10.07 | tzafrir_laptop | mrtass111222, so did your asterisk start successfully? |
19:10.16 | Jason99 | tzafrir_laptop: thanks |
19:11.12 | tzafrir_laptop | Un1x, time for some basic RTFM on asterisk dialplan? get to know the magic s extension? |
19:11.53 | Un1x | lmao yea imma look for some docs on dailplan later |
19:11.56 | Un1x | to lazy atm :p |
19:12.21 | tzafrir_laptop | ~docs |
19:12.23 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
19:12.26 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
19:12.50 | tzafrir_laptop | voip-info.org has a useful page on dialplans setup |
19:14.19 | sp0n9e | can i add spaces in exten statements to enhance clarity? |
19:15.03 | sp0n9e | something like...exten => s, 1, Answer() |
19:17.13 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
19:18.25 | tzafrir_laptop | sp0n9e, generally: no. Only around the '=>' |
19:18.44 | *** join/#asterisk ttymachine (i=w0rmzw3r@ool-18b8840d.dyn.optonline.net) |
19:19.11 | tzafrir_laptop | Otherwise it will complain that it can't find an application called ' Answer' |
19:20.30 | sp0n9e | will it trim the whitespace around 1? |
19:21.08 | ttymachine | anyone know of a password identify addon for asterisk , a caller calls in if his passcode is correct he is accepted into the menu system , if 5 wrong pass codes and his number is blocked. |
19:21.42 | *** part/#asterisk mrtass111222 (n=Tassi@83.229.72.14) |
19:24.19 | clyrrad | ttymachine you can code that in the dial plan |
19:25.15 | ttymachine | i'm a complete n00b to this , I want to set my own system up for learning exp. |
19:27.34 | *** join/#asterisk tempest1 (n=asf@adsl-144-60-181.chs.bellsouth.net) |
19:27.55 | ttymachine | anyone have a code like thatt they can show me for example? |
19:29.53 | clyrrad | reasearch the Read application its what you need |
19:31.26 | Un1x | shit i wish |
19:31.34 | Un1x | i could get my asterisk running normaly lol |
19:40.05 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
19:40.54 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
19:41.22 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
19:43.17 | Un1x | anyone know why i get this |
19:43.18 | Un1x | http://pastebin.ca/105620 |
19:44.25 | [TK]D-Fender | Un1x : Maybe you could apstebin the defective file, and not just the errors it creates. |
19:44.41 | [TK]D-Fender | Un1x : mind you its TELLING your what ares are bad and is should sorta stand out.... |
19:44.53 | [TK]D-Fender | Un1x : I'm betting its apparent withing 2 seconds of viewing... |
19:45.32 | tzafrir_laptop | Un1x, you need 'type=peer' or 'type=friend' or 'type=user' for each section in sip.conf (except the general one) |
19:45.49 | Un1x | ok well i fixed the last 2 error's in the terminator and 100 |
19:45.51 | Un1x | but now it's this |
19:45.54 | Un1x | one sec let me pastebin |
19:46.05 | Un1x | http://pastebin.ca/105623 |
19:46.08 | Un1x | now it's that... |
19:46.22 | clyrrad | I have read a telephone number in using READ and have the value stored in a variable caled dialnumber - how do I get that variable on an outbound channel so that its pattern matched to be sure of a valid number before being set to the Dial application? |
19:46.24 | tzafrir_laptop | BTW: #asterisk-dev is not second-level support for #asterisk |
19:46.47 | Un1x | didn't say it was tzafrir... |
19:47.14 | tzafrir_laptop | well, I don't see anything suspcious there. |
19:47.46 | tzafrir_laptop | the messages from chan_zap are because a 'reload' can't change the type of channels, only their configuration |
19:48.02 | Un1x | oh okays |
19:48.09 | tzafrir_laptop | And the others (indications.conf and CDR) seem rather harmless |
19:49.08 | clyrrad | anyone know the answer to my question? |
19:50.10 | tzafrir_laptop | clyrrad, goto() a context with those patterns in its plan? |
19:50.40 | clyrrad | I will expalin the seneario.... |
19:51.38 | clyrrad | Basically what I have is a bunch of pattern match exten's that are used on all internal phones to be sure the dialed OUT number is valid before calling the Dial() application - now that works fine on all internal phones that are dialing out. The problem is if I have a macro somewhere in the dial plan (lets say for doing call forward) - it will not check that the number is an "allowed" or proper number - I just dials |
19:52.18 | clyrrad | I need a way to avoid that and make the system use the dial rules that the phones do... |
19:52.26 | clyrrad | does that make more sense now? |
19:52.30 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
19:52.43 | [TK]D-Fender | clyrrad : Use a Goto for the read. if it doesn't match the dialplan will continue with the next priorit at which point you can deal with it as you choose. |
19:53.38 | clyrrad | hrm....... |
19:53.53 | clyrrad | so GoTo(dialnumber,1) like that? |
19:59.16 | Un1x | w00t |
19:59.20 | Un1x | finaly i got this shit working lmao |
19:59.39 | Un1x | too bad dont have time for the cid spoof script :/ |
20:00.08 | Un1x | hmm imma tell my freind to call me lol |
20:00.18 | Un1x | and put a auto attendent on |
20:00.18 | Un1x | :P |
20:00.55 | *** part/#asterisk vlrk (n=vlrk@202.65.134.119) |
20:03.20 | [TK]D-Fender | clyrrad : Something like that.... |
20:08.00 | clyrrad | thanks TDK :) |
20:10.06 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
20:13.32 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-106-111.dsl.peoril.sbcglobal.net) |
20:14.00 | MACscr | hey, whats the function called for rewriting the caller id name |
20:15.04 | clyrrad | SetCallerID |
20:15.40 | Un1x | heh |
20:16.35 | MACscr | sry guys, just trying to explain it to someone that supposedly knows how to manage an asterisk system |
20:16.53 | MACscr | im trying to explain that i need caller id rewriting so i know whatcompany teh user called for |
20:18.28 | [TK]D-Fender | MACscr : the CALLERID function. SetCallerID is deprecated |
20:19.27 | Assid | hey tkd |
20:19.31 | Assid | whats cookin |
20:20.20 | MACscr | Un1x, are you getting my messages? |
20:21.20 | Un1x | yea |
20:22.32 | clyrrad | hrm.... TKD - the GoTo idea was neat but does not seem to work becase it looks for a context in the name of the telephone number that was entered - as opposed to trying to pattern match it so that it does the Dial application.... any ideas? |
20:24.35 | [TK]D-Fender | clyrrad : Yeah... you're formatting it wrong... |
20:24.48 | [TK]D-Fender | clyrrad : Pastebin what you've done so far. |
20:24.54 | clyrrad | ah... ok.... |
20:25.06 | clyrrad | here is the pattern that needs to be matched _NXXNXXXXXX |
20:25.09 | clyrrad | will pastebin the rest |
20:25.17 | MACscr | lol, why to voip providers say unlimited incoming, then put an asterisk, then you see that its really only 1500 a month |
20:25.19 | MACscr | i hate that |
20:26.15 | clyrrad | http://pastebin.ca/105659 |
20:28.47 | clyrrad | this is what the CLI Shows -> Executing Goto("SIP/2000-09650f58", "9059998888") in new stack |
20:29.06 | clyrrad | which is wrong becase it thinks the number I dialed is a context.... |
20:30.19 | clyrrad | I need a way to get 'dialnumber' out onto the outgoing channel somehow so that its pattern matched just like a regular outgoing call made from a telephone... |
20:31.44 | [TK]D-Fender | clyrrad : S,16 = bad. read the instructions for Goto..... |
20:32.06 | [TK]D-Fender | clyrrad : and no, it did not this it was a context.... read again |
20:32.40 | clyrrad | Goto(context,extension,priority) |
20:33.22 | clyrrad | what would I put as teh context? |
20:34.07 | clyrrad | exten => _NXXNXXXXXX,6,Dial(IAX2/${ACCOUNTCODE}/${EXTEN},,W) thats the line that should do the dial... |
20:34.36 | clyrrad | am I supposed to set _NXXNXXXXXX as the context? Becase if I do - what if they dont enter the number in that format? |
20:36.24 | [TK]D-Fender | clyrrad : not wuite... read it GAIN..... |
20:36.47 | clyrrad | are you talking about the GoTo sub? |
20:36.54 | clyrrad | there are a few ways it can be written |
20:36.58 | clyrrad | im on the wiki right now |
20:37.38 | clyrrad | shows 6 ways of writing it.... all do not show a pattern match example... im not exactly sure what you want me to see? |
20:38.08 | clyrrad | This is what I am reading http://www.voip-info.org/wiki-Asterisk+cmd+goto |
20:42.51 | clyrrad | TKD? |
20:43.38 | the-chaos5 | have anyone installed asterisk on suse? |
20:46.21 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
20:47.31 | [TK]D-Fender | clyrrad : Please read the brackets on this line carefully : Goto([[context|]extension|]priority) |
20:47.39 | *** join/#asterisk tempest1 (n=asf@adsl-144-58-142.chs.bellsouth.net) |
20:47.52 | the-chaos5 | noone? |
20:47.58 | [TK]D-Fender | the-chaos5 : Seriously avoid. SUSE makes it hell to try and compile kernel modules for it. |
20:48.24 | [TK]D-Fender | clyrrad : Pattern matching is inhreent. its your call format that isn't quite right yet. |
20:49.00 | clyrrad | ok i am taking another look at it |
20:49.33 | the-chaos5 | ok think you i have a chance? |
20:49.41 | clyrrad | brackets messing me up a bit :p |
20:50.11 | clyrrad | I would think dialnumber would need to go where [context] is.... |
20:50.13 | *** join/#asterisk jcollie (n=jcollie@161.210.6.204) |
20:50.35 | jcollie | afternoon all |
20:50.57 | [TK]D-Fender | the-chaos5 : Chance yes, but its going to be hell I'm sure... |
20:51.28 | [TK]D-Fender | clyrrad : You going to create a context for evey possible number? I don't think so... keep at it.. you're close. |
20:52.10 | clyrrad | no - ideally it will just GoTo whatever it pattern matches thats already been defined... |
20:52.14 | clyrrad | thats the plan anyway :) |
20:54.33 | the-chaos5 | whitch system is the best for asterisk |
20:54.59 | [TK]D-Fender | the-chaos5 : Anything "standard" with normal libraries & kernels, etc.... |
20:55.21 | clyrrad | do the '|' mean anyting - or they are just trying to say "OR" as in its optional? |
20:55.25 | [TK]D-Fender | the-chaos5 : Debian, Slackware, RHEL/CentOS. FC can be a little problematic, but not "drastic" |
20:55.52 | clyrrad | chaos5 - CentOS works great with asterisk |
20:56.00 | [TK]D-Fender | clyrrad : You're starting to get it... you need 1-3 parms. and WHICH ones they are depends on how many you specify..... |
20:56.47 | clyrrad | TKD all I really have to specify are 2 things - the number that was entered.... and that it needs to start at priority 1 on the pattern match context.... |
20:57.14 | the-chaos5 | hmmm only one problem im very new with linux"ing" and i have already problems with suse |
20:57.26 | clyrrad | try CentOS |
20:57.57 | [TK]D-Fender | clyrrad : You're almost there.. now ask yourself WHERE are these extens my Goto is looking for.... |
20:58.37 | [TK]D-Fender | the-chaos5 : CentOS is indeed a great start, and you only need to take 2 very well known bug into account and you're set. |
21:01.14 | clyrrad | SWEEEEEEEEEEEEEEEEEET :) |
21:01.20 | clyrrad | thanks TKD |
21:01.26 | clyrrad | I understand what you meant now |
21:02.01 | clyrrad | I was not telling the GoTo to look for a pattern match under the context that defined all the outgoing pattern matches.... |
21:02.43 | clyrrad | therefore here was the proper syntax exten => s,16,Goto(my_phones,${dialnumber},1) |
21:03.18 | Assid | hey would asterisk take advantage of a duo core AMD ? |
21:03.39 | [TK]D-Fender | clyrrad : I knew you'd get it eventually, and I did want you to figure it out for yourself... |
21:03.59 | [TK]D-Fender | clyrrad : Worth more that way. Jut giving you the answer would have taken 2 seconds :) |
21:04.28 | [TK]D-Fender | Assid : no clue |
21:04.54 | Damin | Assid: Yes.. it would.. |
21:05.04 | clyrrad | TKD - Oh yea - I will remember that for life now :) I dont mind experimenting and learning - just needed to be pointed in the right direction - thanks bro! |
21:05.38 | Assid | Damin: i read alot of people were having issues with x86-64 with amd, |
21:06.12 | Assid | thats why most just use the generic kernel and packages |
21:06.22 | *** join/#asterisk swytch (n=ezcall@d83-179-145-193.cust.tele2.fr) |
21:06.28 | Assid | but if you do that, i dont see how it would take advantage of the hardware |
21:07.43 | clyrrad | TKD - do you know if there is an application that you can use to input a 'dialednumber' to test if it matches a given pattern? That would be a neat thing.... |
21:08.23 | [TK]D-Fender | clyrrad : You can create a dummy context with the matching patterns to test it. |
21:08.48 | [TK]D-Fender | clyrrad : where the patterns that match jump back to where you resume knowing the outcome. |
21:09.21 | clyrrad | Oh yea - I guess that would work too - and invalid ones you could just send to Hangup |
21:10.33 | the-chaos5 | so i have instaled it sucessfully but i have problems with starting |
21:10.48 | clyrrad | what distro? |
21:12.46 | [TK]D-Fender | clyrrad : No, invalid ones continue from the non-functional goto so you know what doesn't match |
21:13.40 | clyrrad | im going to implment that dummy extension to protect the call forward feature |
21:15.07 | Damin | Assid: Don't believe what you read. I run most of my production stuff on Dual Core Opteron and Intel boxes.. |
21:15.35 | Damin | Assid: And compile your own kernel if you want the best optimizations for your hardware.. |
21:16.23 | Assid | Damin: yeah always do.. but was just doing a quick study on it.. thats what i came across.. |
21:16.48 | Assid | but there is a benefit from running these high end machines even if they are 64bit right ? |
21:17.31 | rene- | does anybody has a copy of nvlinedetect lying around? i am in dire need of it... |
21:17.37 | Damin | Assid: Yep. Transcoding latency decreases substantially and the number of concurrent channels that you can transcode goes up.. |
21:18.02 | rene- | i asked newmantelecom for a copy but since my dns died for a couple of days i am out of luck in that matter too |
21:19.04 | Assid | how much do you push your dualcore opteron to? like how many simulanous channels and stuff befoee you feel the load |
21:22.15 | *** part/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
21:22.40 | *** part/#asterisk jcollie (n=jcollie@161.210.6.204) |
21:26.02 | the-chaos5 | YEHAAA why do you think on suse its hell it works ;) |
21:26.47 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-233.cybersurf.com) |
21:28.25 | *** join/#asterisk MindHack (n=mindhack@unaffiliated/mindhack) |
21:28.31 | *** join/#asterisk type0 (i=type0@216-67-23-157-cdsl-rb2.cwc.acsalaska.net) |
21:29.29 | type0 | hey.. i realize this isnt much of a help channel or anything other than asterisk |
21:29.38 | type0 | but does someone wanna look at my zone file real quick |
21:30.52 | type0 | http://pastebin.ca/105718 |
21:34.24 | *** join/#asterisk evisu (i=hIRC@bzq-88-152-254-110.red.bezeqint.net) |
21:41.43 | *** join/#asterisk esculapio__ (n=ESCulapi@reserved-231-14.tricom.net) |
21:43.09 | Damin | Asterisk is incapable of more than about 300 SIP channels before it shits itself on any platform, so I push to about 300. ;) |
21:43.43 | Assid | nice |
21:46.22 | *** join/#asterisk CANO-1982 (i=alejandr@190.48.74.28) |
21:56.08 | MACscr | type0, whats your questions about your zone |
21:56.20 | MACscr | and why are you using that type of address for your mx record |
21:56.40 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
22:01.03 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
22:06.26 | SpaceBass | anyone using Gizmo with asterisk? |
22:08.08 | Nivex | I use sipphone with asterisk |
22:08.11 | Nivex | which is fairly close |
22:21.08 | *** join/#asterisk Lukie (n=lantins@lucie.hex.lividpenguin.com) |
22:22.07 | Lukie | Anyone here got a 7961 ? Or 7941 ? I could do with having a word if poss. |
22:26.03 | *** join/#asterisk tarvid (n=tarvid@dpc6919101029.direcpc.com) |
22:28.08 | tarvid | I need to trunk a trixbox to trixbox. I can use SIP or IAX2, but how might I use a stun server in registration? |
22:28.34 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
22:32.30 | [TK]D-Fender | tarvid : * does not support stun and #asterisk does not support Trixbox. |
22:32.42 | tarvid | Ok |
22:32.45 | [TK]D-Fender | tarvid : As the channel topic pretty clearly states |
22:32.46 | tarvid | thx |
22:33.06 | *** part/#asterisk tarvid (n=tarvid@dpc6919101029.direcpc.com) |
22:36.01 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
22:43.10 | *** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
22:44.21 | StewLG | Can someone recommend the best online guide to a from-scratch install of Asterisk? I'm setting up a new install in parallel to my Trixbox install, and want to do things the long way this time. |
22:44.29 | *** join/#asterisk jeebusmobile (n=jeebusmo@ip68-7-6-157.sd.sd.cox.net) |
22:45.31 | Qwell | ~docs |
22:45.33 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
22:45.40 | Qwell | ~wikis |
22:45.41 | jbot | i guess wikis is http://www.voip-info.org |
22:45.48 | Qwell | StewLG: all of those are pretty good |
22:45.53 | StewLG | Thank you. |
22:45.57 | Qwell | I would personally recommend the oreilly book |
22:46.36 | StewLG | I have the oreilley book. Is it current enough to be relevant? |
22:46.43 | Qwell | sure |
22:47.05 | StewLG | Ok then. I'm just used to Linux making prior how-tos obsolete rapidly. |
22:48.56 | StewLG | Off to page 33. |
22:50.24 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
22:51.21 | sp0n9e | okay, i just finished the oreilly book...and now i want to build a home asterisk server to play with...thanks :( |
22:52.35 | StewLG | Is 1.2.7.1 current enough? |
22:52.57 | StewLG | Hmm, maybe not. |
22:53.18 | StewLG | But this is a scratch install. So fine for now. |
22:54.04 | *** join/#asterisk Amilcar_ (n=amilcar@201.34.202.17) |
22:54.07 | [TK]D-Fender | StewLG : Enough to be compatable with the docs you have sure... mostly bug fixes in later releases |
22:55.29 | *** join/#asterisk ptblank (n=MURDER1@68-169-175-248.lmdaca.adelphia.net) |
22:59.55 | sevard | [TK]D-Fender: post this on voip-info.org http://popularitydialer.com |
23:02.34 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
23:02.35 | *** mode/#asterisk [+o mog] by ChanServ |
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23:16.06 | *** join/#asterisk mrh2 (n=chatzill@host-84-9-253-120.bulldogdsl.com) |
23:16.46 | mrh2 | hi is it true agentcallbacklogin is being thrown out? |
23:17.31 | Amilcar_ | mrh2: yeah. :-( |
23:17.34 | Qwell | mrh2: eventually |
23:17.44 | *** join/#asterisk Iam8up (n=iam8up@156.63.171.206) |
23:17.46 | Qwell | but it *WILL* be there in 1.2, just deprecated |
23:18.00 | Amilcar_ | Qwell: 1.4, you meant. |
23:18.05 | Qwell | whatever :p |
23:18.08 | mrh2 | :( worried about what he can do for roaming extensions |
23:18.09 | Iam8up | is there any way to have the verbose text displayed from asterisk -r saved to some sort of a log? |
23:18.09 | Amilcar_ | :-) |
23:18.23 | Amilcar_ | mrh2: yeah, me too. |
23:18.37 | mrh2 | it is pretty business critical |
23:18.54 | sp0n9e | i don't want to sound like an idiot, but sip phones connect to a normal ip network, right? |
23:18.59 | Qwell | When it was announced that it would be deprecated, it was made clear that it is being removed because there are other better ways to do so |
23:19.29 | mrh2 | are there any examples floating around? |
23:19.49 | Qwell | It was also made clear that examples would be included with 1.4 |
23:20.24 | mrh2 | for roaming though not the dynamic queue stuff? |
23:20.41 | Qwell | very simple dialplan logic |
23:21.11 | Amilcar_ | Qwell: like we talk another day, i'm pretty sure that the roaming capabilities of agentcallbacklogin could not be replaced by dynamic members. |
23:21.56 | symlink | Amilcar_: setvar, astdb |
23:22.07 | Amilcar_ | at least, without the use of hacks like storing things in astdb... |
23:22.10 | Qwell | symlink: he was already told all of that |
23:22.14 | Qwell | astdb is HARDLY a hack |
23:22.22 | symlink | astdb is there for this sort of stuff |
23:22.38 | Amilcar_ | Qwell: no, please, i don't think astdb is a hack. |
23:22.52 | mrh2 | how about notation of agent as the originating channel in the cdr? |
23:23.22 | mrh2 | think that is not an option for changing with the dialplan |
23:23.55 | Amilcar_ | mrh2: maybe using custom fields with setvar.... |
23:24.13 | mrh2 | you can't change all the fields |
23:24.18 | mrh2 | just some |
23:24.40 | Amilcar_ | Maybe we have to use this "some".... :-) |
23:27.25 | mrh2 | ok so i guess it is wait and see - really hope there is an actual workable solution for this going forward otherwise i know at least one call centre that will be starting to look for an alternative |
23:27.51 | *** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
23:29.03 | icyfire0573 | When I call music on hold to test it, it says starting music on hold. It then immediatly says stopping music on hold. |
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23:49.22 | *** join/#asterisk pbuckley (n=pbuckley@mail.amcat.co.uk) |
23:49.48 | pbuckley | Hey there, Ive got a problem with the license for the business edition of asterisk, and wondered if anyone can help me? |
23:50.23 | mog | whats wrong pbuckley |
23:51.20 | pbuckley | Ive changed network card in the server, and upon restarting (and configuring the card) BE is no longer registered. I went to run registerbe but its asking for a code which I dont actually have. Someone else did this part and he is not available. |
23:54.49 | pbuckley | Im in the UK btw |
23:58.47 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
23:59.05 | pbuckley | Im either wondering if I can get a code, or find out what it is. Or if I put the hardware back the way it was if it will come back up. |
23:59.17 | mrh2 | not that i'm using the bus edition but might be based on mac address - i'd put the old card back until u can get hold of the person |
23:59.54 | pbuckley | Ill go and give it a go, Ive been sat here for 3 hours so far trying to sort it out. |