| 00:00.11 | *** part/#asterisk drclue (~drclue@ip-65-49-163-54.wireless.dyn.beamspeed.net) |
| 00:00.13 | bmoraca_work | the alternative is to manually program your round-robin...which could get difficult. |
| 00:00.27 | bmoraca_work | (though definitely possible) |
| 00:00.34 | gloin | rrmemory looks fine for this purpose |
| 00:01.07 | gloin | ack, gotta get away from desk now or never |
| 00:01.17 | gloin | maybe brain will work better tomorrow morning |
| 00:01.22 | gloin | thanks bmoraca_work |
| 00:01.47 | bmoraca_work | yep, i'm out the door too |
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| 00:25.23 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 00:25.29 | mattwj2002 | hi guys |
| 00:25.59 | mattwj2002 | anyone ever hear of grnvoip.com? |
| 00:26.13 | mattwj2002 | they appear to have just about every did under the sun |
| 00:27.24 | *** join/#asterisk drclue (~drclue@ip-65-49-163-54.wireless.dyn.beamspeed.net) |
| 00:27.31 | mattwj2002 | oh wait |
| 00:27.37 | mattwj2002 | nevermind I read it wrong |
| 00:28.07 | drclue | Sometimes I can really hate OS distros |
| 00:28.44 | drclue | So where was I when the OS of the moment took a hard left? |
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| 00:38.34 | carrar | sounds like using the wrong OS at that moment |
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| 01:18.08 | drclue | @carrar I've been at moments using the wrong OS since the mid 1970's , but at the moment the wrong OS is whatever I'm using. |
| 01:20.25 | drclue | My distro has been having some intermittent keyring issues over the last six months and occasionally for no apparent reason looses it's mind. I know how to cure it , but I have but one life and cannot really afford to go about fixing OS issues for free |
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| 01:21.26 | drclue | I think it will be quicker and easier to write myself a personal OS hack , which I guess I will do over the wqeekend |
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| 01:27.39 | tzafrir_laptop | grabs the cluebat |
| 01:28.16 | tzafrir_laptop | drclue, hmm... fix those keyring issues? |
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| 01:28.58 | drclue | @tzafir_laptoop You can bet I will recode the distro routines this weekend |
| 01:29.34 | drclue | They have had this problem fo six months and it really gets me pissed |
| 01:29.47 | tzafrir_laptop | drclue, if you can't figure out how to fix the keyring issues, well... |
| 01:30.20 | drclue | I can certainly figure out how to recover , and also know how to fix it , but I should not have to |
| 01:31.10 | tzafrir_laptop | hmm.... I guess that's your workaround for the problem |
| 01:31.38 | drclue | I have recovered from this a few times now , and know at least sorta how it happens and certainly how to fix it, but it really annoys me that I have to fix it myself |
| 01:31.40 | tzafrir_laptop | http://xkcd.com/763/ |
| 01:31.52 | tzafrir_laptop | Anyway, GTG |
| 01:33.38 | drclue | I'd much rather be focusing on my HUD code for AsterClick than trying to figure out the latest hole in my OS distro |
| 01:35.58 | DogBoy | what distro is that drclue |
| 01:36.42 | drclue | Well the dogstro of the day is ubuntu |
| 01:37.46 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
| 01:40.39 | p3nguin_ | laughs uncontrollably |
| 01:41.24 | p3nguin_ | Someone using Ubuntu wonders why things are broken. That's funny. |
| 01:42.04 | DogBoy | what do you use p3nguin_ |
| 01:42.53 | tzafrir_laptop | Someone not bothering to troubleshoot an issue, and then goes on to set up Asterisk. Oh well |
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| 01:45.43 | p3nguin_ | dogboy: Desktop - ArchLinux; Servers - FreeBSD, OpenBSD, ArchLinux, CentOS, SME Server (distro built on CentOS), Windows Server 2003 |
| 01:47.15 | drclue | Actually I have a house full of machines with different distros and none of them can claim sainthood |
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| 01:49.17 | DogBoy | me too but I mostly use debian |
| 01:49.28 | DogBoy | I've used ubuntu before |
| 01:49.57 | p3nguin_ | There's not a lot of operational differences, as far as I know. |
| 01:50.32 | DogBoy | but of the bsd variants I've only used openbsd |
| 01:51.30 | DogBoy | hmm, it's asking me for the ITU-T telephone code |
| 01:51.40 | DogBoy | I guess I can google on that |
| 01:51.43 | DogBoy | no clue |
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| 01:54.03 | drclue | My debian is sorta OK , as it's hard lefts are simply crashes and next boot its happy |
| 01:54.55 | drclue | There is no OS without it's hard lefts , but some hard lefts annoy me more than others |
| 01:57.44 | p3nguin_ | You mean "its har[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[C[Cd lefts" and "it's happy," I presume. |
| 01:57.59 | p3nguin_ | what the heck |
| 01:58.14 | DogBoy | there goes one now |
| 01:58.59 | p3nguin_ | "its hard lefts" and "it's happy" |
| 01:59.04 | p3nguin_ | There! |
| 01:59.24 | p3nguin_ | I have no idea what that was all about. |
| 01:59.32 | drclue | Hard lefts is I guess an ancient phrase that predates Microsoft and Apple so I guess I would simply be showing my gray hairs by mentioning "hard lefts" |
| 02:02.47 | drclue | "hard lefts" is pretty much any instance where the proper interaction with software or equipment results in an outcome beyond the control of the user |
| 02:04.21 | DogBoy | my computer usage predates those companies also |
| 02:04.53 | p3nguin_ | I didn't start using computers until about 1984. |
| 02:05.05 | p3nguin_ | <-- youngster |
| 02:05.29 | drclue | @DogBoy We ought to compare our eyeglass prescriptions I guess :) |
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| 02:07.38 | drclue | @p3nguin So I guess your hard drive if any at the time was 5-10 megs. You could today probably store maybe two of your favorite bands tuness on that MFMor RLL drive |
| 02:08.35 | p3nguin_ | I think the smallest hard drive I still possess is 250M. |
| 02:09.03 | drclue | I remember decorating my Christmas tree with streaming punched tape |
| 02:09.40 | drclue | Gold on one side , green on the other |
| 02:10.49 | drclue | Do not fold spindle or mutilate was printed on most of my programming cards back in the day |
| 02:11.26 | drclue | We had quality cards , as the others did not say shit |
| 02:11.47 | drclue | 150 baud was super fast |
| 02:12.22 | drclue | 4K of memory was as impressive as a 12 inch penis |
| 02:13.01 | coppice | and a lot bigger than 12" |
| 02:13.21 | coppice | nobody used 150bps. 110 was the norm |
| 02:14.12 | drclue | Now wait a minute , American scientific was doing 8 inch floppies and most hard drive platters were under 20 inches |
| 02:15.02 | coppice | hard drive platters were all 14", so the overall drive could fit in an 19" rack |
| 02:15.31 | drclue | Well I was never the norm. I was striping multiple nacho 150 baud modems in the day and was doing stolen speeds near what today is bad DSL |
| 02:16.50 | drclue | Hard drive platters came in a lot of sizes , covering a range of teens |
| 02:18.05 | drclue | I never cared that much as I aimed to take tape , platters or whatever and spread it out over the memory of many mainframes and minis. |
| 02:18.29 | coppice | 14", 8", 5.25", 3.5". attempts to do anything else died very quickly |
| 02:18.55 | drclue | The days when folks did not think there might be folks using there systems without permission |
| 02:20.06 | drclue | I never gave two shits about the medium. I always stripped multiple machines on multiple networks around the world for whatevr I wanted to do |
| 02:21.27 | drclue | Bank of America worked just fine for storage |
| 02:29.03 | drclue | What really pisses me off about when the ubuntu keyring takes a left is that all my providers pull in their horns too and I have to have conversations with everyone and their uncle. |
| 02:30.38 | drclue | I do like the anal retentiveness of my providers , as that is at least in part why I picked them, but keyring failure without cause really makes me want to kill,spindle , mutilate |
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| 03:38.03 | Tulga | I have "[Jul 9 11:37:27] WARNING[15830]: pbx.c:3769 __ast_pbx_run: Channel 'DAHDI/124-1' sent into invalid extension '3030' in context 'from-outside-redir', but no invalid handler" error on CLI. I created 3030 in extensions.conf. where is my mistake? |
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| 03:47.40 | seanjohn | can someone point me to how to use selinux with asterisk? |
| 03:48.07 | joobie | seanjohn, what about it? |
| 03:48.59 | seanjohn | I have always seen, from two years ago, that we have to disable selinux for asterisk. |
| 03:49.17 | joobie | guys, i have a polycom 320 with 2 line keys.. and have one registration to asterisk with the "number of line keys" set to 2 (so that I can select line 2 and dial,etc).. Is there a way I can use the dial cmd so that it dials line 2 of the phone instead of line1? |
| 03:49.51 | joobie | seanjohn, you can run selinux in an audit mode.. i think it's permissive mode frmo memory |
| 03:49.53 | seanjohn | yeah, joobie, dial(sip/trunkname/number) |
| 03:50.06 | joobie | seanjohn, set it to that.. and log what it trips up on |
| 03:50.11 | joobie | fix it.. then log more |
| 03:50.21 | seanjohn | ok, i was going to go that route |
| 03:50.36 | joobie | it's the best route, because the way you've setup asterisk will largely impact your selinux rules |
| 03:51.22 | joobie | btw, that dial(sip/trunkname/number) will dial the handset, but it will dial the line key 1 on the handset |
| 03:51.37 | joobie | it will only dial line 2 of the handset, if line 1 is busy.. |
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| 03:51.46 | joobie | can i set it up so that it always dials line 2? |
| 03:58.54 | ChannelZ | Not unless there is some SIP header you can attach to clue the phone in or something |
| 03:59.39 | Tulga | I want include some files to [from-outside-redir] group. #include <filename> or just include <filename>? |
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| 04:12.28 | joobie | thanks ChannelZ |
| 04:14.23 | joobie | ChannelZ, is it possible to setup an extension on the polycoms so that it's attached to a line key for inbound, but u cant make outbound on it? |
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| 04:14.51 | seanjohn | joobie, you can do dial(sip/trunkname/number@yourcontext,options) |
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| 04:16.34 | joobie | seanjohn, cant seem to see an option that lets me specify line 2? |
| 04:16.55 | seanjohn | are you using zaptel? |
| 04:16.59 | seanjohn | or dahdi |
| 04:17.12 | joobie | .. what im trying to achieve is, each phone has their own line (line1), which is a unique extension per phone.. but the reception phone, i want to make that ring for say 10 seconds.. if it doesnt answer, i want line 2 on all the other phones to ring so anyone can collect hte call |
| 04:17.19 | joobie | dahdi |
| 04:17.55 | joobie | line2 on every other phone i only want to be able to receive calls on.. so line 1 is the only one they can dial out of but they can receive reception calls on line 2 |
| 04:18.08 | seanjohn | I'm not really asking if you have dahdi installed; are you using PRI? |
| 04:18.08 | joobie | if they want to dial out, they have to use line 1 |
| 04:18.16 | joobie | yea |
| 04:18.17 | joobie | pri |
| 04:18.49 | seanjohn | you have to make a dial plan and your dial would be dial(zap/channelnumber/numbertodial@yourcontext) |
| 04:19.48 | ChannelZ | That's not what he's asking |
| 04:19.52 | joobie | i can dial the phone fine |
| 04:20.01 | seanjohn | http://www.voip-info.org/wiki/view/Asterisk+howto+dial+plan |
| 04:20.09 | joobie | seanjohn, that's not the problem bro |
| 04:20.47 | seanjohn | What is he asking ChannelZ. All lines are associated with a tech of sip, zap, iax, h video |
| 04:21.11 | seanjohn | there's no "line 1 line 2 line 3" and so on with asterisk, just the ATA's or devices |
| 04:21.43 | joobie | seanjohn, i have a polycom 320 on each users desk.. the 320 has 2 physical line keys on the phone |
| 04:21.47 | ChannelZ | Yes but he's talking about from the phone's point of view. It's got 2 line keys |
| 04:22.04 | joobie | seanjohn, each phone has its own unique extension registered with asterisk.. and they are registered against line key 1.. |
| 04:22.36 | joobie | seanjohn, what i want to do is, on one of the phones (which is the receptionist), they they dont answer the phone in X seconds, i want all the phones in the office to ring on line 2 so that anyone in the office can collect the call |
| 04:22.39 | seanjohn | joobie: its either ringgroup, pickupgroup, or device |
| 04:22.52 | joobie | line 2 should only be used for this purpose, as an inbound overflow for receptionist.. any calls out should be done from line1.. |
| 04:22.54 | seanjohn | unless using users instead of extensions |
| 04:24.17 | ChannelZ | joobie, can that phone not do separate registrations for the two lines? |
| 04:24.23 | joobie | but whats the best way to setup the phones for this sort of configuration? ie. should i register seperately for each physical line on the phone or should i just do one registration on the phone and do "num of line keys = 2" and then do something funky with asterisk fore this to work |
| 04:24.29 | joobie | ChannelZ, it can |
| 04:25.26 | joobie | in the back of my mind im thinking register each phone to a reception extension on line 2... but (a) i dont know if this is necessary and (b) not sure how can i then make specific phones not be able to dial out on the line 2 but let reception to dial out on line 2 |
| 04:26.01 | ChannelZ | You're really just wanting to use the 'line 2' key as an indication that a ringing call was one the receptionist didn't get? |
| 04:26.11 | seanjohn | i would register separately and control through the dial plan; I don't utilize switchboards |
| 04:26.31 | ChannelZ | I mean you mentioned wanting to block outgoing calls from that key but I'm not sure why it matters? |
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| 04:31.21 | ChannelZ | wanders off for a potty break |
| 04:32.10 | joobie | ChannelZ, yea that's the purpose |
| 04:32.28 | joobie | ChannelZ, it's to do with the number of concurrent calls per user.. i dont want to give them the ability to make too many calls at once |
| 04:32.39 | joobie | this is going to let them make another call if they can on the outbound which iw ant to avoid |
| 04:33.00 | Raden | yawn |
| 04:43.51 | ChannelZ | for what, conferencing? Chances are the phone can do that its self already unless you have turned it off |
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| 04:44.55 | DogBoy | oh snap |
| 04:45.08 | DogBoy | got 1.6 working on debian on nslu2 |
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| 04:50.24 | p3nguin_ | Limiting calls should probably be done with call-limit in each peer definition, I would think. |
| 04:52.11 | p3nguin_ | Having k3b BLASTING the trumpet sound after a successful burn is a bit ridiculous. |
| 04:54.10 | ChannelZ | ImgBurn does that too, the big "ta-da" sound.. scares the shit out of me and I always forget to turn it off |
| 04:54.24 | p3nguin_ | Oh, that was sent to the wrong window. |
| 04:54.57 | p3nguin_ | Had I sent it to the right channel, it would have been preceeded by this: |
| 04:55.01 | p3nguin_ | KDE 3 used to have a way to access the control panel without having to actually run KDE (I think it was kcontrol). Any idea how I can configure how KDE apps use my sound card without actually running KDE? I would really like to make KDE things use PCM so I can control their sound volume more easily. |
| 04:55.31 | DogBoy | oh man I hate those k3b sounds |
| 04:56.10 | DogBoy | my brother has his windows machine hooked up to these speakers and he shuts it down at 3 am, and it's like an amplified windows noise |
| 04:56.48 | p3nguin_ | I recently upgraded all kde3 stuff, so now nothing behaves like I want. I don't use KDE, so I don't know how to configure those things. |
| 04:57.02 | DogBoy | kde sounds is not something that needed to be emulated from windows |
| 05:00.48 | p3nguin_ | I don't know anything about that, either. I haven't ran Windows on my desktop since 2002. |
| 05:01.15 | DogBoy | yea |
| 05:01.18 | DogBoy | me either |
| 05:01.29 | DogBoy | around that time, I think I quit about 2001 though |
| 05:01.36 | DogBoy | or maybe it was earlier |
| 05:04.33 | DogBoy | like trying to remember when I quit smoking or became vegetarian |
| 05:04.39 | DogBoy | who cares |
| 05:04.47 | DogBoy | I turned away from the dark side |
| 05:05.19 | joobie | ChannelZ, each phone has call-limit=2 set on its registration |
| 05:05.36 | joobie | ChannelZ, if i allow this new reception extension to make calls out, it lifts that restriction |
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| 05:18.39 | Raden | p3nguin_, what u use for GUI ? |
| 05:18.47 | Raden | KDE 4 annoying and bloated :( |
| 05:18.59 | p3nguin_ | GUI of what? |
| 05:19.08 | Raden | like KDE ? |
| 05:19.28 | Raden | u dont use kde u said, what do you use ? |
| 05:19.42 | p3nguin_ | e17 |
| 05:19.50 | Raden | what that ? |
| 05:20.06 | p3nguin_ | a window manager that tries to act more like a desktop environment. |
| 05:22.55 | DogBoy | heh |
| 05:22.57 | DogBoy | really? |
| 05:23.13 | Raden | enlightenment ? |
| 05:23.47 | DogBoy | yea that would be the one |
| 05:23.49 | Raden | cause KDE 4 really sucks in my book :( |
| 05:23.56 | Raden | liked 3 a lot better |
| 05:24.06 | Raden | looking for something different now |
| 05:24.43 | DogBoy | using gome right now |
| 05:24.51 | DogBoy | but my fav is ratpoison |
| 05:24.59 | DogBoy | too lazy to set it up |
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| 05:32.05 | p3nguin_ | raden: How about xmonad or awesome? |
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| 06:10.23 | Micc | Is there anyway to force an aastra phone out of DND mode? |
| 06:10.35 | Micc | remotely that is, from asterisk. |
| 06:11.01 | Kyosh | did the user set it in DND? |
| 06:16.58 | Micc | yes, on the phone, they set it with the button on the phone. |
| 06:17.55 | Micc | if I had the phones setup to do xml, I bet theres a way to do it with some xml command. |
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| 06:19.28 | Micc | I got ahold of someone on their cell phone, so I don't need to worry about it until next time. |
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| 06:53.40 | giany | Is there some rule of iptables or some software that we can install to block DoS attacks and authentication requests ( sip register) in Asterisk ? |
| 06:55.30 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 06:55.49 | ChannelZ | well you can block SIP from the outside world if you don't need anonymous sip calls, but you can't really prevent someone from throwing mass amounts of traffic at your IP anyway. |
| 06:56.50 | Alton35 | $IPTABLES -A INPUT -i $OUTSIDE -s 204.155.28.10 -d 0/0 -p udp --dport 5060 -j ACCEPT # SIP from Sipgate |
| 06:56.58 | Alton35 | something like that |
| 06:57.01 | Alton35 | block everything else |
| 06:57.34 | ChannelZ | iptables -A INPUT --in-interface eth0 --protocol UDP --dport 5060 --jump DROP |
| 06:57.51 | ChannelZ | assuming your WAN traffic is separated on eth0 for instance |
| 07:02.00 | giany | ok, thanks for your advices |
| 07:11.19 | carrar | giany, you can get a firewall with a IDP system to look for failed SIP registration signatures and then automate a block on that IP |
| 07:11.52 | carrar | could also do something like that by tailing some logs in asterisk also |
| 07:12.02 | carrar | and block them with iptables |
| 07:12.24 | carrar | as a Juniper fw and IDP system is going to very expensive |
| 07:12.31 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 07:12.35 | *** join/#asterisk ttwhy (~tekkno@p4FECE5FA.dip.t-dialin.net) |
| 07:14.22 | carrar | look at sshguard |
| 07:14.37 | carrar | modify that |
| 07:18.29 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 07:26.35 | Alton35 | there is a firewall script that kicks around the internet |
| 07:26.38 | Alton35 | lemme see if I can find it for him |
| 07:28.16 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:32.52 | Alton35 | http://www.alton-moore.net/downloads/programming/misc/firewall_script <-- decent script, can be modified for your use |
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| 08:32.39 | Mish- | Okay, "exten => _1.,1,Dial(SIP/${EXTEN:1}@02825508477,30,r)" is letting me use 9 to dial an outside line. As I won't be using internale extensions at all, how can I make it so any number dialed is sent externally? |
| 08:34.09 | voxter | any of you guys use polycom ip4000's and ever notice that when speaking the speaker goes into a volume suppression mode? |
| 08:45.18 | *** join/#asterisk N|Xgurru (~NXgurru@119.152.230.43) |
| 08:55.13 | sawgood | If I wanted to copy over a pre-recorded 'greeting' for someone's voicemail box ... (intead of them speaking into the phone) ... what directory would this .WAV file be put in? |
| 08:55.54 | sawgood | I thought maybe it should go in /var/spool/asterisk/voicemail/device |
| 08:55.58 | sawgood | But I was not sure |
| 09:00.13 | sawgood | anyone have any idea where I should place my .wav file to be used as the 'greeting' for an extension |
| 09:02.33 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
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| 09:12.53 | *** join/#asterisk N|Xgurru (~NXgurru@119.152.141.137) |
| 09:18.52 | sawgood | got it ... customized voicemail greetings! |
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| 09:49.06 | _Raptor_ | hi |
| 09:49.32 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.55) |
| 09:49.45 | _Raptor_ | how can i perform a partial matching of the calling number: this does not work: |
| 09:49.58 | _Raptor_ | exten => 27943/090176.,1,Noop(DEEEEEEEE) |
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| 11:20.45 | *** join/#asterisk ichdasich (tfiebig@gallifrey.wybt.net) |
| 11:21.48 | ichdasich | hi there. i installed tzdummy on an debian lenny sparc64 following the instructions posted here, in the section 'installation lenny' |
| 11:21.51 | ichdasich | http://www.voip-info.org/wiki/view/Asterisk+timer+ztdummy |
| 11:22.04 | ichdasich | asteriks refuses to work with that module, and gives [Jul 9 12:58:20] ERROR[323]: asterisk.c:2974 main: You have Zaptel built and drivers loaded, but the Zaptel timer test failed to set ZT_TIMERCONFIG to 160. |
| 11:22.21 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 11:22.30 | ichdasich | /dev/zap is accessible to the corresponding user. |
| 11:23.27 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 11:23.41 | ichdasich | and zttest is working just fine. |
| 11:23.43 | ichdasich | any ideas? |
| 11:42.23 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:48.40 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
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| 12:04.21 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
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| 12:09.07 | Jimmy00784 | Hi, I am an Asterisk enthusiast and new to Asterisk |
| 12:09.16 | Mango | HI Jimmy |
| 12:09.20 | Jimmy00784 | Hi Mango |
| 12:09.59 | Jimmy00784 | I have the software installation completed, and I can connect to the console successfully |
| 12:10.20 | Jimmy00784 | I was hoping to configure my modem to work with Asterisk |
| 12:10.50 | Mango | In what way? |
| 12:11.16 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 12:11.21 | Jimmy00784 | to make outbound calls, or receive inbound calls, or both.... |
| 12:11.42 | Jimmy00784 | It's a voice modem on my Laptop |
| 12:11.52 | Mango | I'm not entirely sure that's possible. |
| 12:12.16 | Mango | It can be done with some models of Intel PCI modem, but since it's a voice modem, you may be out of luck. |
| 12:12.59 | Jimmy00784 | It's hard to believe that an open source project of such great magnitude would not have a provision for voice modem.... |
| 12:14.19 | Jimmy00784 | does asterisk provide a way to introdice new device classes? |
| 12:14.37 | [TK]D-Fender | Jimmy00784: Voice modems aren't full duplex and lack other features. |
| 12:15.02 | Mango | The quality of the ones I have tried is crap, too. |
| 12:15.16 | [TK]D-Fender | Jimmy00784: And no, unless your device is based on one of the 2-3 chipsets the X100P was branded under it is worthless for your project |
| 12:15.26 | Jimmy00784 | Thanks Fender, I understand that... |
| 12:15.46 | [TK]D-Fender | Jimmy00784: No "buts". |
| 12:16.32 | Jimmy00784 | I understand, no buts.... is there any way to use those modems for what ever features they can provide? |
| 12:16.35 | [TK]D-Fender | Jimmy00784: It isn't a worthwhile venture to go writing DAHDI drivers for 2-bit junk modem out there that even has a prayer of working for this. |
| 12:16.41 | [TK]D-Fender | Jimmy00784: No. |
| 12:17.33 | [TK]D-Fender | Jimmy00784: So if you have POTS line you want to use with * then you'll just have to pick up a compatible device |
| 12:18.00 | Jimmy00784 | Ok, in a more broader context, to introduce new device classes, would one have to write DADHI drivers if one is not already available? |
| 12:18.15 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:18.22 | [TK]D-Fender | Jimmy00784: Correct |
| 12:18.30 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 12:18.50 | Jimmy00784 | I see... I appreciate you help on this, Mango, and Fender. |
| 12:18.59 | Jimmy00784 | Thanks |
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| 12:47.57 | wcselby | o/ |
| 12:49.05 | tuxx- | \o |
| 12:49.12 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:51.11 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-uwukkileeuvibojz) |
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| 13:18.31 | *** join/#asterisk jkroon (~jkroon@41.157.80.23) |
| 13:19.01 | jkroon | what could possibly be wrong if I can make outbound calls over a BRI but not receive any? (Digium 410P card) |
| 13:19.28 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 13:20.06 | [TK]D-Fender | jkroon: What do you see? |
| 13:20.34 | jkroon | sync frames in intensive debug ... can pastebin the intensive debug (phone attached to the analog port does ring) |
| 13:21.25 | jkroon | http://pastebin.com/3QWjB4sr |
| 13:21.51 | jkroon | NT2 unit from Telkom. |
| 13:23.51 | jkroon | ok, tried a different (known to work with siemens PABX), same result. |
| 13:24.08 | [TK]D-Fender | jkroon: What does your telco say they see? |
| 13:24.36 | jkroon | if you can get them on the line for me i'll happily ask them :( |
| 13:25.05 | jkroon | but sarcasm put aside the point is that it works when plugged into a siemens PABX. |
| 13:25.25 | jkroon | reality is getting hold of them is difficult to say the least. |
| 13:26.34 | [TK]D-Fender | jkroon: I would certainly call Digium support ASAP if you're under warranty |
| 13:27.18 | jkroon | i am, got a call open with them via email. just going to test with older (2.2) dahdi drivers. I upgraded them last weekend to 2.3 and the unit hasn't been in much use since then. |
| 13:27.23 | WIMPy | I can't see any incomming call. |
| 13:27.41 | *** join/#asterisk Weazel (~bla@keshet.kolcore.com) |
| 13:28.45 | Weazel | hey guys, can anyone help me with dtmf and disa ? i have a context for disa, but for some reason when i get the dialtone to dial, it wouldn't take my dtmf's although it dtmf works on ivr etc |
| 13:30.38 | jkroon | WIMPy, that's the point. |
| 13:31.46 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 13:32.14 | [TK]D-Fender | Weazel: What kind of channel? Pastbin the backup showing they navigated the IVR to get there and include your channel driver configs and all related dialplan |
| 13:32.19 | [TK]D-Fender | ~pb |
| 13:32.20 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:32.21 | [TK]D-Fender | ^^^ |
| 13:34.17 | Weazel | thanks just a sec |
| 13:34.24 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:35.01 | jkroon | [TK]D-Fender, libpri change perhaps? |
| 13:35.48 | [TK]D-Fender | jkroon: Could be... I don't know much about BRI |
| 13:36.04 | jkroon | ah ok |
| 13:38.16 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:38.16 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.21 | Weazel | [TK]D-Fender: sry it took me long time -- http://pastebin.com/ccA2n3rJ |
| 13:43.31 | Weazel | i had to call the guy to make a test so i can copy the verbose |
| 13:44.13 | *** join/#asterisk korcan (~korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 13:44.18 | Weazel | this is when they called the inbound that goes to Custom-destination and to the custom-disa context |
| 13:44.39 | Weazel | they recieve the long dialtone to start the call, but they can't really press anything from that point, and it just goes to timeout |
| 13:44.49 | [TK]D-Fender | Weazel: that isn't the COMPLETE call |
| 13:45.02 | [TK]D-Fender | Weazel: or the complete dialplan |
| 13:45.16 | Weazel | omg sorry i'll repaste |
| 13:45.22 | [TK]D-Fender | Weazel: No mention about * version either |
| 13:46.42 | [TK]D-Fender | Weazel: You should also Answer() first |
| 13:46.48 | Weazel | http://pastebin.com/dRY0WLgi |
| 13:46.52 | Weazel | i'll add the Answer now |
| 13:46.54 | Weazel | and its 1.6 |
| 13:46.56 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:47.05 | jkroon | [TK]D-Fender, ptp vs ptmp |
| 13:47.08 | Weazel | and its 1.6.2.6 |
| 13:47.31 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 13:47.38 | [TK]D-Fender | Weazel: exten => 100,n,DISA(no-password|from-internal) <-- try "," instead of "|" |
| 13:47.54 | [TK]D-Fender | Weazel: Not sure if the instructions are still wrong and it requires "," as the delimiter |
| 13:48.32 | Weazel | i'll try it now thanks |
| 13:51.14 | Weazel | it gives him now an instant hangup |
| 13:51.15 | Weazel | http://pastebin.com/WuLNjgmP |
| 13:51.59 | Weazel | oh wait my bad |
| 13:52.01 | Weazel | sec checking again |
| 13:54.23 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 13:54.25 | Weazel | the Answer() was hanging up the call... but the "," delimiter did the trick |
| 13:54.31 | Weazel | [TK]D-Fender: Thanks alot mate |
| 13:55.39 | *** join/#asterisk coppice (~chatzilla@202.64.175.107) |
| 13:56.13 | [TK]D-Fender | Weazel: You should read the changes between versions .... |
| 14:01.32 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:01.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:02.55 | *** join/#asterisk patrick-- (~patrick@eos.openroot.de) |
| 14:03.39 | patrick-- | Hey all. when redirecting an incoming call from the pstn to an external number in the pstn via my asterisk, how can i send the actual CLIP of the caller and not the asterisk' ? |
| 14:03.47 | *** join/#asterisk N|Xgurru (~NXgurru@115.186.34.64) |
| 14:07.22 | [TK]D-Fender | patrick--: * will pass on whatever the current CID is unless your config told it otherwise or unless your outbound call ignores what * sends. |
| 14:10.10 | Gugge | and most PSTN networks will by default ignore what * sends .... |
| 14:11.56 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:19.24 | giany | anyone can tell me in this line : Executing [receive@fax-rx:20] ReceiveFAX("SIP/1101-083a0378", "/home/tvox/fax_files/fax-2-rx.tif,d") in new stack |
| 14:19.33 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:19.33 | giany | what does the d after , does? |
| 14:21.42 | [TK]D-Fender | giany: "core show application ReceiveFAX" <- |
| 14:25.16 | giany | [TK]D-Fender: thx..but it isn't too usefull |
| 14:26.41 | [TK]D-Fender | giaWell what does it say? |
| 14:26.49 | [TK]D-Fender | giany: Well what does it say? |
| 14:27.22 | giany | http://pastebin.com/Xa3RhCqZ |
| 14:28.40 | [TK]D-Fender | giany: Looks like it ISN'T an option and probably does absolutely nothing |
| 14:28.49 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 14:30.37 | giany | [TK]D-Fender: nevermind , i |
| 14:30.45 | giany | found what is all about |
| 14:30.59 | giany | it enables/disables ECM |
| 14:33.40 | *** join/#asterisk thieums63 (~mathieu@LPuteaux-156-14-100-43.w80-12.abo.wanadoo.fr) |
| 14:34.34 | gandhijee | anyone here have a cisco 7970? |
| 14:39.52 | pabelanger | Any recommendations for g722 phone? |
| 14:40.07 | Naikrovek | pabelanger: Polycom IP 335, 450, 650 |
| 14:41.05 | pabelanger | Naikrovek: danka |
| 14:42.10 | Naikrovek | gandhijee: not a lot of cisco phone owners in here, i don't think. p3nguin has some though, and apparently he loves them |
| 14:42.18 | Naikrovek | not sure about that EXACT model though |
| 14:42.42 | gandhijee | word |
| 14:43.11 | gandhijee | i finally got this 7970 flashed over to SIP, but i think i need someone with CCM to generate a config for me |
| 14:44.09 | Naikrovek | p3nguin has his set up to use standard firmware, not sure he could help with that. he uses the cisco protocl |
| 14:44.21 | stix | Hi guys. Do you have any clue where to look when almost all my calls are being disconnected? I have tried with two different providers. The call gets established and then sometimes after a while it gets disconnected... |
| 14:44.36 | Naikrovek | stix: is NAT involved somwwhere |
| 14:45.12 | stix | yes |
| 14:45.18 | stix | everywhere :) |
| 14:45.25 | Naikrovek | i would be looking closely at that |
| 14:45.43 | Naikrovek | nat holes are created dynamically and they can close automatically |
| 14:45.51 | Naikrovek | during calls is unusual, however |
| 14:46.06 | Naikrovek | most often the symptom is one-way audio |
| 14:46.14 | Naikrovek | but disconnected calls are not uncommon |
| 14:46.19 | stix | hmm okay |
| 14:46.37 | stix | I will look into that now |
| 14:46.57 | *** join/#asterisk Bartockbatz (~chatzilla@c-24-62-161-95.hsd1.nh.comcast.net) |
| 14:47.28 | Naikrovek | ~sipnat |
| 14:47.29 | infobot | i guess sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:47.35 | Naikrovek | stix: ^^^^^^^^^^^ |
| 14:47.40 | Naikrovek | may help you |
| 14:47.55 | stix | which? |
| 14:48.19 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 14:54.46 | Bartockbatz | Hey all |
| 14:55.14 | Bartockbatz | okay - got the dialplan working to dial out - but I want to add one more feature |
| 14:56.05 | Kobaz | features schmeatures... we don't need no stinkin features |
| 14:56.39 | Bartockbatz | ha! Kobaz |
| 14:56.42 | Bartockbatz | exten => _X.,n,Dial(${OUTBOUNDICOM}/${EXTEN},60) |
| 14:57.28 | Bartockbatz | now - where would I look this up - I want to add to this a 3 digit number, then Asterisk prompts you for your number you want to dial |
| 14:57.55 | Bartockbatz | ${OUTBOUNDICOM} does work - I tested it |
| 14:57.56 | Kobaz | wooooow |
| 14:58.00 | [TK]D-Fender | BartYou've already dialed the number you wanted to dial based on that pattern. Why ask for more? |
| 14:58.00 | Naikrovek | yeah wow |
| 14:58.04 | Kobaz | all my unit tests passed from last night |
| 14:58.06 | Kobaz | woooooow |
| 14:58.11 | Naikrovek | woooooooooow? |
| 14:58.21 | Kobaz | they all passed! |
| 14:58.36 | Naikrovek | better check your tests. make sure they don't just return(true); |
| 14:58.43 | Kobaz | i think perl's socket->getline does not work across packet boundaries or something |
| 14:58.48 | Bartockbatz | [TK]D-Fender: the client has 3 digit extensions - wants to have one specifically for getting an 'outside line' |
| 14:58.52 | stix | Naikrovek, I think adding "nat=yes" to the trunk did the trick :) |
| 14:58.55 | Kobaz | i replaced it with my own getline... and now stuff is working nicely |
| 14:59.03 | Naikrovek | stix: nice |
| 14:59.09 | stix | thank you |
| 15:00.27 | Bartockbatz | [TK]D-Fender: I get you - but when the client is paying the bills, I do - unless it is just not freakin' possible |
| 15:00.40 | [TK]D-Fender | Bartockbatz: you jsut showed us a pattern for 2+digits calling out |
| 15:00.59 | Naikrovek | can't you put a password on the trunk that lets you leave the local phone system |
| 15:01.36 | Naikrovek | or (better) just do call logging so you can see what people are doing, then deal with abusers appropriately |
| 15:01.54 | [TK]D-Fender | Bartockbatz: DCC = evil. |
| 15:01.59 | Naikrovek | doesn't know, and has a headache |
| 15:02.16 | Bartockbatz | sorry - been a while since I used IRC - no more DCC |
| 15:03.05 | Bartockbatz | exten => _X.,n,Dial(${OUTBOUNDICOM}/${EXTEN},60) ? |
| 15:03.17 | [TK]D-Fender | [10:57]<Bartockbatz>now - where would I look this up - I want to add to this a 3 digit number, then Asterisk prompts you for your number you want to dial <-- why dial a big pattern and THEN ask what to dial? And what is "add to this 3 digit number"? This pattern is VARIABLE length. Add what? Where? |
| 15:04.15 | Bartockbatz | okay - using your SIP phone - internal extensions are 3 digits |
| 15:05.02 | Bartockbatz | what they want is to reserve a particular 3 digit extension that will prompt you for an external number |
| 15:05.11 | [TK]D-Fender | Bartockbatz: So go make one. |
| 15:05.57 | Bartockbatz | okay - that is where I am a little clueless - I want to add this to my dialplan - what Asterisk application can allow this? |
| 15:06.27 | Bartockbatz | "throw me a bone, people" - sorry for the dumb questions, but I am kind of new to this. |
| 15:06.44 | [TK]D-Fender | Bartockbatz: "core show application read" <- |
| 15:07.15 | Kobaz | sdfsadkfjhkasdhfkjsadf |
| 15:07.24 | Bartockbatz | magic - :) |
| 15:07.24 | Kobaz | thunderbird crashed... and now all it does is crash when it starts up |
| 15:07.56 | Naikrovek | delete profile, start over |
| 15:07.57 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 15:08.04 | Kobaz | hah, no way |
| 15:08.11 | Naikrovek | or, if your OS supports it, revert to previous snapshot |
| 15:08.29 | Kobaz | reverted to 3.0.1 |
| 15:08.32 | Kobaz | i was running 3.1 |
| 15:08.40 | Kobaz | i dont have teh snapsnots |
| 15:08.44 | Naikrovek | no i mean revert profle to previous disk snapshot |
| 15:08.45 | Naikrovek | ah |
| 15:09.22 | Naikrovek | i haven't used thunderbird in a very long time, can you import from one profile to another? |
| 15:09.38 | Naikrovek | if so, MOVE old profile, create new, then import from old, moved busted profile |
| 15:10.43 | Naikrovek | old & busted. new hotness. |
| 15:10.55 | Naikrovek | laughs at his boss. |
| 15:11.14 | Naikrovek | i wrote a query to show monthly minutes usage for our phone system |
| 15:11.24 | Naikrovek | boss is blown away by my sql skills. |
| 15:11.28 | Naikrovek | query is maybe 100 chars long |
| 15:11.29 | Naikrovek | heh |
| 15:11.42 | Kobaz | Naikrovek: oh... umm... i think it might have a profile importer |
| 15:11.42 | Naikrovek | i've written oracle queries that were KBs in size |
| 15:12.01 | *** join/#asterisk gnuday (~gnuday@78-105-162-223.zone3.bethere.co.uk) |
| 15:12.14 | Kobaz | Naikrovek: my biggest query so far spans about 100 lines |
| 15:12.26 | Naikrovek | they can get big |
| 15:12.28 | Kobaz | but that query is also using several views, which are each about 20 lines |
| 15:12.33 | Naikrovek | i used to know how to do some really neat stuff with oracle |
| 15:13.03 | Naikrovek | and i believe in shoving as much of the work as possible into the SQL query |
| 15:13.06 | Kobaz | i like postgres these days |
| 15:13.09 | Kobaz | oh yeah |
| 15:13.13 | Kobaz | application code sucks |
| 15:13.15 | Naikrovek | yeah postgres 9 looks awesome |
| 15:13.17 | Kobaz | throw it all on the database |
| 15:13.18 | gnuday | hi, has anybody managed to get sipwitch and asterisk working together i.e. cross registration. |
| 15:13.49 | *** join/#asterisk kn0x (~pinochle@67.159.48.101) |
| 15:14.50 | Kobaz | Naikrovek: yeah the built-in replication in 9 is going to be pretty cool |
| 15:14.50 | kn0x | anyway to run a script after stop gracefully? |
| 15:15.05 | Naikrovek | Kobaz: that's exactly what i was talking about. yes. |
| 15:15.39 | Kobaz | and exclusion constraints look exciting too |
| 15:15.54 | Kobaz | AND listen/notify will be able to send payloads |
| 15:17.03 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 15:17.31 | Kobaz | and join optimization is cool too |
| 15:22.57 | *** part/#asterisk AndyML (~alauppe@pool-173-49-137-72.phlapa.fios.verizon.net) |
| 15:23.02 | kn0x | or do i have to watch something with waitpid() |
| 15:24.19 | Kobaz | sounds like a good idea |
| 15:25.50 | Kobaz | now we need to have a continuous integration system going |
| 15:27.32 | Naikrovek | anyone know of a CDR query that can show the max number of simultaneous calls at any point in a time period |
| 15:27.38 | Kobaz | hah |
| 15:27.52 | Kobaz | Naikrovek: we had to write some funky stuff to do that |
| 15:28.11 | Naikrovek | yeah i'm thinking about how to do it and it seems non-trivial |
| 15:28.58 | Naikrovek | is calldate the time of the origin of the call or the end |
| 15:29.25 | Raden | Naikrovek, trying to figure out your max concurrent calls ? |
| 15:29.45 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 15:29.47 | Naikrovek | yes, but on a sliding scale. I want to see max simultaneous per month, week, day |
| 15:30.35 | Kobaz | there's always subqueries |
| 15:30.40 | Naikrovek | yeah |
| 15:30.50 | Naikrovek | i mean i have the calldate, i have the duration |
| 15:30.51 | Raden | very possible todo |
| 15:31.03 | Kobaz | Naikrovek: we wrote ours using the cdr csv file, and perl |
| 15:31.09 | Kobaz | i have the code somewhere |
| 15:31.17 | Kobaz | i'll have to ask my brother when he gets into the office |
| 15:32.14 | REdOG | alot of the example extension sections are defined as _X. how do I pass a call to that context when those are the only extensions? I keep getting errors about Channel sent into invalid extension |
| 15:32.24 | Kobaz | Naikrovek: the easiest way is to actually store the current simultenous calls every 5 seconds |
| 15:32.28 | Naikrovek | my stupid right-brained head thinks of things graphically and spatially, which is sometimes a hindrance in thinking of problems like this |
| 15:32.32 | Kobaz | Naikrovek: and then you don't need to calculate it |
| 15:32.58 | Naikrovek | yeah but eh |
| 15:33.01 | Naikrovek | the data is here |
| 15:33.04 | Kobaz | yeap |
| 15:33.07 | Naikrovek | just need to figure out how to get it out |
| 15:33.11 | Kobaz | that's always the problem, isn't it |
| 15:33.20 | Naikrovek | my SQL-fu is weak |
| 15:33.24 | Naikrovek | apparently |
| 15:33.33 | Naikrovek | need to think about this for a few minutes |
| 15:33.41 | Kobaz | here's the main thing with data |
| 15:33.50 | Kobaz | it's always easier to split stuff out, than to piece things together |
| 15:33.52 | [TK]D-Fender | REdSHOW US <- |
| 15:33.53 | [TK]D-Fender | ~pb |
| 15:33.54 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:34.00 | [TK]D-Fender | REdOG: ^^ |
| 15:34.19 | Naikrovek | the parts of my brain that I can't directly control will crank on this then suddenly i'll know the answer, or at least have an approach |
| 15:34.22 | Kobaz | so, your cdr query... is a case of piecing things together |
| 15:34.45 | Kobaz | so in other words... you're screwed |
| 15:35.53 | REdOG | [TK]D-Fender: http://pastebin.com/a22eHBKh |
| 15:35.53 | Naikrovek | well for each second (or each 5 second period of a time period) or whatever I can query the database for all calls started before it and extend out into that moment in time. |
| 15:36.14 | Kobaz | that's going to be a lot of queries |
| 15:36.29 | REdOG | typo in that one on Goto |
| 15:36.32 | REdOG | left in a . |
| 15:36.34 | Kobaz | i know there is a better approach... but I didn't write the code, so i don't know offhand |
| 15:36.43 | [TK]D-Fender | REdOG: exten => 3374070903,1,Goto(time,s.,1) <--- "s"... this is a literal LETTER. your pattern is purely NUMERIC |
| 15:36.45 | Naikrovek | yeah but i can optimize and get smarter about it once it's working inefficiently |
| 15:37.00 | [TK]D-Fender | REdOG: there is no "s" anything in there |
| 15:37.03 | Kobaz | yeah. that's what they all say |
| 15:37.08 | Naikrovek | heh |
| 15:37.22 | [TK]D-Fender | REdOG: Also your exten doesn'tr even haev a "1" priority |
| 15:37.26 | REdOG | I know, Ive been trying all kind of things there ... nothing seems to work.... |
| 15:37.29 | Kobaz | i once inherited a project that was a perl script, that generated an xml file for a realtime web dashboard... the project was to make it update every 5 minutes |
| 15:37.35 | Kobaz | Naikrovek: you know what the problem was? |
| 15:37.44 | Qwell | cron wasn't running? |
| 15:37.44 | bmoraca_work | wooo...i get to develop a residential VoIP billing platform! |
| 15:37.52 | Kobaz | the script took 8 hours to run |
| 15:37.59 | Qwell | ... |
| 15:38.05 | Raden | bmoraca_work, can u build one for me tooo :( |
| 15:38.09 | [TK]D-Fender | REdOG: Well you're juping to an exten for which no match exists, and to a priority which the only one you did make doesn't even have. |
| 15:38.21 | bmoraca_work | raden: if it works well, i could sell it to you :P |
| 15:38.36 | REdOG | How do I move it to [time] then? |
| 15:38.48 | Raden | bmoraca_work, ill just keep working on mine then :( |
| 15:38.52 | Raden | LOL |
| 15:38.53 | Kobaz | Qwell: apparently the guy who wrote it, didn't understand what a database was for... and was doing manual joins in perl, and individual row selects, on >5 million row datasets |
| 15:39.12 | [TK]D-Fender | REdOG: Also you call Return() in there This isn't even a GOSUB like it looks like it deserves to be. |
| 15:39.29 | [TK]D-Fender | REdOG: Go make this a proper Gosub or Marco. |
| 15:39.33 | [TK]D-Fender | Macro* |
| 15:40.35 | REdOG | k |
| 15:40.51 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 15:41.42 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 15:41.48 | Naikrovek | ouch, my ego. |
| 15:42.08 | Naikrovek | was going my query wrong, the billable minutes one |
| 15:42.10 | Naikrovek | oops |
| 15:42.33 | Naikrovek | ... |
| 15:42.39 | Naikrovek | 40,000 minutes in one month |
| 15:42.42 | Naikrovek | can't be right |
| 15:43.50 | bmoraca_work | is that what one of your customers used? |
| 15:43.55 | REdOG | what address do I use in a gosub for that context though? |
| 15:44.18 | Naikrovek | no this is just something i am working on to exersize my mysql/sql muscles |
| 15:44.23 | Naikrovek | nothing hinges on this data |
| 15:44.26 | Naikrovek | oh |
| 15:44.31 | Naikrovek | no this is what i used, my company |
| 15:44.39 | REdOG | gosub(time,s,n)? gosub(time,_X.,n) ? gosub(time,?,?) nothing I try seems to work |
| 15:44.42 | Naikrovek | the organization which employs me |
| 15:45.11 | bmoraca_work | ahhh |
| 15:46.02 | Naikrovek | i found my error because my provider sent me a billing report showing how they calculate it |
| 15:46.27 | Naikrovek | they bill me for all time i use their trunk, not just when a call is answered or whatever |
| 15:46.49 | Naikrovek | 42k minutes in jan '09 |
| 15:46.51 | Naikrovek | jeepers |
| 15:46.59 | Naikrovek | that was when we only had 4 trunks, too |
| 15:48.14 | [TK]D-Fender | REdOG: Make your actual exten "s" <--- |
| 15:48.44 | [TK]D-Fender | REdOG: and then call it properly via Macro, Gosub, etc and pass it PARAMETERS. |
| 15:49.40 | REdOG | I replace all the _X. w/ s? |
| 15:49.59 | [TK]D-Fender | REdOG: Yes. |
| 15:53.03 | kn0x | is there a way to run a script when asterisk shuts down |
| 15:53.26 | [TK]D-Fender | kn0x: add it to the script you use to run * in the first palce. |
| 15:53.27 | REdOG | just add to the init script? |
| 15:59.26 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 16:00.18 | Naikrovek | alright i think i have this worked out. my report matches my providers to within a hundred minutes or so per month. that must be rounding errors |
| 16:00.43 | Naikrovek | i go to 4 decimal places, they go to 1 |
| 16:00.53 | kn0x | [TK]D-Fender: im doing top gracefully though |
| 16:00.53 | Naikrovek | suppose i could adjust that as well but FEH |
| 16:01.44 | [TK]D-Fender | kn0x: What part of "whatever called * should execute whatever you want when it exits" is not completely clear by now? |
| 16:02.09 | [TK]D-Fender | kn0x: * was started via some shell script, right? then MOD IT. |
| 16:05.32 | Naikrovek | whoa |
| 16:05.37 | Naikrovek | office 2010 support ligatures. |
| 16:05.44 | Naikrovek | holy f |
| 16:05.51 | Naikrovek | offtopic but whoa |
| 16:06.59 | *** join/#asterisk jtodd (ska6fzoab8@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:06.59 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:07.48 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 16:08.00 | ZeXr0 | Anyone recalls the conversion of yesterday with incompetent admin and stuff like that ? |
| 16:08.33 | ZeXr0 | Because I've got a very good joke about that... |
| 16:08.46 | Qwell | ZeXr0: there are several such conversations per day |
| 16:08.50 | Naikrovek | lol |
| 16:08.52 | ZeXr0 | :P |
| 16:09.07 | ZeXr0 | Well it seems that our Asterisk server got "hacked" |
| 16:09.11 | Naikrovek | eep |
| 16:09.14 | ZeXr0 | well one of them |
| 16:09.22 | ZeXr0 | Here's the joke |
| 16:09.29 | Qwell | and by "hacked", you mean they guessed the password of "1234"? |
| 16:09.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:09.45 | *** part/#asterisk thieums63 (~mathieu@LPuteaux-156-14-100-43.w80-12.abo.wanadoo.fr) |
| 16:09.50 | ZeXr0 | Qwell : Wow, do you work at my office without me knowing |
| 16:09.51 | ZeXr0 | :P |
| 16:10.07 | Qwell | You mentioned incompetence. That's rule #1. |
| 16:10.18 | Corydon76-dig | Queue SpaceBalls clip |
| 16:10.22 | ZeXr0 | Well then here's the second part of the joke :P |
| 16:10.23 | ZeXr0 | exten = _9NXXNXXXXXX,1,Dial(SIP/14383381940/1${EXTEN:1}) |
| 16:10.23 | ZeXr0 | exten = _91NXXNXXXXXX,1,Dial(SIP/14383381940/${EXTEN:1}) |
| 16:10.40 | Qwell | in [default], right? |
| 16:10.45 | Qwell | I'm shocked. Absolutely shocked. |
| 16:10.46 | ZeXr0 | RIght |
| 16:11.17 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:11.18 | ZeXr0 | All there's missing is a big flashing red light that says UNSECURE ASTERISK SERVER HERE ! |
| 16:11.26 | Naikrovek | doesn't get it. :( |
| 16:11.49 | ZeXr0 | Naikrovek : About everyone can dial in our system, and make phone calls |
| 16:12.14 | Naikrovek | yeah i get the implication, but i gleaned that from the "UNSECURE ASTERISK SERVER HERE" line. not because of anything else |
| 16:12.20 | Corydon76-dig | ZeXr0: do you also have allowguest=yes in sip.conf? |
| 16:12.28 | Qwell | Corydon76-dig: naturally |
| 16:12.32 | Naikrovek | i really need to work on learning the dialplan |
| 16:12.49 | ZeXr0 | Corydon76-dig : OMG ... |
| 16:13.07 | Naikrovek | i have allowguest=yes but i have to because of how my provider works. i firewall the CRAP out of the machine so no one by my provider can reach the phone server though |
| 16:13.32 | drmessano | 12345? I have that same combination on my luggage! |
| 16:13.32 | ZeXr0 | Well Naikrovek, I could say about our server, How do you configure a firewall ? |
| 16:13.41 | ZeXr0 | lol |
| 16:13.43 | Qwell | ZeXr0: README-SERIOUSLY.bestpractices.txt |
| 16:13.47 | Qwell | READ IT. SERIOUSLY. |
| 16:13.49 | Naikrovek | i do it at the router, the firewalling |
| 16:13.52 | Naikrovek | haven't been hacked yet |
| 16:13.55 | ZeXr0 | Qwell, I know, and I didn't set up the box |
| 16:13.59 | Naikrovek | just jinxed himself |
| 16:14.10 | Qwell | well, if you're surprised by the allowguest setting, you should still ready it :p |
| 16:14.11 | Qwell | read* |
| 16:14.18 | ZeXr0 | now I need to wait ... for a feedback to disable the server, and change about every thing in it |
| 16:14.29 | ZeXr0 | I'm surprise that it's set to yes :P |
| 16:14.41 | ZeXr0 | I didn't configure the server |
| 16:14.45 | drmessano | ZeXr0: Until then, what is the IP? |
| 16:14.46 | mocker | ZeXr0: Add a step to enter a passcode for every outbound call. :) |
| 16:15.01 | Corydon76-dig | ZeXr0: allowguest=yes is not insecure by itself... only if you allow outgoing calls in the default context |
| 16:15.49 | Corydon76-dig | The purpose of allowguest is to allow you to publish things like SIP/sales@yourcompanyname.com and have it work with no prior knowledge of a caller |
| 16:15.56 | bmoraca_work | has anyone used PortaOne's RADIUS deal with 1.6.2? |
| 16:16.24 | ZeXr0 | Qwell : Where's the Readme |
| 16:16.35 | Qwell | in the root of the source dir |
| 16:17.45 | drmessano | It's funny that people are so worried about VPN, SRTP, 3 levels of NAT, port knocking, and scripts to detect failed authentication attempts, but they put the lamest, most insecure dialplan ever on a box. |
| 16:18.01 | drmessano | "it starts with the dialplan, stupid" <-- My Tombstone |
| 16:19.33 | drmessano | There needs to be an undocumented alloweveryoneidontcare=yes that sets allowguest and ties the entire dialplan together like a big un-nested wirenut |
| 16:20.02 | Qwell | drmessano: trixbox mode? |
| 16:20.15 | drmessano | HA |
| 16:20.30 | drmessano | fonalityhackscript=veryyes |
| 16:22.26 | drmessano | "There is absolutely NOTHING to worry about. We are NOT doing ANYTHING suspicious here" - Kerry Garrison <-- prints to the console every 10 mins |
| 16:23.49 | *** join/#asterisk nettie (~nettie@stewie.freax.it) |
| 16:24.13 | *** join/#asterisk jetlag (jetlag@pool-173-61-207-68.cmdnnj.east.verizon.net) |
| 16:25.13 | nettie | Hi guys, anyone know how to set the msn of a P-t-P ISDN link please? I tried everything but the MSN sent is always the same.. any idea? thanx in advance |
| 16:25.41 | Naikrovek | kerry works for telephonydepot.com now |
| 16:25.55 | Naikrovek | or at least he's sending email on their behalf |
| 16:25.57 | Qwell | he moved on from 888voipstore? |
| 16:26.04 | Naikrovek | i think so, yes |
| 16:26.35 | nettie | and of course I'm using qozap |
| 16:26.48 | Naikrovek | btw, Qwell, my stomach is finally getting used to the sriracha sauce. soo good |
| 16:27.14 | Qwell | getting used to it? you clearly aren't using enough |
| 16:27.19 | ZeXr0 | Hahaha |
| 16:27.43 | Qwell | Naikrovek: use about 17x as much. |
| 16:28.07 | Naikrovek | well it always tasted lovely but far out it would make my stomach and various unmentionable parts of my digestive system revolt and attempt mutiny |
| 16:28.20 | Naikrovek | now it doesn't |
| 16:28.26 | Qwell | yeah. use more. |
| 16:28.30 | Naikrovek | i am |
| 16:28.37 | Qwell | MORE. |
| 16:28.40 | Naikrovek | lol |
| 16:28.52 | Naikrovek | had it on an omelet a few days ago. omg heaven |
| 16:30.58 | drmessano | When I want my system to revolt, I just sneak some dairy into something |
| 16:31.09 | Naikrovek | really, dairy? |
| 16:31.16 | Naikrovek | oh i'm so happy i don't have that problem |
| 16:31.19 | Qwell | he's hardcore |
| 16:31.19 | drmessano | No hot sauce can match a healthy dose of undigestable lactose |
| 16:31.20 | Naikrovek | loves cheese |
| 16:31.49 | drmessano | I love cheese too.. until 12 hours later |
| 16:32.32 | Qwell | I had a rather amusing experience with hot sauce once.. At a "mexican" restaurant here, we asked for tapatio. They had never heard of it. |
| 16:32.45 | Qwell | we look up, and they have a huge bottle of it. as decoration. |
| 16:32.50 | drmessano | ha |
| 16:33.17 | Qwell | it's still there too.. and they still don't offer it. |
| 16:33.35 | Naikrovek | nice |
| 16:33.52 | Naikrovek | that's like owning a car dealership, having Jaguars everywhere but only selling chevy |
| 16:33.56 | Naikrovek | "what's a jaguar" |
| 16:34.21 | Qwell | they asked "What's tapatio?" so we pointed to the big bottle of it. |
| 16:34.40 | coppice | Naikrovek: an Indian car |
| 16:35.01 | drmessano | That's probably one of those places you ask for something with a Tomatillo sauce and they tell you "no", only to bring out a tray of enchiladas with "green sauce" on them to next table |
| 16:35.24 | drmessano | "Wait, what?" |
| 16:36.33 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:37.02 | drmessano | Americanized Mexican Restaurants - employing clueless help since 1821 |
| 16:37.20 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:37.37 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:37.37 | DogBoy | ever notice how they have mexican cooks in chinese restaurants and chinese ones in mexican ones |
| 16:37.47 | drmessano | lol |
| 16:37.54 | drmessano | That's all over, huh? |
| 16:38.13 | DogBoy | pervasive |
| 16:38.19 | kn0x | ok thanks [TK]D-Fender .. sorry i was slightly confused |
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| 16:38.33 | *** join/#asterisk tzafrir_laptop (~tzafrir@109.65.52.161) |
| 16:38.35 | kn0x | tgif.. |
| 16:38.44 | *** join/#asterisk ideaman (~ihaveapla@c-174-52-20-94.hsd1.ut.comcast.net) |
| 16:39.06 | ideaman | So I have a verizon Mifi Card, and I want to use a wireless router to receive the signal, then hook up a few ip phones wired coming out of it. Is there any devices like that. Somewhat the opposite of a wireless router we know now? |
| 16:39.40 | drmessano | I had two apartments downstairs from mine with the chinese waiters/waitresses for the asian buffet place in one, and the mexican cooks for the same place in the next apartment. Both rented out by the owners. Both had about 7 or 8 people living in them |
| 16:40.10 | drmessano | ideaman: EVDO router |
| 16:40.25 | drmessano | Kyocera KR2 |
| 16:40.32 | drmessano | etc |
| 16:45.14 | ideaman | drmessano: just checking out the both the cradlepoint and the kyocera. So is there any way to get away from the wired USB into the cradlepoint, or is just how it has to be? |
| 16:46.07 | drmessano | Most devices are USB now |
| 16:46.11 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:46.21 | drmessano | The KR2 is an older box, that supports both |
| 16:48.05 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 16:49.25 | DogBoy | one of my favorite restaurant stories is when I was eating in this bowling alley (famous for having good food) and these six japanese girls came in and all six ordered the same identical meal |
| 16:49.51 | Qwell | that isn't the entire story I hope |
| 16:50.11 | DogBoy | it was like the scene in tampopo, something weird too like western barbeque t-bone steak |
| 16:50.23 | DogBoy | you have to know about japanese culture to get it I guess |
| 16:52.10 | Naikrovek | i guess |
| 16:52.11 | Naikrovek | :) |
| 16:53.08 | *** join/#asterisk unisonarthur (~adl505@38.104.201.82) |
| 16:53.19 | DogBoy | http://www.youtube.com/watch?v=PcMaZLiqVpI |
| 16:53.30 | DogBoy | that's the scene I refer to |
| 16:54.55 | coppice | Tampopo is a mass of references to the wild west |
| 16:55.06 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 16:56.18 | DogBoy | it is? |
| 16:57.49 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:58.05 | coppice | it was referred to as a spaghetti eastern :-) |
| 16:58.27 | REdOG | can I get an extension that executes a bash script? |
| 16:59.08 | Kobaz | TrySystem() |
| 16:59.26 | REdOG | ooh, tks |
| 16:59.58 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 17:08.15 | *** join/#asterisk DaveCanoe (~Dave@strike.eicat.ca) |
| 17:10.02 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:10.14 | kn0x | any reason why using originate from console gives me: handle_response_invite: Failed to authenticate on INVITE but the same peer works fine from Dial() in the dialplan? |
| 17:10.38 | Qwell | kn0x: not without context, no |
| 17:11.14 | kn0x | ok.... >originate SIP/18005551212@my-peer application Milliwatt |
| 17:11.19 | kn0x | ^ i get that |
| 17:11.33 | kn0x | but Dial(SIP/18005551212@my-peer) works fine |
| 17:11.50 | REdOG | that works and executes the command howerver my sip client gets a 603 call declined error |
| 17:19.03 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 17:19.07 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 17:19.14 | xheliox | Not really a problem, but maybe a bug? Or just my lack of understanding of how it's supposed to work. But if I use s,1 in a call file, it doesn't go to failed,1 -- but if I use 1,1 it does go to the failed extension. |
| 17:19.33 | xheliox | (if/when the call fails) |
| 17:20.32 | *** join/#asterisk Mango (~iMango@d154-20-93-153.bchsia.telus.net) |
| 17:21.09 | Mango | When I am dialing a SIP URI, is there a way to tell Asterisk that the peer is behind NAT, without creating it in sip.conf? |
| 17:21.13 | REdOG | ah, one must answer...lol duh |
| 17:24.57 | drmessano | Mango: Why would you need to do that? |
| 17:25.37 | Mango | Mainly because I'm ridiculous. |
| 17:25.56 | p3nguin_ | mango: If you call a SIP URI, it is the recipient's responsibility to have it configured accordingly. |
| 17:26.28 | Mango | That's what I thought...figured I'd double check. Thanks! |
| 17:26.41 | p3nguin_ | Think of it like picking up your phone and dialing a phone number. You've dialed it correctly, but there's no answer on the other side... |
| 17:26.52 | kn0x | Mango: it would be the [general] setting i believe |
| 17:26.53 | p3nguin_ | They don't have any phones plugged in! |
| 17:27.01 | p3nguin_ | So of course there's no answer. |
| 17:27.14 | p3nguin_ | But that's not your fault, and there's nothing you can do about it on your side. |
| 17:27.46 | kn0x | oh, the remote peer.. nevermind i misread that |
| 17:28.39 | Mango | I did try it though, just in case ;) |
| 17:31.37 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:32.40 | REdOG | I notice when I do sip show channels there are a few local ip addresses listed that have last message Rx: REGISTER ... where do these come from? |
| 17:33.11 | REdOG | I haven't set up those machines as channels... |
| 17:33.34 | *** join/#asterisk danonura (~danonura@unaffiliated/danonura) |
| 17:33.34 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 17:34.58 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 17:35.22 | seanjohn | NOTICE[7254]: chan_sip.c:17118 sip_poke_noanswer: Peer '2001' is now UNREACHABLE! Last qualify: 0 |
| 17:35.30 | seanjohn | qualify=2000 |
| 17:36.05 | p3nguin_ | yep |
| 17:36.24 | p3nguin_ | You could have also said "qualify=yes" there. |
| 17:36.39 | seanjohn | in sip.conf, all extensions have qualify=yes |
| 17:36.54 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 17:37.21 | seanjohn | the trunks have qualify=100 as they always report accurately |
| 17:37.51 | seanjohn | fender told me to put them at 2000 |
| 17:38.03 | seanjohn | the trunks are fine |
| 17:38.59 | grinder13 | hello! I keep getting some messages like "doing dnsmgr_lookup for '192.168.1.2'" in the CLI, but I have striclty disabled this funciontality in the dnsmgr.conf. how can I turn it off? it seems that this thing creates problem with SIP registrations for my clients. |
| 17:39.21 | seanjohn | dnssrv=no |
| 17:39.53 | grinder13 | in the dnsmgr.conf ? |
| 17:40.07 | seanjohn | your sip_general.conf |
| 17:40.19 | grinder13 | ok, let me check |
| 17:41.05 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:41.15 | seanjohn | grinder13: sorry, its http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup |
| 17:41.52 | grinder13 | yeah, that's what I saw in the sample sip.conf |
| 17:41.54 | grinder13 | thanx |
| 17:48.29 | grinder13 | ok, it seems it works now. thanx seajohn |
| 17:48.39 | grinder13 | going back to work :) |
| 17:56.14 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 17:58.45 | kn0x | the invites even look the same |
| 18:01.15 | seanjohn | p3nguin_: it use to say "options command not supported" for the peer that is failing qualify and it would ignore the response; now, it thinks the peer is unreachable |
| 18:01.41 | seanjohn | how can I fix this, other than putting qualify=no |
| 18:02.05 | p3nguin_ | qualify sends an OPTIONS to the peer. What more do you want? |
| 18:02.28 | seanjohn | i know this. The device isn't reponding to options |
| 18:03.04 | ChannelZ | Isn't responding or is responding with an error? |
| 18:04.25 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
| 18:07.09 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 18:07.29 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 18:07.41 | t_dot_zilla | can you mark packets w/IP Precedence or DSCP in asterisk ? |
| 18:15.25 | kn0x | where does the CLI originate get its callerid settings froM? |
| 18:16.58 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 18:17.30 | ChannelZ | the government |
| 18:18.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:18.56 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 18:19.24 | p3nguin_ | hehe |
| 18:20.21 | t_dot_zilla | can you mark packets w/IP Precedence or DSCP in asterisk? |
| 18:21.40 | kn0x | t_dot_zilla: see tos setting |
| 18:24.13 | seanjohn | ChannelZ: it use to respond with an error and Asterisk would count that as a valid qualify; now, it doesn't respond |
| 18:25.10 | seanjohn | with qualify=no it will ring the device but will not connect the call |
| 18:27.06 | seanjohn | asterisk 1.4 seems to have this long-standing bug |
| 18:28.11 | ChannelZ | is this device remote? |
| 18:28.38 | ChannelZ | it could just as well be a network issue |
| 18:28.45 | seanjohn | no its not remote |
| 18:28.53 | seanjohn | the remote devices are working fine lol |
| 18:29.13 | seanjohn | no change here and everything was fine yesterday |
| 18:38.10 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:40.21 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:40.40 | Slugs_ | Chainsaw, thank god your on. All my calls are being routed to a local fast food chain. They call back the caller id and and yell at me and say I owe them 30,000 dollars for wasting there time for loss of business |
| 18:40.46 | Slugs_ | HELP PLZ! |
| 18:41.27 | p3nguin_ | lol |
| 18:41.59 | Chainsaw | raises eyebrow |
| 18:42.32 | p3nguin_ | That's a good one! |
| 18:42.35 | p3nguin_ | tries |
| 18:42.49 | Slugs_ | hehe |
| 18:45.07 | coppice | don't trust fast food places. they might be cheetahs |
| 18:46.11 | Slugs_ | ChannelZ help! |
| 18:48.54 | Slugs_ | i ment ChannelZ, he knows how to fix these issues |
| 18:50.13 | *** join/#asterisk snayder (~douglas@189.8.197.70) |
| 18:52.24 | snayder | I need help with pjsip library, anybody uses this stack? |
| 18:53.34 | seanjohn | Now this is TRUELY a bug: chan_sip.c:17118 sip_poke_noanswer: Peer '4346074064' is now UNREACHABLE! Last qualify: 10 >> THE DEFAULT QUALIFY IS IN USE OF 2000 (2 seconds) |
| 18:54.34 | [TK]D-Fender | seanHow so? |
| 18:55.02 | seanjohn | Last Qualify: 10 and the device is registered |
| 18:55.36 | [TK]D-Fender | seanjohn: So? |
| 18:55.48 | p3nguin_ | The LAST qualify was 10. Now it is more than 2000. |
| 18:56.03 | [TK]D-Fender | seanjohn: Device regists. Device does respond to a bunch of qualify's. Then it FAILS TO. |
| 18:56.23 | [TK]D-Fender | seanjohn: Answered well beofre. DIDN'T ANSWER NOW |
| 18:56.34 | seanjohn | i hate linksys |
| 18:56.41 | [TK]D-Fender | seanHow is your device's failure to report back on time a BUG? |
| 18:57.07 | seanjohn | i read that it was a bug in 1.4 that hasn't been fixed, as of february |
| 18:57.37 | p3nguin_ | Show me the open bug report. I'm interested. |
| 18:57.56 | seanjohn | not a bug report, an article that described the report |
| 18:58.24 | [TK]D-Fender | seanjohn: I see nothing worthy of counting as "evidence" |
| 18:59.09 | ChannelZ | Slugs_: Do what now? |
| 18:59.48 | seanjohn | on g729, why can't we use inband? |
| 19:00.01 | ChannelZ | because the compression destroys the frequencies |
| 19:00.05 | Chainsaw | seanbright: Because there isn't sufficient audio bandwidth to accurately represent DTMF. |
| 19:00.28 | Chainsaw | seanbright: It is optimised to represent the human voice, which isn't a dual-frequency signal. |
| 19:00.36 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:00.46 | Qwell | Chainsaw: my voice is |
| 19:01.01 | Chainsaw | Qwell: That is an aftermarket upgrade. |
| 19:01.04 | Chainsaw | Qwell: I'm not counting those. |
| 19:01.05 | ChannelZ | If you're one of those tuvolian throat singers you're hosed (or whatver they are called) |
| 19:03.50 | ChannelZ | Tuvian. Whatever. |
| 19:05.39 | *** join/#asterisk REdOG (~REdOG@gentoo/user/redog) |
| 19:06.21 | REdOG | why is my * trying to register a sip channel on my UPS? |
| 19:07.49 | p3nguin_ | Show us the debug. |
| 19:09.15 | REdOG | nm, that's an external connection coming in with an ip reference the same as my internal network |
| 19:17.30 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 19:28.07 | *** join/#asterisk dennisG (~dennisG@2002:541e:88d0:0:213:2ff:fe56:e380) |
| 19:29.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:35.27 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 19:39.35 | seanjohn | Well, [TK]D-Fender, if anyone asks why asterisk is not qualifying their PAP2-NA's, tell them to only use line 2 on them. Line 1 refuses to register or, when both lines are in use, both of them become unreachable as they don't respond to qualify. |
| 19:39.54 | seanjohn | dumb linksys |
| 19:40.34 | seanjohn | i had to pull another one out of the box and ignore line 1 on the other one |
| 19:40.38 | [TK]D-Fender | seanjohn: Then don't use qualify and let them deal with their own NAT handling |
| 19:41.09 | seanjohn | there was no nat between this device |
| 19:41.31 | seanjohn | in-house device. the remote devices work perfect. 100 registered and reachabler |
| 19:41.36 | [TK]D-Fender | seanjohn: Then jsut forget qualify altogether |
| 19:42.04 | seanjohn | i tried doing qualify=no on the devices on these ATA's. The line rings but does not answer |
| 19:42.18 | seanjohn | it's the device's fault, like you said |
| 19:42.22 | [TK]D-Fender | sean got SIP debug etc to back it up? |
| 19:42.49 | seanjohn | i'm in asterisk -r now but, when I call one of the devices, it does not generate output |
| 19:43.05 | seanjohn | even though it follows the dialplan |
| 19:43.35 | seanjohn | everything else shows up in the manager |
| 19:43.56 | seanjohn | pap2= faulty ata |
| 19:44.18 | [TK]D-Fender | seanjohn: You should be seeing *'s packets... |
| 19:44.54 | seanjohn | i should also be seeing output in the manager when one of the pap2's makes or receives a call, other than using originate |
| 19:45.02 | seanjohn | lol |
| 19:45.24 | seanjohn | i fixed it but, for future reference, pap2's suck |
| 19:45.35 | seanjohn | I think I am going to try grandstream |
| 19:45.54 | seanjohn | how did I fix it? only use one line per device |
| 19:47.07 | seanjohn | 2002/2002 192.168.15.12 D N A 1027 OK (16 ms) 2002/2002 192.168.15.12 D N A 1027 OK (16 ms) 2001/2001 192.168.15.13 D N A 1027 OK (16 ms) 2001/2001 192.168.15.12 D N A 1027 OK (16 ms) |
| 19:47.33 | seanjohn | those are the two pap2's that were one pap2 with 2 lines |
| 19:47.54 | [TK]D-Fender | seanjohn: pastebin your peers for them |
| 19:48.33 | seanjohn | everything is going to show fine now fender. I have fixed it |
| 19:48.58 | seanjohn | that was the sip show peers output for those devices |
| 19:49.27 | seanjohn | i have qualify=yes on them and the default qualify=4000 |
| 19:49.43 | seanjohn | my trunks have qualify=100 and they ALWAYS report correctly |
| 19:51.38 | seanjohn | the great thing is, out of all the headache i've had, the people that pay me for voip have never had a problem. It's always problems with my devices inside the house. I'm guessing asterisk doesn't know what to do with 0 ms reponse time or the ATA's don't and they go nuts. |
| 19:57.22 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 20:06.35 | *** part/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 20:14.07 | drmessano | 0ms response time? |
| 20:14.19 | drmessano | Funny, I can get .750 ms |
| 20:14.23 | drmessano | But not zero |
| 20:14.31 | drmessano | Maybe my cables are too long |
| 20:19.19 | xheliox | Yeah. |
| 20:19.24 | xheliox | That's the problem. |
| 20:22.15 | [TK]D-Fender | we'll never know because like all people with "mysterious" problems he has shown the typical amount of back. Namely "Jack Shit" |
| 20:22.44 | [TK]D-Fender | "Oh the problem fixed itself" or "I dealt with it" usually = "I'm a fuck-off" :p |
| 20:22.56 | [TK]D-Fender | (usually) |
| 20:23.13 | Kobaz | [TK]D-Fender: it thought it was more like: i would love to help you with your mysterious problem, but all you've given me to work with is jack, and shit |
| 20:23.27 | *** part/#asterisk snayder (~douglas@189.8.197.70) |
| 20:25.39 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:25.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:25.51 | *** join/#asterisk imox1234 (~imox1234@p4FC5C514.dip0.t-ipconnect.de) |
| 20:26.25 | [TK]D-Fender | checkout time, BBIAB |
| 20:34.20 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 20:34.28 | *** join/#asterisk JerJer (~PhatJ@asterisk/original-h323-guy/JerJer) |
| 20:36.19 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 20:36.37 | JerJer | someone remind me which file i can copy into the source tree to preserve my configure selections from before |
| 20:38.32 | JerJer | grumbles about autoconf |
| 20:39.20 | russellb | you mean menuselect selections? |
| 20:39.23 | JerJer | yes |
| 20:39.26 | russellb | menuselect.makeopts |
| 20:39.35 | JerJer | thank you |
| 20:39.41 | russellb | np |
| 20:39.56 | russellb | hope all is well btw, long time no see |
| 20:41.16 | *** join/#asterisk mindCrime (~chatzilla@12.155.80.90) |
| 20:41.52 | JerJer | things are good - not really doing asterisk/voip any more |
| 20:44.17 | JerJer | i got my insurance license 2 weeks ago and am now doing insurance and bonds |
| 20:44.56 | JerJer | of course the phone system question came up.... I wasn't about to watch someone drop 20 grand on cisco / avaya BS |
| 20:46.55 | voxter | interesting how many people 'got out' of voip. too bad, its right at the most interesting time! |
| 20:47.14 | JerJer | ive been jacked around too much |
| 20:47.52 | voxter | hey by the way, any of you guys use polycom ip4000s? Ive noticed just the other day that the ip4000 vs ip6000, the ip4000 will mute/reduce speaker volume when the microphone is active and the ip6000 does not. I'm curious if thats a setting or by design |
| 20:48.04 | voxter | like a duplex issue or "Feature" |
| 20:48.15 | idespinner | VAD activity maybe? |
| 20:48.19 | idespinner | aastras do that |
| 20:48.49 | voxter | i checked, VAD is off. |
| 20:48.57 | voxter | at least in the polycom's sip.cfg |
| 20:50.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:51.16 | JerJer | russellb: ever play with the fax for asterisk free stuff ? all i ever get is errors... No Fax |
| 20:51.34 | JerJer | T.38 |
| 20:51.37 | russellb | nah, i haven't worked on it, others here have |
| 20:51.40 | russellb | it's supposed to work, heh |
| 20:51.48 | JerJer | who should i pester? |
| 20:52.18 | JerJer | i'm hoping the new 1.6 has some 'fixes' |
| 20:52.29 | russellb | you could try the list for some help maybe |
| 20:52.34 | russellb | otherwise, you can get help with a paid license |
| 20:52.36 | russellb | that's about it |
| 20:52.40 | russellb | T.38 is a pain |
| 20:52.56 | JerJer | yeah, we might have to cough up for a paid license |
| 20:53.29 | JerJer | we also have a lame SPA9000 - its the only thing i had laying around that supported T.38 |
| 20:54.54 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 20:55.11 | JerJer | was pondering doing a hylafax hook, but not sure if that would jive with res_fax_digium |
| 20:55.24 | JerJer | wish this crap was easier :) |
| 20:55.55 | JerJer | hell we might pay for e-fax.... heh |
| 20:56.19 | voxter | I do all efax with hylafax |
| 20:56.26 | voxter | its ... it works.. :P |
| 20:56.33 | voxter | its kind of a pain, but it works. |
| 20:56.39 | JerJer | efax as in efax.com or your own roll ? |
| 20:57.05 | JerJer | what really sucks is i have a box full of Brooktrout and ECON DIVA T-1 fax cards, but no PRI |
| 20:57.07 | voxter | our own roll |
| 20:57.25 | JerJer | voxter: how do you interface with a phone line ? |
| 20:57.35 | voxter | JerJer: iaxmodem + spandsp |
| 20:57.36 | JerJer | or just 'pstn' |
| 20:57.47 | voxter | JerJer: oh, wait, do you mean a phone line as in pstn or as in a fax machine? |
| 20:57.48 | JerJer | yeah - i had that working back in the day |
| 20:58.15 | voxter | JerJer: we get pstn via sip (or pri, no difference) then it goes to an iaxmodem then to hylafax on /dev/ttyIAX<xx> |
| 20:58.18 | JerJer | the 'voip provider' we are using supports T.38 on SIP |
| 20:58.25 | voxter | yeah, mine doesnt. |
| 20:58.31 | voxter | I send it to the pstn using g.711 |
| 20:58.43 | JerJer | and its 'reliable'? |
| 20:58.49 | JerJer | i've always had trouble |
| 20:59.07 | JerJer | (with fax over 711 in general) |
| 20:59.31 | Corydon76-dig | I find that it's much more reliable if you use 10ms packets |
| 21:00.00 | Corydon76-dig | it cuts the latency to the point where faxes generally work all the time |
| 21:00.23 | JerJer | hmm - now is there that setting in sipura devices |
| 21:00.27 | JerJer | looks |
| 21:01.09 | JerJer | RTP packet size? 0.030 ? |
| 21:01.24 | Corydon76-dig | Right, knock that down to 0.010 |
| 21:01.27 | voxter | JerJer: yes its reliable as long as the g711 link to the pstn is "clean" |
| 21:01.59 | voxter | JerJer: i also found that its reliable from something like a PAP2 with a fax machine plugged into it as long as you disable silence suppression and VAD |
| 21:02.11 | JerJer | and echo can |
| 21:02.16 | voxter | yes, and disable echo can |
| 21:02.18 | JerJer | been there :) |
| 21:02.18 | voxter | absolutely |
| 21:02.45 | voxter | t.38 would be nice, but those tweaks and g711 are reliable enough (99.9%) to not force me to start the headache of t.38 |
| 21:12.00 | Corydon76-dig | btw, a clarification to t.38 is working its way to standardization... |
| 21:12.47 | Corydon76-dig | which should make various vendor implementations more interoperable |
| 21:14.17 | Kobaz | sounds good |
| 21:17.18 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
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| 21:31.04 | Joe_CoT | hi guys. make menuconfig won't let me select chan_dahdi. It looks like what it's failing on is tonezone. What is tonezone, how do i get it? |
| 21:31.04 | Qwell | Joe_CoT: dahdi |
| 21:31.46 | Joe_CoT | Qwell, I have dahdi, i have it loaded, I have the source files. Meetme is willing to build (it depends on dahdi), chan_dahdi won't. |
| 21:32.24 | Qwell | What part of dahdi did you install, exactly? |
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| 21:33.39 | Joe_CoT | dahdi, dahdi-linux, dahdi-source |
| 21:33.54 | Qwell | You need dahdi-tools. |
| 21:34.40 | Joe_CoT | looks like I needed libtonezone-dev. found it, thanks |
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| 21:47.13 | [sr] | hi WIMPy |
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| 22:07.06 | Mythmon | I inherited the OSU Open Source Lab ftp mirrors recently, and I noticed that although we have a mirror for the asterisk project, it's automatic rsync updates have been failing for quite some time, and your site doesn't seem to link to any external mirrors any more. |
| 22:07.43 | Mythmon | I just wanted to confirm that our mirror services were no longer being used, and then I was going to delete it from our server to free up space for other projects. |
| 22:08.20 | pabelanger | Mythmon: best to send an email to asterisk-dev mailing list |
| 22:08.52 | Mythmon | pabelanger: ok, i will do that. |
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| 22:38.17 | JamesHarrison | I'm having troubles with a DID on a SIP trunk tying into Asterisk, http://pastie.org/private/tf55txb0bybj8x0ljnaqg has my configs and my sip debug from a test incoming call |
| 22:40.10 | drmessano | JamesHarrison: Look at your contexts |
| 22:40.36 | drmessano | You have incoming but no incoming-pstn, which is referenced in sip.conf |
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| 22:44.39 | JamesHarrison | drmessano: Fixed that, oops, but still no joy; still get the 401 and Ignoring this INVITE request |
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| 22:48.03 | drmessano | JamesHarrison: and insecure= should be port,invite for 1.6.2.x |
| 22:48.12 | drmessano | insecure=port,invite |
| 22:48.17 | drmessano | very doesn't exist |
| 22:50.15 | JamesHarrison | drmessano: bingo, that did the trick... very was the option provided in my supplier's docs, evidently out of date :) |
| 22:50.19 | JamesHarrison | drmessano: cheers for the help! |
| 22:50.52 | drmessano | No probs |
| 22:51.01 | DogBoy | I think it tells you that in the console too |
| 22:51.26 | DogBoy | something like: yo wassup... very has been deprecated |
| 22:51.48 | drmessano | But that's not the case in 1.6.2.x |
| 22:51.54 | drmessano | It was deprecated for 1.4 |
| 22:52.05 | drmessano | It's been disemboweled in 1.6.x |
| 22:52.05 | JamesHarrison | Heh, ouch, well out of date :) |
| 22:52.47 | drmessano | People still referencing 1.2 should be shot |
| 22:52.53 | drmessano | and hung |
| 22:53.01 | drmessano | Hung, then shot, then hung some more |
| 22:53.02 | Qwell | drmessano: You just referenced it. |
| 22:53.11 | drmessano | ~shoot drmessano |
| 22:53.12 | infobot | ACTION shoots drmessano in the head with a spitwad! |
| 22:53.20 | drmessano | ~hang drmessano |
| 22:53.21 | infobot | ACTION grabs drmessano by the neck, slips a noose around his neck and then leads him to the tallest tree around |
| 22:53.25 | drmessano | ~shoot drmessano |
| 22:53.26 | infobot | ACTION shoots drmessano in the head with a frozen turkey cannon! |
| 22:53.30 | drmessano | Done. |
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| 22:54.23 | drmessano | 1.2 should only be used with rotary phones and if you smell really bad |
| 22:54.28 | DogBoy | yea, all it says is: set_insecure_flags: Unknown insecure mode 'very' on line 31 |
| 22:54.59 | shido6 | what do you guys use to monitor your asterisk systems? |
| 22:55.08 | drmessano | shido6: End users |
| 23:06.50 | JamesHarrison | shido6: pretty sure I've seen some munin scripts floating around, as for the actual process monitoring I'm using monit |
| 23:07.17 | shido6 | thats what Ive been reading through for the last 30 minutes |
| 23:07.23 | shido6 | and saw the iphone app, too |
| 23:07.26 | Jumpie | anybody hav experience using ldf/csf i conjunction with iptables? |
| 23:07.27 | shido6 | M/Monit |
| 23:07.33 | Jumpie | er lfd |
| 23:07.35 | shido6 | i think i'll give it a try |
| 23:07.40 | shido6 | does it cost anything? |
| 23:10.28 | drmessano | shido6: Monit, not M/Monit |
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| 23:13.59 | ChannelZ | JamesHarrison: Vitelity? |
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| 23:14.28 | JamesHarrison | ChannelZ: What, my ISP provider? No, siptrunk.co.uk |
| 23:14.51 | ChannelZ | Ah. JUst wondering cuz Vitelity's docs are similarly out-of-date :) |
| 23:15.03 | ChannelZ | Didn't realize you were in UK |
| 23:15.39 | JamesHarrison | Hehe, fair enough :) |
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| 23:17.06 | JamesHarrison | siptrunk seem to be doing just fine with the exception of some terrible documentation, seems to be quite a lack of SIP trunk/DID providers these days |
| 23:17.12 | JamesHarrison | (In the UK at least) |
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| 23:31.18 | ChannelZ | hmm |
| 23:35.34 | grinder13 | hello! I 've setup a SIP trunk like it is described here: http://lists.digium.com/pipermail/asterisk-users/2010-May/248437.html and http://lists.digium.com/pipermail/asterisk-users/2010-May/248441.html The problem is that I am getting the well known "482 loop detected" error. any hints? |
| 23:47.10 | ChannelZ | without seeing your dialplan configs, it sounds like you've probably created a loop then.... |
| 23:48.35 | grinder13 | i 've done it like it is described in that urls, but give me a minute to do a proper job and copy to pb |
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| 23:53.18 | ChannelZ | well I'm not going to read pages of threads and then guess about what you did differently. My guess is you have a context messed up and calls are going into the same context they are trying to leave out of, thus looping |
| 23:54.31 | grinder13 | here you go. Server A: http://pastebin.com/9Xqqf5Q1 , Server B: http://pastebin.com/2tYU1N2Z |
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| 23:56.18 | root52 | Hey all. I am trying to get dahdi to start. I have done this many time before but this time it is odd. When I try to start dahdi it fails with a message about how it failed because of an invalid module format. I look at dmesg and I see this line... dahdi: disagrees about version of symbol module_layout I am not a kernel expert but i may have to be real quick ;-) any thoughts? |