| 00:00.24 | empire667 | Juggie: I'm going to start the machine |
| 00:00.55 | Juggie | empire, nope. |
| 00:01.01 | Juggie | i did nothing with the kernel |
| 00:01.05 | Juggie | other then make one symbolic link |
| 00:02.41 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 00:04.07 | zimdog | Is their a sip command in the cli to show logged in users? |
| 00:05.00 | zimdog | Also a way to just debug a specific sip extension? |
| 00:05.27 | Juggie | 'sip show peers' |
| 00:06.03 | Juggie | 'sip debug ip' |
| 00:06.35 | zimdog | Juggie: sip show peers seems to show users who have not successfully logged in |
| 00:07.51 | zimdog | never mind I guess if they don't have an ip they are not logged in |
| 00:07.57 | zimdog | thanks |
| 00:07.57 | Juggie | yah, when they have an ip and port they are logged |
| 00:10.56 | |Vulture| | and if you turn qualify on you can see their latency... |
| 00:11.07 | zimdog | Juggie: can I log this sip debug IP so I can look at it before it scrolls off the screen |
| 00:11.21 | |Vulture| | ./var/log/asterisk/full |
| 00:11.41 | zimdog | I see a sip history |
| 00:11.56 | *** join/#asterisk sandjunkie (~trilluser@66-55-197-254.gwi.net) |
| 00:12.05 | Juggie | zimdog, use a scroll back buffer? |
| 00:12.21 | Juggie | vul, do u know how i can make my IAX commands (with passwords) not show up in the CDR? |
| 00:12.49 | zimdog | I am but doesn't store enough guess I can crank it up higher |
| 00:12.50 | |Vulture| | Juggie: I pass all mine to another extension, and use global variables |
| 00:12.57 | |Vulture| | globals will protect them |
| 00:12.59 | jerlique | hi - I'm trying to oget agent logins working. I can authenticate to * and then it plays hold music to me. If I hangup, I get logged off as an agent |
| 00:13.12 | *** join/#asterisk dr342346 (~ty@12-202-50-12.client.insightBB.com) |
| 00:13.31 | dr342346 | OK seriously I need a fully functional IAX softphone program that is not glitchy any suggestions |
| 00:13.40 | dr342346 | I have tried IAXcomm too glitchy |
| 00:13.52 | dr342346 | and other i have tried |
| 00:13.52 | brc_ | ~diax |
| 00:13.58 | |Vulture| | hardphones are just so sexy |
| 00:13.59 | brc_ | ~google diax |
| 00:14.08 | dr342346 | does it have Hold / transfer and all those good buttons |
| 00:14.11 | brc_ | dr342346, *FREE* softphones are *CRAP* |
| 00:14.16 | brc_ | no it does not |
| 00:14.50 | dr342346 | also diax seem to be in german |
| 00:14.51 | dr342346 | or such |
| 00:14.58 | dr342346 | or russian |
| 00:15.16 | zimdog | How hard is it to install * as a user in a home directory? |
| 00:15.18 | brc_ | you can a. write your own. b. pay somebody to write it for you. or c. buy a commercial softphone *AND*HIGH*QUALITY*soundcard and headset |
| 00:15.25 | brc_ | zigman, have you even looked at the Makefile? |
| 00:15.30 | dr342346 | I just really want to use the builtin IAX and not go w/ sip since there seem to be nat problems occationally and IAX has good tranveral |
| 00:16.02 | brc_ | diax is very usable in english |
| 00:16.06 | zimdog | brc_: yes Do I just have to change the installer paths? |
| 00:16.23 | brc_ | yup |
| 00:16.25 | brc_ | afaik |
| 00:16.32 | dr342346 | interesting |
| 00:16.37 | dr342346 | i must have gotten the wrong version |
| 00:16.41 | sandnigg0r | can you get a used pbx phone or one from radio shack to work with hardware supported by asterisk? |
| 00:16.42 | zimdog | ok just asking cause didn't see anything in the wiki etc |
| 00:16.45 | brc_ | nothing beats eyeBeam on features though |
| 00:16.48 | sandnigg0r | for cheep? |
| 00:16.59 | brc_ | chicks cheep |
| 00:17.03 | dr342346 | so you say free softphone are there any non-free ones that are good that have all the little features that asterisk supports |
| 00:17.07 | iMediax | What codec is this or where the heck is it getting this 63488 from? "No translator path exists for channel type IAX2 (native 63488)" |
| 00:17.15 | dr342346 | is eyebeam IAX |
| 00:17.20 | brc_ | dr342346, not for IAX |
| 00:17.36 | dr342346 | wow ... so should i just go w/ SIP |
| 00:17.41 | brc_ | SIP can work fine over nat when done properly |
| 00:17.44 | brc_ | *heh*heh*heh |
| 00:17.45 | dr342346 | I use iLBC codec |
| 00:18.40 | brc_ | sandnigg0r, a analog phone can be interfaced to asterisk with the proper hardware yes |
| 00:18.46 | dr342346 | so if there are no good softphones with IAX and there is like 1 hard phone why would people use IAX |
| 00:19.11 | brc_ | sandnigg0r, it's very unlikely a "used pbx phone" will work though... |
| 00:19.16 | sandnigg0r | brc_, i want to hook up a normal pbx phone. how cheep could i get one used or at radio shack? |
| 00:19.21 | sandnigg0r | really? |
| 00:19.23 | Luhiwu | i'm working on a windows iax softphone based on the iaxclientocx |
| 00:19.23 | sandnigg0r | that sucks |
| 00:19.33 | brc_ | dr342346, just look at the name...InterAsteriskExchange |
| 00:19.55 | Strom_TM | sandnigg0r, use either a standard analog telephone set with an ATA or a Zap card, or use a SIP phone |
| 00:20.33 | brc_ | sandnigg0r, if you are looking for a pbx on the cheAp your best bet is to look on ebay for a used system...you can get em quite cheAp |
| 00:20.41 | sandnigg0r | Strom_TM, if i do that will it be voip? or analog? |
| 00:20.50 | brc_ | uh |
| 00:20.55 | brc_ | that don't make no sense at all |
| 00:20.55 | jerlique | Why does an agent get put on hold after loggin? |
| 00:21.04 | brc_ | jerlique, simply the way it works. |
| 00:21.18 | brc_ | jerlique, at asterisk console show application agentcallbacklogin |
| 00:21.22 | *** join/#asterisk wwalker_ (~wwalker@wwalker.sustaining.supporter.pdpc) |
| 00:21.52 | Strom_TM | sandnigg0r, with an analog phone, analog until it gets digitized, either at the ATA or in the zap card |
| 00:21.55 | sandnigg0r | brc_, i mean if i use analog phones. When i call one phone from the other will it convert the audio and send it to the other phone or what? |
| 00:22.01 | jerlique | so does this mean that the agent will be called by * when an applicable comes in? |
| 00:22.06 | Strom_TM | with a sip phone it gets digitized inside the phone |
| 00:22.07 | brc_ | sandnigg0r, yes |
| 00:22.17 | sandnigg0r | Strom_TM, ok thats what i was wondering |
| 00:22.24 | brc_ | jerlique, please go read the documentation I gave you |
| 00:22.32 | jerlique | I am now thanks :) |
| 00:22.34 | sandnigg0r | brc_, thanks |
| 00:22.39 | sandnigg0r | Strom_TM, thanks |
| 00:22.47 | brc_ | if you've got any questions after that leme know though |
| 00:22.49 | Strom_TM | sandnigg0r, even digital phones have to do analog-digital conversion before the transmitter and receiver will send and receive audio |
| 00:22.53 | sandnigg0r | Strom_TM, thats the cheepest way to set up asterisk? |
| 00:23.11 | Strom_TM | depends on what you want to do with it |
| 00:23.18 | sandnigg0r | do you know a good zap or ata card that is supported by asterisk |
| 00:23.27 | Strom_TM | um |
| 00:23.32 | sandnigg0r | all i want to do is set up 2 phones on my home network |
| 00:23.37 | dr342346 | I guess ill use SJPhone w/ SIp |
| 00:23.40 | sandnigg0r | to learn asterisk |
| 00:23.40 | dr342346 | it seems to be working |
| 00:23.43 | *** join/#asterisk Nukemizer (~Nuke@65.103.231.133) |
| 00:23.49 | dr342346 | are there any better SIP softphones |
| 00:23.54 | sandnigg0r | its just a development network |
| 00:23.59 | dr342346 | or any recommendations other than SJphone |
| 00:24.05 | sandnigg0r | lab asterisk pbx rather |
| 00:24.10 | sandnigg0r | so i can learn |
| 00:24.25 | brc_ | cheep n. A faint, shrill sound like that of a young bird; a chirp. cheap adj. 1. a. Relatively low in cost; inexpensive or comparatively inexpensive. |
| 00:24.26 | Chuji | ~softphones |
| 00:24.36 | Chuji | ~softphone |
| 00:24.37 | jbot | something that should be drug out into the street and shot |
| 00:24.39 | brc_ | dr342346, yes, eyeBeam from xten.com |
| 00:24.40 | sandnigg0r | hummm |
| 00:24.51 | |Vulture| | wow... Im deff not helping a guy with a name like that |
| 00:24.54 | sandnigg0r | to get started i can set up asterisk with softphones |
| 00:25.09 | sandnigg0r | |Vulture|, its liget |
| 00:25.20 | |Vulture| | how so? |
| 00:25.23 | sandnigg0r | |Vulture|, it only offends white people |
| 00:25.32 | sandnigg0r | middle easterns love me |
| 00:25.34 | ManxPower | Unfortunatly SoftPhones must rely on the underlying OS as well as the underlying sound card and drivers. This can cause extreme pain. |
| 00:25.36 | |Vulture| | I am middle eastern |
| 00:25.44 | sandnigg0r | yeah |
| 00:25.48 | sandnigg0r | i should do the analog way |
| 00:25.54 | sandnigg0r | thanks for the good infomaction |
| 00:26.00 | sandnigg0r | information* |
| 00:26.13 | ManxPower | Or a cheap IP phone or a cheap ATA |
| 00:26.58 | dr342346 | thanks downloading now |
| 00:27.08 | zimdog | I am getting this message. SIP/2.0 484 Address Incomplete would this be a configuration problem in the xlite phone or in the isp conf? |
| 00:27.11 | dr342346 | can you intereface this eyeBeam w/ video over SIP |
| 00:27.51 | |Vulture| | sandnigg0r: taajrt? |
| 00:28.07 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-6-3.d4.club-internet.fr) |
| 00:30.52 | brc_ | howdy |Vulture| |
| 00:31.27 | |Vulture| | sup brc_ |
| 00:31.41 | brc_ | C# |
| 00:32.06 | brc_ | it's not half bad |
| 00:32.15 | implicit | its all bad |
| 00:32.17 | sandnigg0r | i love it |
| 00:32.24 | implicit | i know |
| 00:32.35 | sandnigg0r | i been coding in c# from when vs.net was in beta |
| 00:32.46 | brc_ | VisualStudio v8 is fairly nice |
| 00:33.22 | implicit | brc_: if you like ugly bloated pieces of shit that force you to run the most viral operating system ever invented, you may be going crazy enough to 'think' it is fairly nice |
| 00:34.04 | brc_ | ~seen blitzrage |
| 00:34.05 | jbot | blitzrage is currently on #asterisk. Has said a total of 53 messages. Is idling for 3h 49m 57s |
| 00:34.33 | Juggie | ~seen Juggie |
| 00:34.36 | jbot | juggie is currently on #asterisk (1d 8h 45m 20s). Has said a total of 234 messages. Is idling for 3s |
| 00:34.44 | sandnigg0r | brc_, woodbiy or something like that? |
| 00:34.52 | |Vulture| | looking for yourself again Juggie? |
| 00:34.57 | sandnigg0r | would-be |
| 00:35.00 | brc_ | haha |
| 00:35.21 | Juggie | wanted to see how many messages |
| 00:35.45 | |Vulture| | ~seen |Vulture| |
| 00:35.46 | jbot | |vulture| is currently on #asterisk (3h 6m 32s). Has said a total of 33 messages. Is idling for 1s |
| 00:36.09 | |Vulture| | you win |
| 00:36.43 | Juggie | i was helping some guy last night with doing click to talk kinda functionality. |
| 00:36.47 | Juggie | thats why :) |
| 00:37.12 | Juggie | someone had written a patch for Dial() to allow you to do commands after the dial on the second line, but before the bridge |
| 00:37.27 | Juggie | so i cursed them out, and told him how to do it with Dial & a Macro |
| 00:37.31 | Juggie | without a patch |
| 00:38.10 | Juggie | eg if u wanted to give the person who answered the second call a message say with a ticket number, and then wait for a # to bridge the call. |
| 00:39.23 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 00:39.38 | ctooley | JerJer, are you around? |
| 00:39.43 | |Vulture| | ah gotchya |
| 00:39.54 | Juggie | if you do Dial with Mm |
| 00:40.00 | Juggie | you can run music on hold on the original caller |
| 00:40.07 | Juggie | and the second caller will run through the macro |
| 00:40.18 | Juggie | and they dont bridge until the macro is complete. |
| 00:40.58 | JerJer | ctooley: nope |
| 00:40.58 | jskcr | hy all |
| 00:41.03 | jskcr | lol |
| 00:41.10 | PoWeRKiLL | Evening :) |
| 00:41.21 | jskcr | Jer Jer, you need to setup a faq site for nuphone newbies. |
| 00:41.41 | jskcr | err nufone sorry lol |
| 00:41.44 | JerJer | are you volunteering to write it? |
| 00:41.51 | |Vulture| | isn't that what the wiki is for? |
| 00:41.56 | jskcr | yea, for a 800 account sure |
| 00:42.18 | jskcr | 1000, minutes a month Ill write a wiki and talk about your service on the radio show I do |
| 00:42.45 | Juggie | how hard can it be |
| 00:42.46 | implicit | jskcr: you sure sell yourself cheap |
| 00:42.49 | Juggie | its just a IAX link isnt it |
| 00:43.06 | implicit | jskcr: something you could get $20 retail to do all that work |
| 00:43.08 | empire667 | juggie: can you please check on your machine is you have /etc/dev/ttyI0 or /dev/ttyI0 |
| 00:43.13 | |Vulture| | the radio adv. is more worthy depending how how many listeners |
| 00:43.37 | empire667 | is ==if |
| 00:43.48 | jskcr | Quite a bit actually. |
| 00:44.15 | Juggie | empire, no i do not have that. |
| 00:44.26 | empire667 | ok thanks |
| 00:44.32 | jskcr | I would write a wiki, just to see a stop to the stupid questions that have nothing to do with asterisk but still require some looking up. |
| 00:44.58 | blitzrage | brc_: yo |
| 00:45.08 | |Vulture| | jskcr: you mean like, how do I install *? |
| 00:45.23 | |Vulture| | actually how do I installl FC3 |
| 00:45.26 | blitzrage | jskcr: send me FAQ questions and answers for the FAQ section on asteriskdocs.org |
| 00:45.50 | jskcr | No, more like how to setup things the the speex codec and the recommended settings. |
| 00:45.53 | ctooley | Ok, so if I need to answer a lot of SIP calls from a SIP Termination provider. more than one Asterisk server can handle. But all fo the calls will come in on one SIP account |
| 00:45.54 | ManxPower | blitzrage, What's your e-mail address? I have an imap folder called Asterisk-FAQ |
| 00:46.11 | blitzrage | ManxPower: leif at leifmadsen dot com |
| 00:46.40 | ManxPower | Do you really want me to bounce 83 messages to you? |
| 00:47.03 | ctooley | so I need some kind of Proxy, like SER, that can register the SIP account and hand off the work to the Asterisk boxes |
| 00:47.04 | jskcr | blitzrage: when you guys update the cvs on asteriskdocs.org, I will |
| 00:47.46 | ManxPower | blitzrage, looks like I can attach them all to one message |
| 00:47.54 | blitzrage | jskcr: a bunch of new stuff just got committed today |
| 00:47.59 | blitzrage | jskcr: and thats no excuse |
| 00:48.11 | *** join/#asterisk porkchop (~porkchop@porkchop.nat.cccp.porkchop.net) |
| 00:48.14 | jskcr | blitzrage: heheh, Ill d/l the cvs and write a speex codec setup for ya :) |
| 00:48.14 | blitzrage | ManxPower: perfect |
| 00:48.20 | blitzrage | jskcr: please do! |
| 00:48.33 | ManxPower | blitzrage, on it's way |
| 00:48.41 | blitzrage | ManxPower: perfect |
| 00:48.51 | jskcr | poker cool. |
| 00:48.57 | blitzrage | ManxPower: I'll add the FAQ stuff before you docs... which I have to figure out what to do with... |
| 00:49.03 | blitzrage | not sure where to put it on the site... |
| 00:49.15 | blitzrage | aiight, I'm out |
| 00:49.19 | brc_ | blitzrage, |
| 00:49.21 | brc_ | howdy |
| 00:49.23 | brc_ | bye |
| 00:49.47 | Juggie | does asterisk work well with any T1/E1 hardware other then digium? |
| 00:49.53 | brc_ | no |
| 00:49.57 | ManxPower | blitzrage, LOL! There's not THAT much stuff there. Most of it is waves or codecs or AGI |
| 00:50.09 | jskcr | Juggie: Ive heard a few horror stories and fewer sucess stories. |
| 00:51.00 | Juggie | i ask because i had a meeting with the lawyers at work about using asterisk, and the problem is, we coudnt RFP for cards usable with Asterisk because there are only one... the laywer mentioned that could cause some problems related to some trade laws. |
| 00:51.03 | brc_ | if you must you can still use tormenta 2 cards...but the digium te cards are a much newer design |
| 00:51.17 | brc_ | Juggie, ahh |
| 00:51.24 | brc_ | in that case, yes |
| 00:51.31 | Juggie | companies could bid challenge apparentally |
| 00:51.37 | Juggie | because we are keeping them out of the RFP |
| 00:51.44 | brc_ | your pm's |
| 00:51.52 | ManxPower | blitzrage, sending again. My mailserver thought the attachments were named .com and so would not let me send them. |
| 00:52.06 | zimdog | will a sip shown work through dialup just to test nat settings etc? I understand the audio would be bad quality? |
| 00:52.23 | Juggie | sip would work with gsm on dialup probally |
| 00:52.30 | talkwebhosts | keeps saying the number is busy |
| 00:52.38 | brc_ | zigman, TOTALLY depends on what codec you use |
| 00:52.47 | brc_ | sip just sets the session up... |
| 00:52.48 | talkwebhosts | where should i look to fix this? |
| 00:53.00 | brc_ | talkwebhosts, how about some more info? |
| 00:53.16 | talkwebhosts | i configured the sip file broadvoice |
| 00:53.24 | jskcr | blitzrage: On the speex stuff, I should just keep it in the same xml format as the cvs? |
| 00:53.24 | talkwebhosts | and restarted asterisk |
| 00:53.25 | brc_ | zigman, look at the speex codec...you'll have to tune it for low bandwidth |
| 00:53.30 | talkwebhosts | and keeps saying number is busy |
| 00:53.33 | talkwebhosts | when I try to call it |
| 00:53.38 | brc_ | jskcr, yeah, it's docbook format |
| 00:53.42 | jskcr | cool |
| 00:53.56 | jskcr | zigman, if ya want the setup for speex just /msg me |
| 00:54.47 | talkwebhosts | when i reload the config files |
| 00:54.55 | zimdog | Juggie: Think maybe I am having a nat problem. I have xten connected throguh dialup. have opened ports udp 5060 and 10000-20000 on the firewall. I can call an extension from the xlite phone on the ouside and the extension rings but no audio either way. If I call the xlite phone it does nothing but a fast busy |
| 00:54.58 | talkwebhosts | i got this : Unable to lookup sip.broadvoice.com |
| 00:55.12 | talkwebhosts | NO such host sip.broadvoice.com |
| 00:55.32 | jskcr | zimdog. thats a udp transport problem |
| 00:55.42 | jskcr | try setting up the host in a dmz |
| 00:56.08 | jskcr | then firewall the host so only the providers have access |
| 00:56.57 | zimdog | jskr: just playing right now on a home setup so don't really have that option |
| 00:57.13 | jskcr | ahh, what kind of router are you using? |
| 00:57.30 | zimdog | jskr: I do have a dedicated server though and was thinking about installing it ouside the fireall and then tunneling in with iax |
| 00:57.35 | zimdog | freesco |
| 00:57.59 | jskcr | zimdog, thats one of the reason I love iax, you only have one port to worry about. |
| 00:58.44 | talkwebhosts | any idea why the number is busy? |
| 00:59.34 | zimdog | jskr: I was asking earlier how hard it would be to run * self contained so I could install as a user on dedicated server. |
| 01:00.01 | jskcr | zimdog, you would need to setup it in a chroot envoirment it is not too hard |
| 01:00.15 | jskcr | I run mine as asterisk user |
| 01:00.33 | zimdog | jskr: is there any docs on doing this? |
| 01:01.52 | jskcr | zimdog, theres is a script that comes with amp that sets the permissions to use a asterisk user and group. |
| 01:02.25 | brc_ | ~seen atacomm |
| 01:02.26 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 2d 23h 5m 33s ago, saying: 'anyone want a IP 3000 conference phone? looking to replace ours with a IP 4000 model. Barely been used, in great condition.... looking for around $500'. |
| 01:02.33 | jskcr | basicly its just creating a asterisk user and group and chowning the /usr/lib/asterisk and /var/log/asterisk and /etc/asterisk directories. |
| 01:03.17 | jskcr | It does not require ports that need root IE < 1024. |
| 01:03.18 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 01:03.49 | jskcr | err dammit > 1024 |
| 01:04.04 | zimdog | jskcr: Ok and I have not done a chroot environment before. But this would be a standard unix configuration like doing a ftp chroot? |
| 01:04.35 | jskcr | you dont really need to setup a chroot enviorment. |
| 01:04.46 | jskcr | I would recommend just using a regular user. |
| 01:05.07 | zimdog | ok I would be safe just running as a normal user. |
| 01:05.19 | jskcr | IE create a asterisk user, if you are unfamiliar with setting up a chroot . |
| 01:06.15 | zimdog | I guess that would work. * install only in a few directories correct |
| 01:06.16 | jskcr | zimdog, as long server is firewalled you should not have a problem IE allow only you external phones and sip providers then if you do need external public access setup a pptp vpn. |
| 01:06.49 | jskcr | or ipsec if your a paranoid.\ |
| 01:07.22 | zimdog | The dedicated server I have is outside of a firewall I believe. |
| 01:07.35 | zimdog | I use it for webserver currently |
| 01:08.08 | zimdog | I was thinking of running * on it and then IAX into my home office to get pstn from my zaptel |
| 01:08.41 | jskcr | Thats a good idea, since iax only uses on port you can easily setup the vpn rules for it. |
| 01:08.58 | jskcr | err s/vpn/firewall sorry |
| 01:09.05 | jskcr | The red wine is kicking in :P |
| 01:09.18 | zimdog | Ok when I install * on this server would I still need to download zaptel and libpri just for sip and IAX? |
| 01:09.41 | ManxPower | zimdog, not unlesss you need ztdummy for IAX2 trunking or MeetMe |
| 01:10.19 | hmmhesays | is IAX2 trunking still having issue's? |
| 01:10.23 | jskcr | Ya ya only need the ztdummy for music on hold or the conferencing |
| 01:10.41 | zimdog | Well of course I want to be able to play with as meany features as possible. So I need zaptel for sure. What is libpri needed for |
| 01:10.47 | Poincare | how do i take a second line with a grandstream? |
| 01:11.18 | hmmhesays | what do you mean |
| 01:11.23 | *** join/#asterisk liquidno2 (~foo@cs68201147-69.sw.rr.com) |
| 01:11.27 | Poincare | well, if there is a second call |
| 01:11.43 | Poincare | how do i switch from the existing call to the new incoming call? |
| 01:11.48 | hmmhesays | just a guess, cause I don't know for sure... but the flash button? |
| 01:12.27 | zimdog | jskcr: for ztdummy to compile do I need to edit makefile first? |
| 01:12.33 | shido6 | yes |
| 01:12.36 | jerlique | How do I go about having different callerid for internal extensions as opposed to dialing externally. |
| 01:13.01 | shido6 | jerlique set it in that protocols conf file or set it per call in the dialplan :) |
| 01:13.17 | shido6 | set it in that protocols conf file using callerid="Name <number>" |
| 01:13.22 | jerlique | ok -thanks I'll take a look. |
| 01:13.26 | implicit | sup |
| 01:13.38 | shido6 | "/usr/src/asterisk/configs" is your friend |
| 01:14.00 | Juggie | if i combine the CDR records from two boxes into one table, how do i disginguish which record is from which server? |
| 01:14.02 | jskcr | zimdog what os and distro? |
| 01:14.15 | Poincare | hmmhesays: off course it was toooo simple :-) thanks |
| 01:14.32 | *** join/#asterisk netsurfer (netsurfer@84.12.25.194) |
| 01:14.48 | zimdog | jskcr: It is a custom Redhat 7.2 install |
| 01:15.50 | zimdog | I wonder how hard it would be to compile it locally and then copy it over to the dedicated server |
| 01:15.56 | jerlique | there is no protocols.conf file, even in that dir. google has no reference either |
| 01:16.16 | jerlique | where can I find the specifications on it. |
| 01:17.49 | jskcr | zimdog install the kernel source and the development so it can compile it should install just fine |
| 01:17.56 | jskcr | brb heating grill |
| 01:18.58 | shido6 | no no no |
| 01:19.01 | shido6 | if u are using sip |
| 01:19.03 | shido6 | u edit sip.conf |
| 01:19.06 | shido6 | and edit the user and peer |
| 01:19.07 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 01:19.14 | shido6 | and add a line for callerid |
| 01:19.15 | *** join/#asterisk guugmember (~nramos@200.6.202.231) |
| 01:19.20 | shido6 | if u are using iax |
| 01:19.22 | shido6 | then u edit iax.conf |
| 01:19.30 | shido6 | that is what I meant by the protocol's conf file |
| 01:20.04 | shido6 | u need look no further then your own box as the samples should all be there |
| 01:20.50 | lohelle | I have a script to generate a call file.. when I run ./generate 12345678 it adds all info required to make a call to Zap/g2/12345678 to a call file.. (cont) |
| 01:21.42 | lohelle | the script is now modified to be like this ./generate 12345678 3 60 , where 12345678 is number to call, 3 is retries and 60 is waittime |
| 01:22.11 | talkwebhosts | if it says unable to lookup sip.broadvoice.com what should i do? |
| 01:23.24 | lohelle | how can I create a context to inpt this to the system app.. ? I can do it with _XXXXXXXX,1,System(/root/generate ${EXTEN}) to add just phone number.. but how to add more that one input? |
| 01:23.56 | jerlique | Ahhh, cool |
| 01:24.48 | zimdog | jskcr: Ok thanks. |
| 01:25.05 | *** part/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 01:25.23 | Juggie | lohelle, look at Read() to read input |
| 01:25.57 | lohelle | ok |
| 01:26.33 | lohelle | if someone is interested.. the two scripts is at www.tech-support.no/generate.txt (and generate2.txt) |
| 01:26.35 | talkwebhosts | alright everyone |
| 01:26.47 | talkwebhosts | i got it to answer a call |
| 01:27.22 | talkwebhosts | but it displayed this NOTICE pbx.c:1318 pbx_extension_helper: Cannot find context from broadvoice |
| 01:27.48 | jerlique | shido: I have callerid set in sip.conf already, but it presents this number internally as well as externally. What I want is it to show callerid 204 for an internal call and say xxxxx204 for external calls. |
| 01:28.01 | talkwebhosts | what do i do? |
| 01:28.46 | Juggie | jer, then in your callerid set the number to 204, and in your outgoing context write some code to modify the callerid to add the rest of the digits |
| 01:29.00 | jerlique | no worries... |
| 01:29.53 | talkwebhosts | jerl |
| 01:30.01 | talkwebhosts | any idea of the pbx extension helper error? |
| 01:30.17 | *** join/#asterisk Pulu (~chatzilla@64.200.224.158) |
| 01:30.56 | *** part/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 01:31.35 | *** join/#asterisk dca (~teliax@c-67-166-37-218.client.comcast.net) |
| 01:31.57 | Juggie | looks like the context isnt in extensions.conf? |
| 01:31.58 | jerlique | dont really know, but are you sure you have written the context listed in the broadvoice section?? |
| 01:32.03 | dca | is there anyway to test SMP on my 5.3 install on a dual xeon? |
| 01:32.18 | Juggie | dca, test how? |
| 01:32.20 | jerlique | also try search google for "pbx_extension_helper: Cannot find context from" |
| 01:32.20 | dca | opp, sorry though i was in #freebsd |
| 01:32.45 | guugmember | where can i find information about AGI? |
| 01:33.23 | talkwebhosts | ok |
| 01:33.57 | guugmember | http://www.voip-info.org/wiki-Asterisk+AGI |
| 01:34.24 | hmmhesays | that's a good start |
| 01:36.19 | guugmember | AGI seems like a huge power of asterisk, am i right? |
| 01:36.39 | implicit | no |
| 01:36.49 | implicit | it is just a part, i wouldn't say it is a 'huge' part |
| 01:36.50 | hmmhesays | just enables people to easily write functionality |
| 01:36.56 | implicit | its not really 'required' for people to do anything |
| 01:37.00 | hmmhesays | he said "power" not part |
| 01:37.18 | implicit | oh, |
| 01:37.26 | implicit | eh, it is okay |
| 01:37.37 | implicit | eh, good if you want to do some rapid prototyping |
| 01:37.40 | hmmhesays | it has it's place |
| 01:38.00 | guugmember | i mean, it lets you do what other PBX dont, like Avaya |
| 01:38.19 | guugmember | AGI is the power of Asterisk in term of integration, right? |
| 01:38.47 | implicit | there are many different ways to interact with asterisk agi is one of them |
| 01:38.48 | ManxPower | guugmember, AGI is ONE of the ways you can extend Asterisk. |
| 01:39.09 | *** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net) |
| 01:39.09 | ManxPower | There is also the Manager Interface and the C API. |
| 01:40.11 | guugmember | ManxPower, the maganer interface? will search for it |
| 01:40.12 | *** join/#asterisk djMax (~djMax@dsl093-190-107.nyc2.dsl.speakeasy.net) |
| 01:40.31 | djMax | is there an easy way to normalize voicemail before saving/sending? |
| 01:40.48 | ManxPower | djMax, not really |
| 01:41.12 | djMax | I suppose a cheating way would be to alter the sendmail script? |
| 01:41.30 | guugmember | ManxPower, what is the most advance and flexible? |
| 01:41.44 | guugmember | ManxPower, tool to extend Asterisk |
| 01:41.51 | djMax | extend in what way? |
| 01:41.56 | hmmhesays | it depends on what you are looking to do |
| 01:42.12 | liquidno2 | like a pair of plyers |
| 01:42.42 | freat | I just jumped on... but I notice you're talking about API's... has there been anywork to integrate JTAPI w/ Asterisk? |
| 01:43.19 | freat | JTAPI = Java Telephony API |
| 01:43.54 | liquidno2 | What is the ring detection called? cadence? |
| 01:44.03 | liquidno2 | ring cadence? |
| 01:44.31 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 01:47.32 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 01:52.05 | redder86 | distinctive ring |
| 01:54.43 | talkwebhosts | how do i setup extensions? |
| 01:55.04 | talkwebhosts | i want it to answer and a voice to say what the extensions are |
| 01:55.20 | talkwebhosts | i am getting this error now No such host: 2001 |
| 01:55.33 | talkwebhosts | Unable to create channel of type SIP |
| 01:55.41 | Qwell | somebody has some more reading to do |
| 01:56.06 | talkwebhosts | point me to the right direction |
| 01:56.17 | Qwell | voip-info.org |
| 01:56.19 | Qwell | ~docs |
| 01:56.20 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:56.40 | *** join/#asterisk guugmember (~nramos@200.6.202.231) |
| 01:57.51 | dr342346 | ok problems --- firstly i dont really want to spend 60 $ on eyebeam, 2ndly I was trying to use SJPhone which does everything i need except that... when you transfer to 700 to "park" a call SJPhone just transfers and kills the line and never tells you want extension to pick up on IE 701 .. 702 etc... Does anyone know of a way to fix this or another SIP softphone that doesnt have this same problem other than that I am quite satisfied w/ the |
| 01:58.42 | *** join/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 01:59.36 | dr342346 | >>?? |
| 02:10.17 | Mavvie | who knows, besides e614.org and e164.arpa, any other NAPTR based DNS directories? |
| 02:13.01 | Nethab | Mavvie: have you looked at dundi |
| 02:13.21 | Mavvie | Nethab: yes, but that doesn't work with NAPTR records, does it? |
| 02:13.51 | Nethab | it works just as well as e164, and has a dns interface too |
| 02:14.00 | Mavvie | aha, which zone? |
| 02:14.27 | Nethab | but like e164 you have to put your numbers in, or service providers have to |
| 02:14.55 | dr342346 | <PROTECTED> |
| 02:15.48 | Mavvie | Nethab: aha. But right now I'm looking for DNS servers which have NAPTR records. |
| 02:16.06 | Nethab | which has them all? |
| 02:17.15 | Mavvie | *any* |
| 02:17.20 | Nethab | i think the North American phone companies keep their records pretty secret unless you have a direct peering agreement with them |
| 02:18.57 | Nethab | in europe several countries keep a central list to help keep track, but here, they're quite the protectionists |
| 02:21.02 | Mavvie | aha, do you know one there? |
| 02:21.17 | Nethab | in europe? |
| 02:21.39 | Mavvie | yes |
| 02:22.11 | Nethab | nope sorry |
| 02:22.21 | Mavvie | but you know which country? |
| 02:22.30 | Mavvie | a/which/a |
| 02:22.57 | Mavvie | aha, .dk has one |
| 02:24.33 | brc_ | dr342346, so in otherwords, x doesn't do what I want, but y does, but I'm too cheap to pay for y, so how can I make x do it for free |
| 02:25.37 | Qwell | brc_: I couldn't have said it any better |
| 02:27.02 | Chuji | well my xten phone paired up to my bluetooth headset it almost cool |
| 02:27.08 | Chuji | Wish it got better range |
| 02:27.34 | Chuji | Bluetooth doesn't get near the range advertised |
| 02:27.48 | Qwell | 30' is in the spec, right? |
| 02:27.53 | Chuji | Yeah |
| 02:30.02 | jerlique | I'm playing with queue and agents. When the incoming call comes in, it doesnt get allocated to any queue. How can I see why? |
| 02:30.50 | *** join/#asterisk ChkDigit (~mike@static65-87-226-124.regina.accesscomm.ca) |
| 02:31.06 | Chuji | jerlique : Are you queuing the call? |
| 02:31.52 | ChkDigit | Has anyone ever experienced trouble with an FXO module for a Digium TDM400P not putting the line back "on-hook?" |
| 02:32.10 | *** join/#asterisk jerlique (jerlique@lnk254.adl0.adsl.esc.net.au) |
| 02:32.23 | jerlique | sorry did I miss an answer- was d/c |
| 02:32.29 | ChkDigit | IRQ for my card is unique unto itself. |
| 02:32.35 | Chuji | ChkDigit : Yup, when it couldn't detect disconnect |
| 02:33.07 | ChkDigit | How is it fixed, or is it defective hw? |
| 02:33.20 | Chuji | jerlique : What does your queue line look like? |
| 02:33.31 | Chuji | jerlique : in extensions.conf |
| 02:33.57 | Chuji | ChkDigit : Just plugged into a pots line? |
| 02:34.09 | ChkDigit | Chuji: Yup. |
| 02:34.36 | Chuji | ChkDigit : put the logger in debug and see it shows anything interesting |
| 02:34.46 | ChkDigit | Chuji: TDM400P, 3 FXS modules, 1 FXO... |
| 02:35.00 | jerlique | exten => s,3,Queue(tech|T) |
| 02:35.03 | ChkDigit | Chuji: As in the kernel module debug logging? |
| 02:35.13 | Chuji | no, logger.conf |
| 02:35.56 | ChkDigit | k... |
| 02:38.08 | Chuji | ~tdm400p |
| 02:38.25 | Chuji | ~tdm400 |
| 02:40.35 | jerlique | I've also tried with out the tech declaration... |
| 02:41.40 | *** join/#asterisk lilneon (~tj_r3@200.108.19.99) |
| 02:41.44 | lilneon | hi everyone |
| 02:42.37 | lilneon | hey guys anyone knows how to make sjphone or some other softphone work on winXP behind a proxy and firewall?? |
| 02:43.26 | lilneon | hello? where is everyone? |
| 02:44.06 | ChkDigit | Chuji: Turned on logging, and now it is doing it intermittently... |
| 02:44.09 | brc_ | all of our support agents are currently busy assisting other customer^H^H^H^H^H^H^Husers |
| 02:44.27 | brc_ | all of our messsages will be answered in the order they were received |
| 02:44.47 | ChkDigit | I get the message zt_set_hook zt hook failed to set hook: Device or resource busy. |
| 02:45.01 | brc_ | you are chatter number 242, your estimated wait time is fifty nine minutes and twenty seven seconds |
| 02:45.28 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 02:45.42 | brc_ | lilneon, basically, yes it can be done, yes it's a pain in the ass and will take you hours upon hours to figure out |
| 02:46.52 | lilneon | brc_: care to give me a head start? i tried this your-freedom.net thing which claims to bypass the proxy and firewall.. but sill nothing |
| 02:46.53 | brc_ | simple way would be to forward port 5060 and 10000 - 20000 to the phone inside the nat |
| 02:46.56 | jerlique | brc_ what are you refering to with the queues? I'm trying to set this up now |
| 02:47.04 | brc_ | jerlique, eh? |
| 02:47.09 | brc_ | I was making what is known as a joke |
| 02:47.19 | brc_ | you're trying to setup wait time announcements? |
| 02:47.25 | dan2 | is it possible to intercept calls? |
| 02:47.34 | dan2 | like if someone is ringing Zap/2 |
| 02:47.34 | brc_ | dan2, eh? |
| 02:47.41 | brc_ | see pickup groups |
| 02:47.43 | brc_ | wiki. |
| 02:47.45 | dan2 | and you are on Zap/3, but want to pick it up, with an extension |
| 02:47.49 | brc_ | yes it is |
| 02:47.51 | lilneon | brc_: that would be a good option, but i don't have admin access to the router.. or server room for that matter |
| 02:48.06 | brc_ | lilneon, can you use IAX2 instead? |
| 02:48.14 | brc_ | figure out what your time is worth |
| 02:48.40 | lilneon | brc_: yeah i could use iax2.. but they still have problems registering |
| 02:48.40 | brc_ | then look at the price of an iaxy |
| 02:48.48 | brc_ | then again, the iaxy's are far from prefect... |
| 02:49.05 | brc_ | I've had good luck with eyeBeam and xpro from xten.com |
| 02:49.11 | lilneon | brc_:other ppl connect to me fine... they can't ven ping my server |
| 02:49.30 | brc_ | what you do really depends on the specific type of nat you have, and if there is nat on the server side, etc |
| 02:49.46 | jerlique | hehehe!! |
| 02:49.59 | jerlique | sorry.. catching conversations midway through... |
| 02:50.07 | brc_ | uhu |
| 02:50.25 | brc_ | lilneon, ? |
| 02:50.41 | jerlique | do you know by any chance how to set them up? I get calls coming in, but they are not being allocated to the queue.. |
| 02:50.44 | brc_ | lilneon, if you care to give me the specifics of where you have nat I might be able to make some suggestions |
| 02:51.05 | brc_ | jerlique, well of course they aren't sent into the queue if you aren't telling them to go there |
| 02:51.09 | brc_ | it's not magic you know |
| 02:51.12 | brc_ | although |
| 02:51.14 | brc_ | wow! |
| 02:51.17 | brc_ | that's a great idea |
| 02:51.25 | brc_ | res_mindread |
| 02:51.38 | lilneon | brc_:the nat is on the client's side, with the softphone.. client(winxp,softphone)->proxy/nat->dsl->internet->myasterisk |
| 02:51.42 | brc_ | we might need chan_mindreadinginterface too |
| 02:51.55 | brc_ | asterisk has a public ip address? |
| 02:52.24 | brc_ | jerlique, I have no clue what you mean "but they are not being allocated to the queue.." |
| 02:52.41 | lilneon | brc_: well... port forwarded to it.. and works fine with other users... except those poor guys behind the nat/proxy @ the college |
| 02:52.45 | brc_ | lilneon, obviously you have nat=yes in sip.conf? |
| 02:52.52 | brc_ | what ports did you forward? |
| 02:53.20 | brc_ | ahhhhh |
| 02:53.21 | brc_ | hhhhh |
| 02:53.22 | brc_ | hh |
| 02:53.27 | lilneon | brc_:fowarded 4568-4570 (for IAX) and 5060 for SIP |
| 02:53.35 | brc_ | and 10000 to 20000? |
| 02:53.56 | lilneon | brc_:...um.... nope |
| 02:54.06 | brc_ | that's rtp |
| 02:54.14 | brc_ | the college is most likely doing...stuff... |
| 02:54.17 | brc_ | might even be blocking it |
| 02:54.27 | lilneon | brc_: yeah that is wha ti figured |
| 02:54.38 | brc_ | I'd forget about it...with sip at least |
| 02:54.47 | lilneon | brc_:cuz they can't even call digium's servers as atest |
| 02:56.39 | lilneon | brc_: so it seems both IAX2 and SIP blocked... dumb college :S..kids i tutor would have loved it too.. not to mention given me some extra cash for coffee :) |
| 02:57.22 | Juggie | sweet.... |
| 02:57.32 | Juggie | i just wrote a patch for cdr_addon_mysql |
| 02:57.36 | Juggie | my first patch.. yay. |
| 02:57.40 | implicit | why not just change the port? |
| 02:57.45 | Juggie | all be it useles since realtime is comming |
| 02:57.47 | Juggie | but none the less |
| 02:57.57 | lilneon | implicit: talking to me? |
| 02:58.11 | implicit | ya |
| 02:58.36 | lilneon | implicit: change it to wat? did u have success with a simila scenario? |
| 02:58.49 | implicit | yes, change it to something that is not blocked |
| 02:58.56 | implicit | and put your server on that port too |
| 02:59.14 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 02:59.17 | Nethab | with IAX2 you can get by with one port, his suggestion is to find one that's open |
| 02:59.33 | lilneon | hmmm |
| 02:59.35 | lilneon | ok.. |
| 02:59.39 | Nethab | even if it's 80 |
| 03:00.23 | lilneon | Nethab,implicit: ok so once i got the open port.. how do i set it in IAX?.. don't remember a 'port:' in the .conf file is there? |
| 03:00.35 | lilneon | or is it in iaxprov |
| 03:01.15 | Nethab | it's in iax.conf |
| 03:01.40 | lilneon | really? cool.. ok |
| 03:01.59 | lilneon | gone to try it out guys |
| 03:02.02 | lilneon | hope it works |
| 03:03.00 | Juggie | anyone with more then one asterisk server want to try a patch for the CDR module, so u can tell which record in your cdr table is generated by which server? |
| 03:03.08 | harryvv | What can one exptect to charge for a small 25 phone based * network on a percentage of a commercial one? I keep hearing the commercial ones can run 10-30k |
| 03:03.38 | implicit | lilneon: sorry i was afk go ahead and message me |
| 03:03.44 | Juggie | harryvv, you could find someone to do it for hardware+per hour |
| 03:03.53 | Juggie | it shoudnt be too expensive |
| 03:04.02 | Nethab | i think in iax.conf it's called bindport |
| 03:04.18 | lilneon | NetHab: ok cool |
| 03:04.20 | implicit | harryvv: $10k is decent |
| 03:04.47 | implicit | (reasonable) |
| 03:04.50 | Juggie | 10k? |
| 03:04.57 | Juggie | including hardware? |
| 03:05.03 | implicit | no |
| 03:05.03 | harryvv | Customer would be happy if thay even save 25% under that of a commercial system? |
| 03:05.04 | implicit | just for work |
| 03:05.07 | implicit | not for hardware |
| 03:05.16 | harryvv | Thats the total package I am talking about |
| 03:05.37 | Juggie | hell, 25 sip phones with some outbound lines... i'd do it for alot less then 10k |
| 03:05.39 | implicit | if they want you to buy hardware it will be closer to 20k |
| 03:06.10 | implicit | Juggie: i can afford to charge 10k |
| 03:06.21 | harryvv | yea, the cards can be as much as 1k$ each? |
| 03:06.53 | Juggie | harry, do you have stats on your peak line usuage? |
| 03:06.55 | bjohnson | harryvv: new wiring? |
| 03:06.55 | Juggie | *usage |
| 03:07.18 | ph3nix | a commercial phone system can go for way more than 30k |
| 03:07.21 | harryvv | bjohonson, okay how many customers object to replacing there phones :) |
| 03:07.30 | harryvv | asuming thay are newer :) |
| 03:07.38 | bjohnson | harryvv: depends on how much they like them |
| 03:07.45 | harryvv | true |
| 03:07.56 | bjohnson | right .. what kind of phones .. you're changing the question now |
| 03:07.59 | jerlique | can anyone offer some suggestions to help me get agents/queues working? |
| 03:08.18 | Juggie | sip phones can go from 100-500$ |
| 03:08.47 | Juggie | i finally got the Mitel 5055 working properly and not crashing, the firmware is buggy |
| 03:09.00 | bjohnson | harryvv: 25 phone system you're looking at $10k in hardware + wiring + labour |
| 03:09.12 | bjohnson | harryvv: likely $20k to finish |
| 03:09.14 | brc_ | lilneon, you know....you could always just run iax2 over another port....... |
| 03:09.45 | brc_ | +oh |
| 03:09.50 | Nethab | brc_: we told him that, try port 80 |
| 03:09.58 | lilneon | brc_: i was just told that.. question though, once i bind the port wont itaffect the other ports? |
| 03:10.00 | brc_ | yis |
| 03:10.02 | harryvv | bjohnson wiring is done by me or a wiring tech. Ive done collage cat5 install before. But then again mabey want to keep my hands clean :) |
| 03:10.03 | bjohnson | harryvv: I don't know if voip is cheaper to purchase than a new pbx .. you win out on features and operating cost savings |
| 03:10.09 | brc_ | lilneon, yep |
| 03:10.16 | Nethab | you'll have to tell all your clients the new port in their phones |
| 03:10.20 | brc_ | lilneon, oygjfiil |
| 03:10.46 | brc_ | lilneon, or you could just use iptables to forward what ever alternate port you decide to use to 4569 iirc... |
| 03:10.52 | lilneon | brc_:hmm.. k |
| 03:11.05 | brc_ | might be complicated |
| 03:11.08 | brc_ | can be done |
| 03:11.26 | lilneon | brc_: yup.. another few sleepless nights |
| 03:11.30 | bjohnson | harryvv: you have a comparison cost for a new system? I'd guess it would be similar |
| 03:11.34 | Nethab | bad firewalls bad |
| 03:12.06 | Juggie | harryvv, 10k max for hardware |
| 03:12.16 | Juggie | thats with good phones |
| 03:12.33 | harryvv | bj, okay bpx for pbx phone for phone config for config asterisk vs commercial licenced system is about the same and this includes voip right? |
| 03:13.06 | brc_ | noooo |
| 03:13.15 | bjohnson | harryvv: I don't understand |
| 03:13.18 | shido6 | heheh |
| 03:13.19 | brc_ | me either |
| 03:13.32 | harryvv | comparing |
| 03:13.36 | brc_ | what's pbx |
| 03:13.38 | brc_ | er |
| 03:13.39 | brc_ | bpx |
| 03:13.49 | harryvv | pbx |
| 03:13.54 | harryvv | typo |
| 03:14.21 | netsurfer | hey bjohnson did u get the NAT problem fixed |
| 03:14.30 | Juggie | harryvv, if you are unconfident in asterisk get dualmode phones |
| 03:14.35 | bjohnson | netsurfer: not yet |
| 03:14.46 | Juggie | so if asterisk blows up in your face, you can drop in a cisco or mitel voip server |
| 03:14.49 | Juggie | however that wont happen |
| 03:14.59 | brc_ | dual mode? |
| 03:15.01 | brc_ | wtf? |
| 03:15.05 | Juggie | yah... |
| 03:15.06 | brc_ | SIP is sip |
| 03:15.11 | Juggie | duh |
| 03:15.15 | Nethab | a phone that does more than one protocol |
| 03:15.19 | Juggie | but the phones support more then one protocol |
| 03:15.25 | postel | a rose is a rose is a rose |
| 03:15.28 | Juggie | mitel 5215/5220 do minet & sip |
| 03:15.34 | bjohnson | harryvv: in a brand new .. walk into an empty building and start install system, I would think a * system would cost about the same as a commercial pbx |
| 03:15.35 | brc_ | ugg |
| 03:15.42 | Juggie | cisco 7960 and others does skinny, mgcp and sip |
| 03:15.43 | Juggie | etc. |
| 03:15.58 | harryvv | juggie no its not that I am unconfident but...that is a good idea. Its good to have a backup system in the even the system goes down and the legality that comes with it. |
| 03:16.01 | bjohnson | harryvv: operating costs and features would favour the voip solution |
| 03:16.11 | brc_ | bjohnson, I think you'll find that asterisk can be considerably less after you factor in the commercial licensing costs for some of the advanced features asterisk has out of the box |
| 03:16.22 | Juggie | harryvv, then i suggest going with a major phone manafacture which has dual mode phones |
| 03:16.24 | bjohnson | harryvv: it is rarely a walk into an empty building situation |
| 03:16.27 | brc_ | emailed voicemail...I've heard avaya systems are about $6k per 50 users for this alone |
| 03:16.29 | Nethab | and figure commercial systems come with a per diam cost |
| 03:16.33 | brc_ | call queues... |
| 03:16.38 | Nethab | per anum i mean |
| 03:16.51 | shido6 | not to mention you dont need PRI's or E1's to get it up and running :) |
| 03:17.00 | brc_ | harryvv, yes it is |
| 03:17.04 | bjohnson | brc_: depends on size .. he's looking at a 25 phone system .. not into the big boy territory |
| 03:17.08 | shido6 | u can go all IP for inbound and outbound dialing |
| 03:17.19 | postel | Juggie: yeah, if you can afford deploying CCM *and* asterisk in a corp env just to rollover to the other pbx if things go funny stay with CCM and get a support contract from cisco, you'll haev engibeers knocking on your door in 39.7 mins |
| 03:17.22 | harryvv | bjohonson I would suspect that is the case. Typically a existing digital phone system is in place and adding on to there existing commercial system might be brought up in a discussion. |
| 03:17.34 | brc_ | harryvv, your "backup system" can not be a totally different system with totally different configuration and totally different firmware for the phones and still be considered a backup |
| 03:17.42 | Juggie | postel, i ment to NOT purchase a CM or MINET voip server |
| 03:17.44 | Juggie | just the phones |
| 03:17.49 | brc_ | a plan e, *maybe*, a backup? no. |
| 03:17.50 | harryvv | bj, when I get confortable in small network sizes then move up of course. |
| 03:17.54 | Juggie | but if all hell breaks loose, or legality gets in the way |
| 03:18.03 | Juggie | you have the option of putting in one of those boxes |
| 03:18.13 | brc_ | legality? what on EARTH are you talking about? |
| 03:18.14 | Juggie | and switching the mode of your phone |
| 03:18.16 | Juggie | so your phones arnt lost |
| 03:18.34 | Juggie | brc_, some places dont like OS because their is unlimited liability |
| 03:18.51 | brc_ | first of all, there, not their |
| 03:19.17 | Nethab | get two servers running HA-software, with shared storage, that's what we had where i just worked |
| 03:19.22 | Juggie | regardless of my spelling, my point is still valid |
| 03:19.29 | harryvv | brc, when a phone or asterisk fails in the most critcal time of need like a 911 call does not go though. |
| 03:19.35 | brc_ | second of all, if you read the end user license agreements that come with any commercial software at all you will find that they absolve themselves of any and all liability |
| 03:19.39 | harryvv | Thats a big legal issue. |
| 03:19.50 | harryvv | I see |
| 03:19.53 | bjohnson | Juggie: you're approach assumes a server problem and everything else continues to work .. backup hardware and a good backup system would handle that quite nicely |
| 03:19.57 | Juggie | brc_, in the case of my organization, we negociate liability with the vendors |
| 03:20.30 | bjohnson | listen to Nethab |
| 03:20.42 | Juggie | so we have a contract with say microsoft which goes beyond what is in the EULA |
| 03:21.00 | brc_ | harryvv, maybe I am not following what the hell you are talking about at all. If you have gizmo phones running a sip image connected to asterisk, and your backup plan is to switch to a gizmo image and a gizmo server, you ain't gonna do that instantly when your asterisk server goes down... |
| 03:21.09 | Nethab | listen to me? I don't even listen to ,e |
| 03:21.10 | Nethab | me |
| 03:21.18 | Juggie | brc_, no one is arguing that you can |
| 03:21.30 | brc_ | sure sounded like that's what harryvv expected |
| 03:21.45 | brc_ | talking about 911 calls and all... |
| 03:22.12 | bjohnson | harryvv: it is common to have an emergency phone that will still work if the power goes out .. just make sure you have at least one of those and you should be ok |
| 03:22.32 | brc_ | also known as TBFRP |
| 03:22.41 | ChkDigit | Ok, I'm getting a little further with my Zap FXO module... |
| 03:22.42 | brc_ | the big fine red phone (on the wall) |
| 03:22.46 | Nethab | you have to look at it this way, if the power goes out, your phones will go down anyway, getting an FXO gateway hooked up for emergency failover isn't that hard |
| 03:22.54 | harryvv | yea, worst case scenario |
| 03:22.56 | Juggie | its not easy to keep voip phones running when the power goes out |
| 03:22.58 | Juggie | but its possible |
| 03:23.00 | brc_ | sure it is |
| 03:23.01 | ChkDigit | It appears as though it will hangup the line, then pick it right back up. |
| 03:23.17 | Juggie | you need to have enough money to have a nice battery backup |
| 03:23.20 | brc_ | Juggie, why is it *any* different to an analog pbx? |
| 03:23.24 | Nethab | if your phones support multiple gateways, use the FXO as a backup |
| 03:23.35 | Juggie | brc_, higher power demands |
| 03:23.38 | bjohnson | POE with battery backup would work in a power outage situation and wouldn't be possible with a standard pbx |
| 03:23.44 | Nethab | and keep only that and Power Over Ethernet if your phones have that |
| 03:23.51 | bjohnson | but I don't think harryvv was really talking about power outages |
| 03:23.55 | brc_ | me either |
| 03:24.00 | bjohnson | I think he was referring to stabilty |
| 03:24.16 | PatrickDK | hmm, normal pbx just needs a ups on the pbx, not the whole network/servers |
| 03:24.36 | Nethab | if you get phones that can use multiple gateways, use a SIP based FXO gateway. WOn't that work as a backup |
| 03:24.40 | brc_ | I have a seperate network for the phones |
| 03:24.40 | Juggie | voip needs a ups on the pbx and the poe switch. |
| 03:25.16 | brc_ | you should have your data network on ups's anyway though |
| 03:25.17 | harryvv | When I worked for a global fortune 500 company both data centers went down. It was panic time as all servers went down because of a stupid mistake by one of the firealarm inpectors. It causes the company to loose 1 million in transactions that day. |
| 03:25.37 | bjohnson | PatrickDK: depends on "normal" .. that is true for handsets that get power from the line they are connected to .. exactly like a power of ethernet voip solution |
| 03:25.38 | Juggie | harryvv, someone hit the big blue button in our lab one day too :) |
| 03:25.41 | Juggie | shut down the entire thing. |
| 03:25.42 | brc_ | harryvv, yup |
| 03:25.44 | harryvv | Took hours to bring up all the server |
| 03:25.45 | harryvv | :) |
| 03:25.49 | bjohnson | power OVER ethernet |
| 03:26.04 | brc_ | 802.3af |
| 03:26.07 | Chuji | ~poe |
| 03:26.08 | jbot | poe is probably Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt |
| 03:26.16 | Chuji | heh |
| 03:26.18 | Juggie | hah |
| 03:26.18 | Juggie | nice |
| 03:26.21 | Nethab | is there a good SIP FXO gateway out there? |
| 03:26.23 | Chuji | ~POE |
| 03:26.24 | jbot | extra, extra, read all about it, poe is Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt |
| 03:26.26 | Juggie | but wrong answer for telephony |
| 03:26.32 | brc_ | harryvv, what size company are you considering this all for? |
| 03:26.39 | bjohnson | Nethab: how many ports |
| 03:26.44 | Chuji | what the hell is jbot thinking? |
| 03:26.51 | brc_ | could be that asterisk just isn't a good fit for what you require |
| 03:26.56 | Nethab | for emergencies, probably only need one port right? |
| 03:27.04 | harryvv | brc, I will start small mabey a insruance company with some small branch office sites. |
| 03:27.28 | brc_ | Nethab, depends on the size of the company |
| 03:27.28 | brc_ | harryvv, numbers would be very helpful... |
| 03:27.28 | Juggie | harryvv, are 25 phones on one site, or multiple? |
| 03:27.28 | brc_ | 20 extensions? 500 extensions? |
| 03:27.32 | bjohnson | Nethab: Sipura SPA 3000 .. one fxo, one fxs .. in power outage or loss of ethernet connection, the fxs directly connects to the fxo |
| 03:27.36 | harryvv | brc, Im not at that stage yet. |
| 03:27.41 | bjohnson | Nethab: $100 USD |
| 03:28.01 | harryvv | I would have to inquire on a company by company basis then go from there. |
| 03:28.05 | Nethab | i have a 3000, but if the office has all SIP phones, if * goes down, could they fail over to a FXO gateway |
| 03:28.11 | Juggie | harryvv, would your first implementation be one site, or across many sites. |
| 03:28.30 | postel | brc_: really? there are possible voip deployments that * cannot scale to? <brc_> could be that asterisk just isn't a good fit for what you require |
| 03:28.50 | Chuji | Nethab : Yeah, you would just need to plug the fxs of the 3K into an FXO gateway |
| 03:28.56 | brc_ | of course |
| 03:29.08 | postel | brc_: share some examples with the rest of us |
| 03:29.10 | brc_ | *nothing* can scale infinitely |
| 03:29.37 | postel | brc_: NO deployment is infinite, its pretty deterministic if you ask me |
| 03:29.39 | brc_ | I don't have personal experence with anything over 50ext |
| 03:29.40 | bjohnson | totally depends on what you're comparing |
| 03:29.46 | Sedorox | Question... I just setup FWD via IAX.. (according to their IAX thing on their website)... when I dial '393612' I get '404' displayed on my phone.. and there is nothing in the asterisk console screen... any ideas? |
| 03:29.49 | Juggie | i wonder what would happen if u put 4 96port cards in a box |
| 03:29.53 | Juggie | thats 384 lines |
| 03:29.57 | Juggie | er more |
| 03:30.02 | Juggie | no nm thats right. |
| 03:30.06 | Chuji | Juggie : Too many interrupts |
| 03:30.13 | harryvv | yea |
| 03:30.14 | bjohnson | for someone that is completely happy with their one line home phone system, the thought of * is just ludicrous |
| 03:30.20 | jerlique | sedorox: so you did enable IAX on FWD? |
| 03:30.26 | harryvv | risking a irq problem Like I did last night. |
| 03:30.26 | Nethab | Chuji: with the 3000 i would think going SIP in to FXO out would be easier |
| 03:30.42 | Juggie | Chuji, are you sure? even with serial, usb, printer port disabeled |
| 03:30.50 | Juggie | only a network card & video |
| 03:30.54 | Nethab | my question is can the spa-3000 be used as a sip gateway to the FXO port |
| 03:30.54 | Sedorox | jerlique: yes.. it says it registers with them... and I see the _393. extentions load... but when I dial it.. I just get that... |
| 03:31.05 | Chuji | Juggie : Not bios inturrpts, Interrupts per second |
| 03:31.26 | brc_ | postel, what's the largest asterisk deployment you've done? |
| 03:31.26 | bjohnson | Nethab: I don't understand |
| 03:31.32 | jerlique | did you wait ten minutes for FWD to enable it? what does 'iax2 show registry' show |
| 03:31.34 | Qwell | If I wanted just 1 FXS port, what would you guys recommend? |
| 03:31.34 | Chuji | Nethab : Yeah, I was thinking you meant in the event of a power outage |
| 03:31.57 | Sedorox | jerlique: its been over 30 mins... |
| 03:31.58 | Sedorox | ummm |
| 03:31.59 | brc_ | Qwell, depends |
| 03:32.04 | Chuji | Qwell SPA 1001 |
| 03:32.05 | brc_ | Qwell, are you looking for an ata? |
| 03:32.05 | postel | brc_: i dont work with * in the corp, we use CCM, i use * in my house caz i cant afford CCM |
| 03:32.06 | bjohnson | Juggie: for that many fxs ports you would switch to a channel bank array |
| 03:32.11 | brc_ | SIPura's work great |
| 03:32.13 | Qwell | brc_: doesn't matter to me much, really |
| 03:32.15 | Sedorox | 65.39.205.121:4569 589476 64.251.71.178:4569 60 Registered |
| 03:32.30 | Juggie | bjohnson, not fxo ports, i was talking 4xTDM405P something like that |
| 03:32.39 | jerlique | is the phone connected to * or FWD. what protocol you using? |
| 03:32.45 | brc_ | postel, so what do you think of CCM? |
| 03:32.53 | Chuji | postel : Even if you could afford it, why would you want CCM instead of *? |
| 03:33.13 | harryvv | btw, from my experaince last night with a irq conflict on a x100p that even interoffice options cannot work like vm. I did not have a second phone to test the softphone could contact the second one. Regardless thats not good that vm would be blocked because the irqs decided to do a switch. |
| 03:33.17 | Chuji | postel : For the home enviornment, * is far better |
| 03:33.26 | bjohnson | Qwell: some of the voip providers include hardware .. you could look at that if looking to sign up anyway. eg Broadvoice has free SPA 1001 |
| 03:33.27 | Nethab | i think the question would be, how 'available' can asterisk be |
| 03:33.40 | Qwell | bjohnson: I'll just be using the pstn. I already have an fxo |
| 03:33.40 | Sedorox | the phone is connected to *, which is connected to FWD.... for the phone to that * box.. I thinkG729 or something.. and for IAX.. I do have allow=ulaw... like I said.. followed there instuctions down pretty well |
| 03:33.41 | postel | Chuji: caz i can find my way round it and i have access to cisco resources and people |
| 03:33.43 | brc_ | postel, do you admin the ccm at your office? |
| 03:33.50 | brc_ | how many people are on it? |
| 03:34.10 | bjohnson | Qwell: or you can buy for $65 .. but for $80 you can get SPA 2000 with 2 fxs |
| 03:34.20 | postel | brc_: the network side of it, im an architect, i deal with trunks, the engineers handle the software |
| 03:34.23 | Qwell | bjohnson: buy what for $65? |
| 03:34.24 | jerlique | which phone are you using? is it sip or iax? |
| 03:34.24 | brc_ | ahh |
| 03:34.27 | Qwell | oh, 1001, right |
| 03:34.45 | Sedorox | sip... its a BT100... |
| 03:35.00 | jerlique | do a 'sip show peers' |
| 03:35.02 | zimdog | What is the quickest way to get sip working with nat? I have the * server behind a nat firewall. I have opened prots 5060 and 10000-20000 udp. The xlite client can call an extension on the server but no audio is transmitted. The sip phone behind the firewall with * cannot call the outside sip phone goes straight to fast busy |
| 03:35.03 | Sedorox | I have it working fine to another asterisk box... over IAX.. and to another sip phone... |
| 03:35.15 | jerlique | oh ok. |
| 03:35.16 | Sedorox | I can dial all extentions on the system.. the phone works.. save the FWD exten... |
| 03:36.05 | zimdog | I was looking at putting an asterisk on my external dedicated server but this might be to complicated. I am looking at SER right now which looks like it may solve the problem but seems like more than what is needed. |
| 03:36.10 | Sedorox | its like it doesn't seperate the extention out... I tried just 393.. and 393612 (their time thing).. and 393.. then waiting.. and nothing... |
| 03:36.16 | bjohnson | harryvv: that would have been a hardware problem and could be fixed in a variety of ways |
| 03:36.25 | brc_ | zimdog, NAT=yes? |
| 03:36.36 | jerlique | ahh its 613 isnt it? |
| 03:36.39 | brc_ | zimdog, sip.conf. read the sample. set nat=yes |
| 03:36.50 | Sedorox | : How can I tell it is working? |
| 03:36.50 | Sedorox | amir: Test outbound by calling 393-612 to .... |
| 03:36.51 | brc_ | zimdog, also, set externip= in sip.conf |
| 03:36.59 | Sedorox | 613 didn't work either |
| 03:37.13 | Nethab | 612 is time server |
| 03:37.24 | Nethab | that worked for me even when echo server didn't |
| 03:37.25 | bjohnson | Qwell: I think the 1 port fxs units at voipsupply are $65 |
| 03:37.28 | jerlique | have you assigned the number to an extension ? |
| 03:37.39 | Qwell | bjohnson: I still don't trust the pricing at voipsupply |
| 03:37.53 | Nethab | voipsupply's shipping is a ripoff |
| 03:37.59 | Nethab | voxilla is cheaper |
| 03:38.11 | zimdog | brc_: had the nat=yes for the extension entrty in the sip.conf already will try the externip |
| 03:38.21 | zimdog | thanks |
| 03:38.25 | Sedorox | I have the... exten => _393.<.......> stuff in a context called [FWD-out] and I included it in with the other extentions of that nature that gets called |
| 03:38.29 | Qwell | They had a tdm40b listed as like $250, so I asked them about it, and they said "oh, oops, the price is wrong. hang on, we'll fix it." "ok, done, try now." "umm, $400?" "oh, sorry...one more try" |
| 03:38.33 | Sedorox | I haven't worked with this part of * before.. so... |
| 03:38.33 | brc_ | zimdog, then setup a stun server and tell xlite what it is |
| 03:38.40 | jerlique | to you have the [fromiaxfwd] context |
| 03:38.43 | brc_ | or use fwd's stun server |
| 03:38.45 | bjohnson | Qwell: then voxilla |
| 03:39.02 | Sedorox | ok.. its included in local |
| 03:39.09 | Sedorox | well.. I have it as [FWD-in] |
| 03:39.10 | zimdog | is SER a stun server? |
| 03:39.12 | Sedorox | but yes.. have that setup too |
| 03:39.29 | dan2 | how do I make stutter tone for mailboxes only go to specific Zap/n |
| 03:39.30 | jerlique | do you include => fwd-in |
| 03:39.39 | Nethab | in extensions.conf my FWD setting is like this exten => _91393.,1,Dial(IAX2/fwd-out/${EXTEN:5}) |
| 03:39.48 | bjohnson | zimdog: if you have a server at the remote location anyways, run * there and use iax to connect the 2 iax serevers .. then sip can be used on that LAN |
| 03:40.02 | Sedorox | both of those context's are included in the local contect... and when I do a reload on asterisk.... I see it talking about the _393. loading |
| 03:40.12 | brc_ | zimdog, no |
| 03:40.14 | brc_ | google. |
| 03:40.14 | jerlique | whats it say |
| 03:40.18 | Sedorox | Nethab: mine is similar.. just following the guude.. then I'm gonna be chnaging |
| 03:40.37 | Sedorox | <PROTECTED> |
| 03:40.46 | Sedorox | through priority 3 |
| 03:41.36 | bjohnson | brc_: is it possible to use FWD as a STUN server? |
| 03:41.42 | brc_ | no |
| 03:41.50 | brc_ | you can leach off of their stun server though |
| 03:41.57 | zimdog | bjohnson: I don't really have two locations. I am looking at maybe where some users could connect through the internet to my * server |
| 03:41.59 | brc_ | all a stun server does is tell you your public ip |
| 03:42.22 | coppice | and port |
| 03:42.22 | jerlique | brc: what software package do they use? |
| 03:42.43 | bjohnson | zimdog: I'm looking at the same for SPA 2000s and would like to find a way to not require the externip setting in sip.conf |
| 03:43.04 | Sedorox | hmmm |
| 03:43.20 | brc_ | jerlique, for their stun server? I do not know |
| 03:43.24 | bjohnson | zimdog: I know that FWD has a stun server available to the public but don't understand the role of the stun server. |
| 03:43.24 | Nethab | zimdog: i have my * server behind my linksys, and two users at remote dsl locations behind their own linksys |
| 03:43.31 | brc_ | bjohnson, see above |
| 03:43.40 | bjohnson | brc_: gotcha |
| 03:43.58 | brc_ | so then your sip client puts the correct ip in the SIP headers |
| 03:44.15 | Sedorox | any ideas? |
| 03:44.25 | bjohnson | brc_: so I don't dial to it and then redial to my * server .. I just connect to the * server but config to use a stun server in the spa config |
| 03:45.06 | brc_ | http://www.google.com/search?q=stun+sip&sourceid=mozilla-search&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-US:unofficial |
| 03:46.44 | *** join/#asterisk Tommmo (~tps@203.62.181.52) |
| 03:46.54 | jerlique | no sorry. |
| 03:47.04 | Tommmo | is it possible to have different sets of hold music for different contexts within asterisk? |
| 03:47.12 | Juggie | yes |
| 03:47.23 | jerlique | actually have hyou done iax2 debug peer |
| 03:47.31 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 03:47.34 | jerlique | and sip debug peer |
| 03:47.44 | jerlique | and watch the logs? |
| 03:47.52 | jerlique | s/watch/watched/ |
| 03:48.05 | Juggie | Tommmo, you'll notice musiconhold.conf has dif sections |
| 03:48.08 | Sedorox | no... |
| 03:48.11 | Sedorox | good idea |
| 03:48.14 | Juggie | or classes |
| 03:48.17 | Juggie | whatever they call it |
| 03:48.22 | *** join/#asterisk zimdog (~zimdog@c-67-164-190-201.client.comcast.net) |
| 03:48.26 | Tommmo | ok thanks |
| 03:49.11 | firestrm | anyone here have a grip on which are the good,bad and just plain ugly as far as SIP or IAX ip phones?. I need to buy 60 of them, i need something that is bulletproof, doesnt have to have many bells and whistles just reliable and failure resistant |
| 03:49.58 | Tommmo | budget for each, firestrm? |
| 03:50.06 | firestrm | < $100 |
| 03:50.20 | Tommmo | i bought one of those sipura 841's and it seems great so far... |
| 03:50.22 | Sedorox | hmmmmmm |
| 03:50.25 | zimdog | brc_: sorry computer rebooted |
| 03:50.29 | Juggie | Tommmo, using the musiconhold tag u should be able to override it per sip user etc. |
| 03:50.34 | firestrm | or is $100 unrelistic? |
| 03:50.37 | Tommmo | yeah Juggie just found a doc about it |
| 03:50.40 | Juggie | k |
| 03:50.57 | *** join/#asterisk JochenA (~jochen@12-216-247-62.client.mchsi.com) |
| 03:51.00 | Tommmo | firestrm, i think this was $85 |
| 03:51.06 | Tommmo | only 1 ethernet port though ... |
| 03:51.25 | firestrm | Tommmo, thats fine.. only needs one.. |
| 03:51.32 | Sedorox | got it |
| 03:51.37 | firestrm | they are for remote branch phones.. |
| 03:51.42 | porkchop | I have a pair of Grandstream Budgetone 100s. $80ish. They're not bad, they do exactly what they're built to, but their codec set is a little limited. And theres no speeddial... they're pretty barren phones. |
| 03:51.46 | crazyhick | is the upgrade for the sipure 841 avaible yet? |
| 03:51.49 | porkchop | Not bulletproof |
| 03:51.52 | Sedorox | for some reason.. I couldn't have it included in the local context.. had to be included from the default context |
| 03:51.55 | Tommmo | not sure crazyhick |
| 03:51.58 | porkchop | The latest firmware broke the message button |
| 03:52.05 | shido6 | porkchop |
| 03:52.11 | shido6 | thats not true anymore |
| 03:52.14 | shido6 | they have ilbc |
| 03:52.15 | shido6 | and g729 |
| 03:52.16 | shido6 | now |
| 03:52.17 | shido6 | well |
| 03:52.21 | shido6 | they always had g729 |
| 03:52.24 | Tommmo | has anyone managed to get one of the new grandstreams? |
| 03:52.25 | shido6 | they now have ilbc |
| 03:52.27 | Tommmo | with the large LCD display |
| 03:52.28 | shido6 | that works great now |
| 03:52.39 | shido6 | on the LAN :) |
| 03:52.39 | porkchop | ilibc sucked, did it not? |
| 03:52.42 | Tommmo | i've ordered one from voipsupply.com but they don't answer emails :) |
| 03:52.44 | ph3nix | i like the polycom |
| 03:52.49 | crazyhick | i will never buy another grandstream. |
| 03:52.52 | Tommmo | yeah i have found the polycom excellent |
| 03:53.03 | porkchop | http://www.grandstream.com/images/BudgeTone.jpg <-- what I have |
| 03:53.16 | bjohnson | but different price bracket with the polycoms |
| 03:53.21 | ph3nix | i have a polycom and a 7960 |
| 03:53.42 | crazyhick | porkchop how long have you been using it. I have had 3 everyone of them died within a year |
| 03:53.43 | ph3nix | the 500's are like 150$ |
| 03:54.03 | porkchop | crazyhick: I've been using them all of three weeks. |
| 03:54.05 | Sedorox | jerlique: thanks for your help |
| 03:54.10 | brc_ | if you want a cheap phone I hear the SIPura phoens agre good |
| 03:54.56 | bjohnson | brc_: is there "issues" with "leaching" off the FWD STUN server? |
| 03:54.58 | Sedorox | I don't have problems with thius Budgetone 100 |
| 03:55.05 | Sedorox | <$70 |
| 03:55.38 | firestrm | thanks.. reliable is key.. it will cost me $600.00 for travel to fix em. and i have to give a year warrenty. |
| 03:56.05 | Juggie | ewww |
| 03:56.26 | Juggie | has anyone ever seen asterisk totally nuke cdr :) seems my iax is looping or something somehow |
| 03:56.30 | Juggie | writing records over and over |
| 03:56.40 | bjohnson | firestrm: up your price by $300 and give them a few spares |
| 03:56.45 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 03:56.53 | ariel_ | Good evening all |
| 03:57.01 | firestrm | but its a 20k contract for almost nothing.. 3 * boxen connected to 50 phones.. |
| 03:57.11 | bjohnson | harryvv: ^^ |
| 03:57.36 | bjohnson | harryvv: those are cheaper phones that what we were talking about |
| 03:58.34 | ariel_ | bjohnson, did you get your nat problem fixed? |
| 03:59.28 | bjohnson | ariel_: not yet .. I'm looking for a way to not need the externalip or the remote lan subnet of the remote sip |
| 03:59.36 | firestrm | <bjohnson> i have spares allready worked in.. the problem is its a northern community where most of the people dont shut down their computer because the tech flew in on a skiplane 3 years ago, started it up.. and never told them how to shut them down.. |
| 03:59.52 | bjohnson | ariel_: reading about the use of stun servers .. my spa 2000s are supposed to support that concept |
| 04:00.17 | Qwell | Who was talking about the relays the other night? |
| 04:00.24 | ariel_ | actually most sip devices do support stun servers. |
| 04:00.32 | ariel_ | But remember it's adding another layer. |
| 04:00.59 | Nethab | the only problem for me is that my phones ip is the same as my * box according to stun because of nat |
| 04:01.05 | bjohnson | ariel_: but doesn't require any additional config if the device is moved |
| 04:01.39 | bjohnson | Nethab: they are one different subnets? |
| 04:01.42 | ariel_ | bjohnson I was reading something in the provisioning guide about mac address tags. |
| 04:01.49 | Nethab | no all behind one linksys |
| 04:02.05 | bjohnson | Nethab: then why are you using stun? |
| 04:02.05 | jerlique | Is there a debug comment for queues? |
| 04:02.19 | bjohnson | ariel_: yes/ |
| 04:02.22 | bjohnson | yes? |
| 04:02.23 | ariel_ | Nethab, only one device makes it at a time through a linksys port just one connection. |
| 04:02.26 | Nethab | because some of my users are remote on their own dsl |
| 04:02.45 | ariel_ | bjohnson, do you need the pdf for this? |
| 04:02.47 | Nethab | everything works fine is i don't allow reinvites |
| 04:03.07 | bjohnson | ariel_: I don't know. What is it? Is it in cvs stable? |
| 04:03.23 | ariel_ | reinvites are a mess they don't really work with nats and fire walls. |
| 04:04.12 | ariel_ | bjohnson, no it's the office sipura provistion doc. |
| 04:04.56 | ariel_ | ./office/ official |
| 04:05.30 | bjohnson | are the mac addresses somehow used for NAT transversal? |
| 04:05.40 | ariel_ | Nethab, besides if you allow reinvites you don't have good cdr's on accounting of time use. |
| 04:05.47 | Nethab | mac addresses are not portable beyond your local subnet |
| 04:06.24 | ariel_ | bjohnson, when the sipura gets login it can find the server and tell it where it is. This way you an setup via pre-defined scripts there settings. |
| 04:06.47 | Nethab | not concerned about cdr really, i just don't want my grandma in texas to travel through california to talk to my aunt in washington |
| 04:06.53 | ariel_ | it just uses the mac address to the server to get the right config file. |
| 04:07.43 | ariel_ | Nethab, how do you know that they don't just do that any way. Have you seen the internet routes? |
| 04:08.10 | Nethab | because my 128k dsl can't handle them talking and me talking to someone else |
| 04:08.16 | ariel_ | I can traceroute something from here in Miami and to NJ and it ends up going via dallas. |
| 04:08.38 | ariel_ | ah then get them fwd numbers |
| 04:09.13 | ariel_ | or put your own router like m0n0wall. |
| 04:09.30 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.167) |
| 04:09.35 | sudhir492 | Hi all |
| 04:09.48 | Nethab | everything works fine, it's just going through my * server |
| 04:09.54 | ariel_ | sudhir492, welcome and join the fun. |
| 04:10.05 | sudhir492 | sure. Thanks |
| 04:10.21 | ariel_ | Nethab, well you have to live with some things to get it right. |
| 04:11.25 | firestrm | next Q.. if i have a pots line that i want to answer with *, but its in a remote location.. do i have to build another * box for that location? of is their some sort of FXO -> ip converter? |
| 04:11.33 | ariel_ | This world is just full of compromise |
| 04:12.03 | ariel_ | firestrm, use a ata like sipura 3000 |
| 04:12.51 | firestrm | ariel_, that will work on incoming pots calls? |
| 04:13.10 | ariel_ | firestrm, yes it has one fxo and one fxs port on it. |
| 04:13.19 | firestrm | or is that a FXS to ip |
| 04:13.37 | firestrm | ariel_, cool thanks.. i will chack that out |
| 04:13.43 | ariel_ | no fxo goes connected to the pots line fxs gets connected to the phone. |
| 04:14.38 | firestrm | ariel_, there is no phone to connect to.. all phones are IP phones fed from a central * server, i need a way to get a remote pots line to a central * server |
| 04:14.38 | *** join/#asterisk h3x (~Justino@nv-65-40-157-57.sta.sprint-hsd.net) |
| 04:15.03 | ariel_ | firestrm, yes that is a way. |
| 04:15.56 | bjohnson | firestrm: I love my SPA 3ks but if you need more than 1 pstn line ..there may be better ways |
| 04:16.11 | firestrm | ariel, thanks.. so for 10 1b lines i would have to have 10 sipura 3000 units.. yes? |
| 04:16.44 | PatrickDK | fiestrm, are all 10 pots lines in the same location? |
| 04:16.45 | bjohnson | firestrm: 1 nice selling point to the 3k is the failover connection between the fxs and fxo .. plug in an analog phone and you have emergency failover |
| 04:16.47 | ariel_ | firestrm, no in that case you need a channel bank |
| 04:17.11 | bjohnson | well .. he COULD use 10 SPA 3000s |
| 04:17.18 | firestrm | bjohnson, good point.. perhaps at least 1 1b line will be spa 3ks |
| 04:17.21 | bjohnson | just might not be the best |
| 04:17.21 | ariel_ | 10 pots lines are not your normal send over the net to me type of setup. |
| 04:17.43 | bjohnson | right .. should be looking at PRI if over 8 line |
| 04:18.37 | PatrickDK | heh, wish I could get a pri for a descent price here |
| 04:18.49 | firestrm | bjohnson, i dont know if they will give us a pri.. its a middle of nothern nowhere community. the 1b lines come in as analog from a city 110km away |
| 04:18.50 | PatrickDK | have 11 lines, for pots, paying like $250 |
| 04:18.59 | PatrickDK | for pri they wanted $530 |
| 04:19.10 | Juggie | thats 23 lines though |
| 04:19.12 | Juggie | thats not horrible |
| 04:19.13 | bjohnson | PatrickDK: full pri? |
| 04:19.17 | PatrickDK | no, partial |
| 04:19.19 | Juggie | oh |
| 04:19.25 | bjohnson | right .. double the lines for dounle the price |
| 04:19.27 | Juggie | thats different |
| 04:19.33 | bjohnson | plus pris have some neat tricks |
| 04:19.39 | PatrickDK | ya, I wanted pri |
| 04:19.40 | Juggie | pri's should be cheapr |
| 04:19.46 | Juggie | they cost the telco less you'd think |
| 04:19.50 | PatrickDK | then the told me they couldn't even give it to my, they made a mistake |
| 04:19.59 | firestrm | so what is a channel bank, and where do i get one? |
| 04:20.26 | PatrickDK | channelbank basically joins or splits 23 pots lines to one t1/pri line |
| 04:20.26 | bjohnson | from what I read, PRIs cost about 23 times what the single pstn lines cost .. so no savings there .. but you get some neat features |
| 04:20.43 | bjohnson | PatrickDK: up to 23 |
| 04:20.44 | Juggie | a channel bank will break out a t1 into analog ports |
| 04:21.05 | ariel_ | firestrm, you need one and an asterisk box at that location. |
| 04:21.05 | bjohnson | err .. might be 24 |
| 04:21.19 | PatrickDK | bj, 23 |
| 04:21.33 | PatrickDK | there are 24channels, but one is used for signaling |
| 04:21.36 | Tommmo | 1 taken for signalling |
| 04:21.37 | Tommmo | yeah |
| 04:21.42 | bjohnson | firestrm: tzanger suggested the adit600 for fxo configs .. check ebay |
| 04:21.44 | Juggie | t1 can be 24 though |
| 04:21.50 | Juggie | you dont have to have the signaling channel |
| 04:21.53 | Tommmo | yeah if a single D channel is used among multiple |
| 04:21.54 | ariel_ | If there bring 10 pots lines to your location in the boonie you need to check with them due to they might already have them as a t1 line there. |
| 04:22.04 | PatrickDK | juggie, only if you kill your voice quality |
| 04:22.16 | ariel_ | So that means you just get a digium T100p and a small asterisk and use iax2 trunking to send them over the net. |
| 04:22.19 | *** join/#asterisk juice (~juice@mo-65-41-96-204.dyn.sprint-hsd.net) |
| 04:22.25 | Juggie | PatrickDK, perhaps, but its possible |
| 04:22.35 | bjohnson | firestrm: plus t1 card from digium plus cabling to split the channel bank output to analog jacks |
| 04:23.38 | Tommmo | any recommended brands of channel banks ? |
| 04:23.39 | Tommmo | e1 |
| 04:23.44 | firestrm | bjohnson, can i have the channel bank remote and the t1 card local with 10MB full-duplex VLAN inbetween? |
| 04:23.46 | Juggie | firestrm, what do u want to do, use asterisk with a channel bank to provide analog ports? |
| 04:23.49 | bjohnson | PatrickDK: I don't have a cb .. but I read it is common to use the 24th port in that situation |
| 04:24.04 | PatrickDK | firestrm, no problem |
| 04:24.07 | ariel_ | adtran makes what I call the best for the price of channel banks. |
| 04:24.18 | bjohnson | adtran750? |
| 04:24.22 | PatrickDK | bjohnson, ya, can be, actually, if he is just doing channelbank to asterisk, run e1, get 30channels |
| 04:24.32 | PatrickDK | unless a e1 channel bank gets to be too hard to find |
| 04:24.40 | firestrm | Juggie, no im joining remote, no phone service communitis with terrestrial microwave voip phone service |
| 04:24.43 | ariel_ | e1 in the states to a c/b bad idea. |
| 04:24.55 | ariel_ | been there tried that. |
| 04:25.46 | ariel_ | adtran's 750/850 cb are about 400 to 500 dollars on ebay. they do both it just depends on the board you get it with. |
| 04:25.50 | bjohnson | firestrm: don't know what issues you might face if switches and routers and stuff between the cb and the * t1 |
| 04:25.58 | Juggie | firestrm, i'm not sure about running the t1 across the vlan i dont know how you'd do that... no idea :) |
| 04:26.01 | zimdog | arrrrg. I think I am going backwards with this nat problem |
| 04:26.14 | Juggie | you could put an * on both ends |
| 04:26.19 | Juggie | and trunk them over the 10mbit vlan |
| 04:26.22 | Tommmo | firestrm, can i make a suggestion |
| 04:26.28 | bjohnson | zimdog: I think I just got mine working by using a stun server |
| 04:26.35 | firestrm | Tommmo, go ahead |
| 04:26.42 | PatrickDK | firestrm, oh heh, no, you can't connect a channelbank to the network |
| 04:26.57 | Tommmo | put a RAD e1/t1 to ethernet converter on both ends |
| 04:27.02 | Tommmo | connect the channelbank to that |
| 04:27.07 | Tommmo | then a t1 will pop out on your end |
| 04:27.19 | Tommmo | no need for asterisk on each end |
| 04:27.37 | Juggie | Tommmo, how much does t1<=>ethernet cost? |
| 04:27.41 | Tommmo | few hundred |
| 04:27.51 | Juggie | does it operate with an ip? |
| 04:27.52 | Tommmo | i think around $1000 AU , maybe $600-700 US ? |
| 04:27.57 | Juggie | is it an ethernet device? |
| 04:28.02 | bjohnson | must be |
| 04:28.05 | Juggie | or does it just take the entire cable |
| 04:28.06 | Tommmo | yeah |
| 04:28.08 | Juggie | k |
| 04:28.12 | Juggie | that may work |
| 04:28.14 | zimdog | bjohnson: cool if it works give me some pointers. I just connected to the local lan with the xlite machine and now I can't connect at all. I get call not approved |
| 04:28.14 | Juggie | but at 600us |
| 04:28.18 | Juggie | * could be cheaper. |
| 04:28.19 | firestrm | Tommmo, are these mysitcal devices made by RAD? |
| 04:28.21 | Tommmo | it could be |
| 04:28.27 | Tommmo | but there is less to go wrong |
| 04:28.31 | Juggie | true. |
| 04:28.43 | bjohnson | zimdog: might be device specific .. don't know xlite .. I used SPA 2000s |
| 04:28.45 | Tommmo | and if it's a remote site, it may be worthwhile |
| 04:28.59 | Juggie | Tommmo, at least with * on both ends he could do some local routing as well |
| 04:29.01 | Juggie | for local calls |
| 04:29.04 | ariel_ | dell sc420 less then 400 dollars. nice asterisk trunk failover device at the other end. |
| 04:29.14 | zimdog | bjohnson: what stun server are you using? and is it outside the nat |
| 04:29.18 | Tommmo | Juggie, could do, i guess it depends what he's trying to achieve |
| 04:29.22 | Juggie | whats a sc420 do? |
| 04:29.28 | bjohnson | zimdog: take config that works on local lan. Enabled NAT support and NAT keep alive,. Enabled STUN server and put the FWD one. Done |
| 04:29.29 | Yoda-BZH`ZzZ | une bonne nuit je vous souhaite ! / A good night I wish you |
| 04:29.36 | ariel_ | Juggie, it's a Dell Server |
| 04:29.47 | Juggie | oh ok |
| 04:29.54 | Juggie | i thought it was something non server of some sort. |
| 04:30.23 | zimdog | so you just pointed the stun server setting on the phone to fwd's stun server? |
| 04:30.30 | ariel_ | Juggie, only ones non server that I have used are ata gateways like the mediatrik and audiovox. |
| 04:30.42 | ariel_ | They have there good points and many bad points. |
| 04:30.58 | Juggie | ariel_, with a 10mbit vlan i would put * on both ends. |
| 04:31.14 | bjohnson | zimdog: yes .. and enabled nat on the phone. Oh. and added nat=yes to sip.conf (only change I remember keeping) |
| 04:31.17 | ariel_ | Juggie, so would i. |
| 04:31.27 | Juggie | gives you routing to push local calls, as well as trunk calls over the vlan with iax to the other box then out to the pri or whatever |
| 04:31.32 | NormAst | Speaking of PRI's I have and * box with 4 PRI's in it and another box with a second pri. I can do Dial(Zap/G1/#) on the first box and it will get an outbound trunk.. If all lines are full I want it to use the PRI on the second box.. Any easy way of doin' this and keeping the dial plan simple? |
| 04:31.33 | jerlique | ariel: whats wrong with the mediatrix? |
| 04:32.02 | ariel_ | NormAst, dial rules. |
| 04:32.02 | NormAst | Virtual PRI group? |
| 04:32.07 | NormAst | no..!!! |
| 04:32.12 | zimdog | bjohnson: did you et the stun server ip from the wiki? |
| 04:32.28 | ariel_ | jerlique, nothing once you get them configured. Worst configuration setup I have run into. |
| 04:33.17 | bjohnson | NormAst: check out the superdial macro on the wiki |
| 04:33.41 | zimdog | bjohnson: never mind found it will give it a try |
| 04:33.48 | NormAst | Hmm...Like the idea of a Virtual PRI group... Take 3 PRI from Box A and 1 from box B... And do a Dial(Zap/VPRI_GROUP/#) |
| 04:33.52 | NormAst | :) |
| 04:33.56 | NormAst | dreaming... |
| 04:33.58 | h3x | NormAst: its easier than you think |
| 04:34.02 | bjohnson | NormAst: tries one route and if not available, fails gracefully back to extensions.conf for a second call to superdial through another channel |
| 04:34.03 | ariel_ | jerlique, Also they have a bad 2 sec dead silence that I have not been able to get ride of when transfering a call over. |
| 04:34.12 | h3x | if its "busy" then the +101 line in your dialplan can dial IAX to the other box |
| 04:35.11 | bjohnson | trust me .. check out the superdial macro |
| 04:35.20 | h3x | i havent heard of superdial |
| 04:35.22 | h3x | maybe its coz im running stable |
| 04:35.39 | ariel_ | h3x, hello how are you doing? |
| 04:35.39 | bjohnson | it's a macro .. not a * application |
| 04:35.44 | h3x | good |
| 04:36.04 | bjohnson | rolls the 1+101 stuff into a macro |
| 04:36.10 | ariel_ | macro is a fancy name for predone scripts. they work and can be your friend. |
| 04:36.23 | h3x | oh |
| 04:36.37 | bjohnson | adds a few other args like setting CID and cdr account |
| 04:38.19 | jerlique | ariel: great - I have a couple coming to me for eval shortly. |
| 04:39.11 | jerlique | Are there any commands to enable debuggin for queues? I am allocating a call to a queue and its not appearing in it. |
| 04:39.40 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 04:40.20 | NormAst | Can two * boxes share a Dial Plan? |
| 04:40.40 | h3x | yes |
| 04:40.43 | firestrm | NormAst, good wuestion |
| 04:40.57 | firestrm | w/q |
| 04:41.03 | NormAst | I know cp extensions.conf to second server.. |
| 04:41.05 | h3x | there are examples of how to do that in the extensions.conf and stuff |
| 04:41.05 | file[laptop] | DUNDi... IAX2 Switch... |
| 04:41.07 | h3x | no |
| 04:41.24 | h3x | like file said... |
| 04:41.29 | NormAst | :( |
| 04:41.54 | file[laptop] | those are the two that come to mind... |
| 04:42.13 | ariel_ | Well it's late and I need to get up early. (Baby in the house). See you all in the morning some time I hope. |
| 04:43.37 | Nethab | what about a shared database |
| 04:43.56 | file[laptop] | realtime yes... you could |
| 04:45.39 | NormAst | file[laptop] Is there a better way of linking two * boxes other than IAX. I have alot of TDM to TDM bridging, in a single box. And I need to add another. |
| 04:45.58 | Juggie | iax is the best way |
| 04:46.02 | file[laptop] | indeed |
| 04:46.23 | NormAst | Alot of over head? Jitter? |
| 04:46.32 | Juggie | depends on your lan |
| 04:46.44 | Juggie | if setup iax to run ulaw |
| 04:46.45 | Nethab | trunking should take care of overhead, jitter is a function of your connection |
| 04:46.53 | Juggie | then there is no transcoding |
| 04:47.00 | NormAst | none. |
| 04:47.09 | Juggie | because audio comes off your tdm card as ulaw and then is passed |
| 04:47.19 | NormAst | So I guess TMD in to IAX to TMD out. |
| 04:47.22 | Juggie | asterisk doesnt touch it |
| 04:47.41 | Juggie | yeah that will work |
| 04:47.58 | NormAst | Any options I might want to add to the IAX conf? |
| 04:48.00 | Juggie | if you are doing alot of TDM->TDM bridging, whats your application? |
| 04:48.23 | Juggie | NormAst, remove all the accounts except for the guest account |
| 04:48.34 | Juggie | set your bandwidth to high, disallow all codecs, and allow ulaw |
| 04:48.35 | NormAst | I have pri's to different locations and let people call between the locations for a flatrate fee. |
| 04:49.15 | Juggie | calling internal extensions? or regular telco numbers? |
| 04:49.30 | NormAst | Regular telco numbers. |
| 04:49.49 | Juggie | and u keep a list of free numbers at each location? |
| 04:50.02 | Juggie | or free exchanges |
| 04:50.03 | NormAst | I have a list of ACCESS number |
| 04:50.03 | Juggie | or w hatever |
| 04:50.24 | NormAst | and a dialplan that let's them call only ON-NET locations. |
| 04:50.58 | Juggie | and how are you doing your routing to those locations at the moment? |
| 04:51.09 | NormAst | I back haul the PRI from the location. |
| 04:51.36 | Juggie | ok i think i got it... |
| 04:52.20 | Juggie | now, did u want to add asterisk servers to your location where you host one now? |
| 04:52.23 | Juggie | or at one of the remote sites? |
| 04:52.46 | NormAst | I am going to add them add my Co-Location.. |
| 04:53.01 | firestrm | ariel_, you suggested using the spa-3000 for single incoming b1, what would that look like as far as aterisk and incoming calls are concerned? Im not sure how i would intergrate that into my extensions.conf |
| 04:53.14 | NormAst | Some time's the back haul charges are cheaper than the bandwidth required. |
| 04:54.38 | SexyKen | Is there real time call monitoring in Asterisk? |
| 04:55.36 | *** join/#asterisk PTG123 (Preston@ip68-106-19-249.ph.ph.cox.net) |
| 04:55.38 | netsurfer | SexyKen - sip show channels works for sip |
| 04:56.03 | SexyKen | netsurfer I mean the ability to listen to a conversation. So a tech admin can listen to his techs phone calls in real-time |
| 04:56.32 | netsurfer | ah |
| 04:57.27 | netsurfer | it used to be possible with sip but the command was removed, can only do it with zap channels now,ZapBarge I think it is |
| 04:57.34 | bjohnson | firestrm: it just comes in as a SIP call .. send it to a context that has a s extension |
| 04:58.27 | firestrm | bjohnson, so the spa will take care of picking up the line and transfering to * ? |
| 04:58.35 | bjohnson | yes |
| 04:59.02 | zimdog | what is rport for ? I think maybe this is causing a problem with nat and a polycom being able to register |
| 04:59.06 | bjohnson | little trickery required to get the 3k to send cid info to * BEFORE answering the call .. but info on the wiki for that |
| 04:59.10 | firestrm | bjohnson, im looking at a few units on ebay.. anthing to watch out for.. eg can they be perminatly locked somehow? |
| 05:00.16 | bjohnson | I don't think so .. but spa deals on ebay are rare. Just shop at voxilla or voipsupply |
| 05:00.38 | bjohnson | few users sell their spa's. They're too good |
| 05:01.04 | firestrm | bjohnson, thanks.. |
| 05:02.56 | zimdog | how do you connect to a cisco ata186 ? I went to the ip and it says invalid access. Do i need a certain port? |
| 05:03.54 | Tommmo | zigman, try http://ip/dev |
| 05:04.29 | zimdog | thanks Tommo |
| 05:04.44 | firestrm | anyone try the Telefinity phones in iax mode? Reliable? |
| 05:05.17 | bjohnson | I think Aginamu has one of the Telefinity |
| 05:06.43 | firestrm | bjohnson, i will look for him to come online and ask.. thanks again :) |
| 05:07.31 | bjohnson | np |
| 05:09.24 | firestrm | ugg.. voxilla only ships UPS.. that rules them out.. bummer.. good price too.. |
| 05:10.07 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 05:10.17 | jerlique | Parsing '/etc/asterisk/queues.conf': Found |
| 05:10.17 | jerlique | <PROTECTED> |
| 05:10.17 | jerlique | <PROTECTED> |
| 05:10.18 | jerlique | <PROTECTED> |
| 05:10.18 | jerlique | <PROTECTED> |
| 05:10.19 | jerlique | <PROTECTED> |
| 05:10.20 | jerlique | <PROTECTED> |
| 05:10.22 | jerlique | <PROTECTED> |
| 05:10.24 | jerlique | <PROTECTED> |
| 05:10.25 | h3x | NormAst: the problem with hauling lines as tdm is you need echo cans |
| 05:10.26 | jerlique | ooops Sorry!!! |
| 05:10.38 | liquidno2 | paste buffer attack! |
| 05:10.40 | h3x | and its illegal to haul TDM local dialtone across LATA boundaries |
| 05:10.59 | firestrm | LATA? TDM? |
| 05:11.13 | h3x | tdm, basically not VoIP :P |
| 05:11.21 | shido6 | <PROTECTED> |
| 05:11.23 | implicit | TDM == time division multiplexing |
| 05:11.23 | h3x | time division multiplexing |
| 05:11.30 | h3x | jinx |
| 05:11.35 | implicit | :) |
| 05:11.51 | firestrm | what is lata? |
| 05:12.00 | firestrm | sounds like a car :) |
| 05:12.04 | implicit | it is |
| 05:12.04 | h3x | local something transport area |
| 05:12.06 | h3x | its a group of area codes |
| 05:12.23 | *** join/#asterisk ionix (ionix@MTL-HSE-ppp186113.qc.sympatico.ca) |
| 05:12.31 | firestrm | so its illegal to transport a phone call over area codes? |
| 05:12.32 | ionix | Hey, how do I delay a call in outgoing to be placed ? |
| 05:12.39 | implicit | firestrm: not quite |
| 05:12.45 | ionix | I am trying to make a callback AGI |
| 05:12.51 | h3x | i said group of area codes. |
| 05:13.06 | jerlique | Is it normal for asterisk to ignore foney applications and not give errors, eg exten => s,4,Fake() |
| 05:13.12 | h3x | its very illegal to haul local dialtone T1s across lata boundaries |
| 05:13.16 | h3x | is my point |
| 05:13.20 | h3x | without converting it to voip first |
| 05:13.23 | liquidno2 | Local Transport and Access Area |
| 05:13.25 | h3x | at the very least |
| 05:13.33 | h3x | even then you are in a shitload of trouble if you pass the caller id through |
| 05:13.39 | h3x | yeah thats it |
| 05:14.13 | firestrm | why would they be so concerned about that? |
| 05:14.24 | h3x | because its long distance revenue bypass |
| 05:14.29 | ionix | fcc regulation firestrm |
| 05:14.30 | liquidno2 | LATA's are Bells life blood |
| 05:14.31 | h3x | for not only telcos but the government |
| 05:14.34 | firestrm | so is voip |
| 05:14.39 | liquidno2 | they get to charge more to cross them |
| 05:14.43 | liquidno2 | and they love it |
| 05:14.46 | h3x | NormAst, IXCs charge to cross them |
| 05:14.49 | h3x | i mean NO |
| 05:14.51 | h3x | not normast |
| 05:14.53 | h3x | stupid nick completion |
| 05:15.23 | h3x | LECs transport inside of the lata |
| 05:15.27 | NormAst | not in canada. |
| 05:15.34 | firestrm | so they must be pissed about VOIP then.. it does the same thing effectivly |
| 05:15.57 | h3x | well canada is different, but i suspect its worse |
| 05:16.03 | bjohnson | hehe |
| 05:16.15 | bjohnson | I always suspected it couldn't be worse than US |
| 05:16.27 | firestrm | h3x, not worse, just taxed more |
| 05:16.28 | h3x | well its all run by canada bell |
| 05:16.41 | h3x | or bell canada as us non frogs would say hehe |
| 05:16.42 | bjohnson | not exactly |
| 05:16.52 | firestrm | h3x, you can do anything you want in canada, as long as you pay a tax on ut |
| 05:17.02 | liquidno2 | There was interesting question that Cavuto asked the CEO of AT&T the other day... |
| 05:17.02 | firestrm | s/ut/it |
| 05:17.05 | h3x | they have clecs (a few) and ixc's but not a whole damn lot |
| 05:17.09 | bjohnson | crtc .. Bell is a huge influence |
| 05:17.16 | liquidno2 | about if it was the right decision to spin the bells off |
| 05:17.23 | h3x | the other problem is moving tdm calls around with no carrier license |
| 05:17.26 | h3x | in general |
| 05:20.50 | NormAst | yea... The CRTC tries to help. |
| 05:21.04 | h3x | the fcc sucks. |
| 05:21.06 | h3x | we're so screwed soon |
| 05:21.54 | Juggie | "ohhh canada... our home and native land...." |
| 05:21.55 | *** join/#asterisk sandjunkie (~trilluser@66-55-197-254.gwi.net) |
| 05:22.02 | implicit | hehe |
| 05:22.15 | h3x | they are gonna let all these emerging voip providers operate under this clout of "enhanced service provider" for a little while and then fuck everybody up requiring a carrier license |
| 05:22.41 | implicit | h3x: then people will go to canada and run the same shit |
| 05:22.42 | h3x | then the state puc's will get slammed with applications and wont be able to fulfill them for 8-14 months |
| 05:22.51 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 05:23.02 | h3x | and meanwhile they have to cease operating their businesses and lose all their customers |
| 05:23.12 | h3x | Qwest has already halted my order of two tdm DS3s |
| 05:23.18 | implicit | internet is internet |
| 05:23.20 | h3x | because they are under fire from the FCC for selling TDM to voip providers |
| 05:23.37 | implicit | is that illegal?? |
| 05:23.40 | h3x | facing a $1.2M fine per occurance due to the AT&T callvantage shit |
| 05:23.50 | h3x | its not illegal, but the way it gets billed is i guess |
| 05:23.58 | implicit | how is it getting billed? |
| 05:24.01 | h3x | the problem is their contract has to be redone and now i have to wait 4 or 5 months |
| 05:24.21 | h3x | apparently they aren't dealing with the tax and usf issues correctly |
| 05:24.23 | AgiNamu | <PROTECTED> |
| 05:24.28 | AgiNamu | And it does not take the branch |
| 05:24.31 | AgiNamu | Why? |
| 05:24.32 | brc_ | k42 |
| 05:24.34 | brc_ | h3x, |
| 05:24.34 | brc_ | dude |
| 05:24.36 | brc_ | WHATUP! |
| 05:24.38 | h3x | hi |
| 05:24.40 | h3x | not much |
| 05:24.42 | implicit | h3x: i think there are just loopholes :) |
| 05:24.46 | h3x | probably going to sell my data center. |
| 05:24.48 | brc_ | ELI SUCKXORZ |
| 05:24.52 | implicit | h3x: that they don't like |
| 05:24.58 | h3x | they will only take calls from me as voip now, and their voip product dosent really exist yet |
| 05:25.01 | AgiNamu | Do I need to do GotoIf($[Len(${ACCOUNTCODE}) != 10]?50)? |
| 05:25.02 | brc_ | they are over charging for ld |
| 05:25.10 | AgiNamu | All the samples for GotoIf are GotoIf($.... |
| 05:25.13 | AgiNamu | What's the $ for? |
| 05:25.16 | brc_ | uh |
| 05:25.16 | brc_ | dude |
| 05:25.19 | brc_ | a variable? |
| 05:25.21 | h3x | and its the same shitty sonus gateways that global crossing is using |
| 05:25.22 | brc_ | oh |
| 05:25.23 | brc_ | wait |
| 05:25.25 | brc_ | an expression |
| 05:25.28 | AgiNamu | So I have to do that |
| 05:25.32 | brc_ | $[...] means eval the ... |
| 05:25.34 | h3x | brc_: well "i said that would happen" |
| 05:25.36 | AgiNamu | OH ok |
| 05:25.38 | AgiNamu | I thought it was only for vars. |
| 05:25.39 | brc_ | did not |
| 05:25.39 | AgiNamu | thanks |
| 05:25.44 | h3x | yeah i did |
| 05:25.49 | brc_ | nuhu |
| 05:25.51 | h3x | i said that ive never seen a clec bill long distance correctly |
| 05:25.51 | h3x | ever |
| 05:26.27 | jerlique | hey is it ok to paste an 6 line error in here? |
| 05:26.37 | brc_ | sure |
| 05:26.39 | jerlique | <PROTECTED> |
| 05:26.39 | jerlique | <PROTECTED> |
| 05:26.39 | jerlique | Ouch ... error while writing audio data: : Broken pipe |
| 05:26.39 | jerlique | Floating exception (core dumped) |
| 05:26.39 | jerlique | iax# Junk at the beginning 49443303 |
| 05:26.39 | jerlique | Warning, flexibel rate not heavily tested! |
| 05:26.41 | brc_ | only after 10 though |
| 05:26.45 | NormAst | not the limit is 5.. |
| 05:26.46 | brc_ | if it's busy always use |
| 05:26.47 | brc_ | ~paste |
| 05:26.48 | jbot | [paste] see http://paste.husk.org |
| 05:26.49 | brc_ | ~pastebin |
| 05:26.50 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 05:26.58 | h3x | dude |
| 05:27.04 | h3x | you could have typed two more characters |
| 05:27.08 | h3x | and just said 'pastebin.ca' |
| 05:27.10 | *** join/#asterisk Alric (~nbowyer@64.6.45.2) |
| 05:27.18 | brc_ | yes |
| 05:27.19 | brc_ | I know |
| 05:27.25 | h3x | haha |
| 05:27.26 | AgiNamu | no way dude? |
| 05:27.30 | bjohnson | firestrm: ^^ |
| 05:27.31 | jerlique | hehehe. any idea while asterisk is core dumping? |
| 05:27.34 | brc_ | but it was faster to just use the bot |
| 05:27.38 | Nugget | you guys bitching about the paste is now twice the number of lines as the paste. |
| 05:27.43 | h3x | so im getting fed up with all this bullshit |
| 05:27.45 | brc_ | but of course |
| 05:27.49 | h3x | im going to sell my datacenter |
| 05:27.50 | AgiNamu | nugget, fuck, you're just making it worse. |
| 05:28.00 | implicit | Nugget: lol |
| 05:28.03 | Nugget | yeah but I use all lower-case letters so it's not as bad. |
| 05:28.04 | implicit | h3x: good |
| 05:28.06 | h3x | global crossing quit selling tdm voice |
| 05:28.07 | AgiNamu | we're gonna turn this into a "Help me start a VoIP company step-by-step" posted to the -docs list :) |
| 05:28.13 | h3x | mci is a pain in the fucking ass |
| 05:28.28 | h3x | you cant give mci enough traffic unless yer doing international calling afghanistan every day to run up a bill and meet their minimums |
| 05:28.41 | Nugget | AgiNamu: where "Help" means "just tell me what to type, I'm in a hurry" :) |
| 05:28.53 | AgiNamu | LOL |
| 05:28.57 | h3x | and qwest wants me to use their shitty ass voip which isnt even tested well yet and they dont offer toll free |
| 05:29.00 | AgiNamu | Afghanistan? That's not expensive. |
| 05:29.07 | h3x | plus it only supports g.729 so thats useless |
| 05:29.13 | brc_ | hahahah |
| 05:29.15 | brc_ | yep |
| 05:29.30 | AgiNamu | Cuba is expensive. |
| 05:29.33 | h3x | and i had so many problems with GX and their crappy ass sonus gateways |
| 05:29.40 | AgiNamu | or Inmarsat |
| 05:29.41 | bjohnson | Cuba has cheap cigars |
| 05:29.46 | h3x | i dont even wanna try qwest's |
| 05:29.47 | bjohnson | mmmm |
| 05:29.48 | implicit | AgiNamu: actually afghanistan is expensive, even though cuba is more expensive :) |
| 05:29.50 | h3x | well |
| 05:29.59 | h3x | cuba is cheap if you route it through a carrier in like, romania |
| 05:30.03 | AgiNamu | actually, non-sat, Cuba is my most expensive route |
| 05:30.05 | AgiNamu | and its uscks |
| 05:30.09 | AgiNamu | like 20% asr |
| 05:30.15 | AgiNamu | and HUGE PDD. and lots of FAS |
| 05:30.31 | bjohnson | and good cigars |
| 05:30.38 | bjohnson | mmmmmmmmmmmm |
| 05:30.41 | AgiNamu | like, thru a guy with a cellsocket and a direcway connection? |
| 05:30.47 | h3x | hahahah |
| 05:30.49 | h3x | hahahahhahahhaahahhah |
| 05:30.51 | AgiNamu | Cuba is like 40 cents , in country, with a decent plan |
| 05:31.05 | h3x | rotflmao |
| 05:31.10 | h3x | dude thats so funny |
| 05:31.17 | h3x | sell a "BFE VoIP Starter Kit" on ebay |
| 05:31.21 | AgiNamu | lol |
| 05:31.26 | harryvv | Talking about afganistan I was relieved that the passanger manifest of the crashed 727 did not have somone I knew on board. My highschool classmate from years ago is is in afganstan as a peace corp worker. |
| 05:31.30 | h3x | gsm cell phone, cell socket, direcway |
| 05:31.32 | AgiNamu | Oh wait, we can just use Bellster! haha |
| 05:31.42 | h3x | and dot matrix printer and some excel spreadsheet to bill people |
| 05:31.44 | harryvv | I know somone from my church there who is installing repeaters in afganistan |
| 05:31.50 | AgiNamu | h3x, with a 20 cent margin , I'd go for it if castro wouldnt castrate me if caught |
| 05:32.04 | AgiNamu | I *almost* did that in Guatemal |
| 05:32.04 | h3x | dude thats a great idea though |
| 05:32.05 | h3x | i mean |
| 05:32.09 | h3x | theres a E1 -> GSM box out there |
| 05:32.10 | AgiNamu | Buy an E1, slap ADSL on it |
| 05:32.17 | AgiNamu | Buy locak for 2 cents, sell for 6 |
| 05:32.22 | h3x | you stuff it with 30 SIM cards |
| 05:32.23 | AgiNamu | I'd get as many minutes as I want |
| 05:32.42 | AgiNamu | h3x, you aren't shitting me? |
| 05:32.46 | AgiNamu | Where can I buy such a device? |
| 05:32.52 | h3x | no really theres a couple brands of those |
| 05:32.54 | AgiNamu | I'd DEFINATLEY hit it if I could just pop in SIM cards |
| 05:32.57 | h3x | and you put a big antenna on it |
| 05:33.10 | AgiNamu | I did the numbers, I'd make $10K a month easy |
| 05:33.17 | h3x | http://www.pulsewan.com/gsm_t1e1_1.htm |
| 05:33.19 | AgiNamu | so long their telecom agency doesnt bust my ass |
| 05:33.20 | h3x | theres one |
| 05:33.42 | h3x | wow |
| 05:33.43 | AgiNamu | oh of course, their cell phones here in guatemala |
| 05:33.44 | h3x | they have gsm to voip now |
| 05:33.50 | AgiNamu | are MORE epxneisve than calling from TDM in USA to a cell |
| 05:34.04 | AgiNamu | how much are these E1->GSM thingS? |
| 05:34.09 | h3x | i have this great idea to go to jail fast |
| 05:34.13 | h3x | i donno |
| 05:34.18 | h3x | quite a bit |
| 05:34.23 | h3x | so my idea is |
| 05:34.31 | h3x | have a couple asterisk boxes call each other |
| 05:34.36 | h3x | and every 30 to 60 seconds or whatever |
| 05:34.39 | h3x | well they make a call both ways |
| 05:34.42 | h3x | and dont supervise it |
| 05:34.52 | h3x | and every so often it makes a new call and hangs up the old one |
| 05:35.04 | h3x | it probably wouldnt take too long til you got a goomba killing you though |
| 05:35.09 | AgiNamu | sounds like bellsta |
| 05:35.33 | h3x | sounds like napster |
| 05:35.45 | liquidno2 | well they changed the name |
| 05:35.50 | liquidno2 | outfwd |
| 05:35.57 | liquidno2 | I am curious to know why |
| 05:36.33 | AgiNamu | "yo yo, 'dis be P-U-tothevizzle. big up yourself and go out dere and connect up all ya's asstricks" |
| 05:36.36 | h3x | um |
| 05:36.38 | h3x | hrm |
| 05:36.42 | h3x | i guess i should set up vegas with this shit |
| 05:36.56 | AgiNamu | changed what name? bellster? |
| 05:37.01 | liquidno2 | yes |
| 05:37.01 | AgiNamu | Cause bellsouth sued him? |
| 05:37.10 | liquidno2 | bellster changed their name to outfwd |
| 05:37.12 | AgiNamu | he should use 1% of the money he makes off of VON and fight it :D |
| 05:37.19 | AgiNamu | which is an insane fucking amount btw |
| 05:37.19 | liquidno2 | no idea why the name chagned |
| 05:37.32 | AgiNamu | is that sarcasm? |
| 05:37.33 | silik0n | fuck bellsouth |
| 05:37.39 | AgiNamu | fuck bellsta |
| 05:37.40 | silik0n | they are a buncha tools |
| 05:37.45 | liquidno2 | but they have removed all reference to bellster from the site |
| 05:38.09 | AgiNamu | surprised jeff didn't use that for publicity |
| 05:38.12 | h3x | hAha it uses credits |
| 05:38.13 | h3x | this is like |
| 05:38.14 | AgiNamu | "look look, we're beeing repressed" |
| 05:38.20 | h3x | man |
| 05:38.26 | h3x | i could rack up some serious credits with two pris |
| 05:38.26 | AgiNamu | OK, so I open my local line, and people do 4 cent calls |
| 05:38.28 | liquidno2 | help help i am being oppressed |
| 05:38.37 | AgiNamu | and then I can do some .6 cent calls to USA .... WTF? |
| 05:38.40 | h3x | im gonna send all my non rboc traffic to it |
| 05:38.41 | h3x | hahahahah |
| 05:38.59 | AgiNamu | bellster == marketing |
| 05:39.11 | liquidno2 | it isn't a bad idea really.. |
| 05:39.19 | AgiNamu | or a fun project for some kid to do between popping zits and reading slashdot |
| 05:39.28 | liquidno2 | considering that people can make free "local" calls to their area |
| 05:39.40 | AgiNamu | uh, the quotes belong around the "can" and "free" |
| 05:40.07 | liquidno2 | if I call anything in my lata... it is free |
| 05:40.08 | AgiNamu | pulver knows damn well it don't work like that. he just likes doing stuff to make more money. in that way, its nothing special :) |
| 05:40.15 | AgiNamu | it's free? |
| 05:40.48 | liquidno2 | unless you are being analytical and infering the cost of the line as not being a free call. |
| 05:41.13 | liquidno2 | I am speaking strickly perminute charges for the call |
| 05:41.53 | AgiNamu | why do I pay like .x cents then for every call I make? |
| 05:41.56 | h3x | PhoneFree to start charging for calls |
| 05:41.56 | h3x | January 24, 2001 |
| 05:42.03 | h3x | That makes as much sense as netzero |
| 05:42.05 | h3x | hahahahha |
| 05:42.27 | Nugget | I pay for my ssl certificates from freessl.com, too. teh intarweb is such a scam. |
| 05:42.37 | AgiNamu | from .5 to 1.something cents for usa? |
| 05:43.24 | h3x | reciprocal compenstation |
| 05:43.25 | liquidno2 | I pay nothing for "local" calls |
| 05:43.31 | AgiNamu | liquidno2, if that was true then I'll be a millionare. I'll pay what, $30 to get "Free" calling to an area? even at 1/3 usage, that's 14K minutes |
| 05:43.31 | liquidno2 | right |
| 05:43.32 | NormAst | me too. |
| 05:43.56 | Nugget | everyone should take this time to get enum e164.org support working in their asterisk dialplan. |
| 05:44.06 | AgiNamu | nugget, does it support IAX2? |
| 05:44.12 | Nugget | it's protocol-independent. |
| 05:44.16 | AgiNamu | so if you leave your phone on 24/7, does that work? |
| 05:44.35 | AgiNamu | run up 40K+ minutes and the telco doesnt care? |
| 05:44.40 | Nugget | it's a way to map PSTN numbers to direct voip addresses (sip or iax) |
| 05:44.53 | liquidno2 | AgiNamu: I can pick up the phone and call the otherside of town (which is still in the same LATA) put down the phone... and come back days later and it doesn't cost me anything. |
| 05:44.55 | AgiNamu | nugget, i'll do that .we're finally lighting up customers |
| 05:44.59 | liquidno2 | but it does tie up my line. |
| 05:45.04 | NormAst | they will care.. |
| 05:45.06 | Nugget | so if you dial +1 512 538-0508, your asterisk will be smart enough to know that you can just as easily dial sip:nugget@slacker.com |
| 05:45.09 | AgiNamu | ok, im indirectly sayign that "it's not free" |
| 05:45.16 | AgiNamu | you pay MUCH more than you'd pay normally |
| 05:45.20 | Nugget | so instead of paying your provider for the PSTN routing, you just hit me directly |
| 05:45.28 | AgiNamu | by paying a whatever montly fee, that includes 1000 minutes. or 5000 minutes. |
| 05:45.45 | AgiNamu | so it's free until it's abused. and then its not free, and pointless. |
| 05:46.04 | AgiNamu | so, in the case of bellster, if im gonna get REAL usage out if it, and start doing 5K+ minutes a month, I somehow doubt the telco is just gonna smile |
| 05:46.08 | liquidno2 | AgiNamu: your ambiguity is appreciated. |
| 05:46.44 | AgiNamu | all the telcos do that. Packet8 ran into the problems since people brought their phones over to guatemala |
| 05:46.54 | PTG123 | Hey does anyone know how to print a text file to a diff on unix in an easy way? |
| 05:46.55 | Nugget | AgiNamu: http://slacker.com/~nugget/asterisk11.php |
| 05:46.58 | AgiNamu | and used their "free" "unlimited" calls a LOT |
| 05:47.17 | AgiNamu | now, if you're not in the states, they charge you $10 more, and limit your calling to 1000m inutes |
| 05:47.20 | AgiNamu | and then charge you... |
| 05:47.34 | liquidno2 | and this discussions point is what>? |
| 05:47.55 | AgiNamu | is that bellster doesnt work |
| 05:48.07 | AgiNamu | since it exploits a fluke in marketing |
| 05:48.17 | liquidno2 | I suppose that is up to intreprtation |
| 05:48.17 | AgiNamu | and that nothing is free |
| 05:48.30 | AgiNamu | you pay $40 a month for a phone line, then are happy you can make a few "free" calls? |
| 05:48.46 | AgiNamu | and btw, that doesnt apply to a lot of people |
| 05:48.48 | jerlique | Hi - trying to report a bug and was wondering how to get the latest cutting egde code. I can CVS release version but not sure how to get latest?? |
| 05:49.02 | AgiNamu | cvs checkout asterisk ? |
| 05:49.07 | AgiNamu | www.asterisk.org hit download |
| 05:49.33 | jerlique | 'cvs checkout zaptel libpri asterisk ' is this the one? |
| 05:49.41 | jerlique | (thats where I am..) |
| 05:49.41 | AgiNamu | yhea, the cvs instructions |
| 05:49.57 | liquidno2 | I am of the theory that just because you don't like the way something works... doesn't mean it is necessarily broken either. |
| 05:50.07 | jerlique | so if you dont specify the -r v1-0 it automatically gets the latest code? |
| 05:50.17 | AgiNamu | thats right |
| 05:50.21 | jerlique | thanks. |
| 05:50.31 | AgiNamu | liq, referring to what? |
| 05:50.39 | liquidno2 | bellster |
| 05:51.05 | AgiNamu | No, I dont like it because of the security impliciations, the impracticallity for MOST people, and that it's lame (quality wise) |
| 05:51.10 | jerlique | me? asterisk crashing when going into queues... |
| 05:51.14 | *** join/#asterisk PakiPenguin (~uppal@202.176.254.1) |
| 05:51.17 | AgiNamu | the fact that your telco might not be impressed doesnt phase me nearly as much |
| 05:51.51 | AgiNamu | in fact, I've half a mind to demonstrate this first hand |
| 05:52.02 | liquidno2 | feel free |
| 05:52.21 | AgiNamu | yea, i might. I have ot get asterisk /iax on a laptop |
| 05:52.26 | AgiNamu | and find an open hotspot |
| 05:52.29 | AgiNamu | i dont wanna get arrested |
| 05:52.37 | AgiNamu | but um, it'll happen sooner or later |
| 05:52.55 | AgiNamu | a few calls go out to people threatening, CallerID/ani shows YOUr number |
| 05:53.31 | AgiNamu | and really... how much arey ou gonna save??!? damn , i mean... if i want a shitty call, i'll use skype or something |
| 05:53.42 | AgiNamu | even with TDM, prices are cheap all around the world. |
| 05:53.55 | AgiNamu | except where they are expensive, and bellster wont work there. |
| 05:53.59 | AgiNamu | :P |
| 05:54.09 | AgiNamu | even so, it's brilliant marketing |
| 05:54.46 | AgiNamu | anywyas, gonna watch Dune |
| 05:54.48 | AgiNamu | night all |
| 05:55.04 | liquidno2 | is that on TV? |
| 05:55.06 | Nugget | I run asterisk on my powerbook because I have better luck with iax2 getting out through shitty hotel internet systems. |
| 05:55.08 | AgiNamu | no, DivX |
| 05:55.10 | AgiNamu | burned to DVD |
| 05:55.15 | AgiNamu | playback in a nice Philips DVP 642 |
| 05:55.18 | Nugget | I run a local x-lite talking to the local asterisk which routes to my real server. |
| 05:55.21 | liquidno2 | thought my Tivo missed it |
| 05:55.26 | AgiNamu | x-lite blows :) |
| 05:55.31 | AgiNamu | liquid, im in guatemala anyways... |
| 05:55.34 | Nugget | it's the best option in os x, though. |
| 05:55.37 | AgiNamu | so i doubt we've got same programming schedules |
| 05:55.37 | liquidno2 | ah |
| 05:55.38 | implicit | AgiNamu: yeah |
| 05:55.49 | AgiNamu | x-lite gives me carpal when i tr to configure it |
| 05:56.01 | AgiNamu | idiots thinking we wnana pretend we have a freaking cellphone |
| 05:56.13 | harryvv | agi, what? |
| 05:56.13 | Nugget | yeah, it blows goats to configre. |
| 05:56.17 | jerlique | is it normal for mpg123 to go into a full CPU utilisation when I stop now or stop when conv in * |
| 05:56.23 | liquidno2 | well its official AMP sucks |
| 05:56.34 | AgiNamu | harryvv its programmers who think they are UI/UX expers |
| 05:56.40 | AgiNamu | and come up with a "cute" interface |
| 05:56.51 | AgiNamu | and then apply it to everything. so instead of having a nice tabbed view |
| 05:56.53 | AgiNamu | and so on |
| 05:57.01 | harryvv | agi, I see alot of arogance in dev channels |
| 05:57.04 | AgiNamu | you've got their idiotic Cellphone-style menus to configture anything |
| 05:57.44 | AgiNamu | yea, it goes hand-in-hand with development. esp. if it's something like asm.. or even C |
| 05:57.54 | harryvv | Any recomends for a good overall wifisip phone? |
| 05:58.00 | AgiNamu | people just can help feeling they're the most fucking brilliant person in the world to ever write code :) |
| 05:58.10 | Nugget | yeah, x-lite is tragic. it's got the whole "let's pretend this isn't a real computer with a keyboard and a mouse -- instead lets mimic a shitty cell phone interface down the last little annoying detail because obviously customers want to use a menu they're familiar with" |
| 05:58.25 | AgiNamu | even if they're writing the most hideous thing you've seen.. sheesh. I've heard self-proclaimed experts in C# say "C# is javascript and VB mixed" |
| 05:58.33 | Nugget | harryvv: do NOT buy the zyxel/pulver phone. |
| 05:58.37 | AgiNamu | nugget: damn straight |
| 05:58.55 | harryvv | Nugger, yea heard somethign about those phones. |
| 05:58.58 | AgiNamu | these people also usually say C# doesnt have pointers or sanything |
| 05:58.59 | AgiNamu | anyways |
| 05:59.02 | Nugget | they're awful. |
| 05:59.08 | AgiNamu | If FireFly didnt crash every 2 minutes |
| 05:59.13 | Nugget | there are two new wifisip phones on the market, but I haven't used either. |
| 05:59.23 | jerlique | happyh to say that current fixes the queue crash problem I've been trying to fix for the last 5 hours :) |
| 05:59.23 | AgiNamu | and if it did 2-way IAX2 audio, I'd use it :D |
| 05:59.28 | Nugget | someone else can be the pioneer with them. I got burned on the zyxel. |
| 05:59.33 | harryvv | whats better then xlite? |
| 05:59.40 | AgiNamu | FireFly rocks... when it works. |
| 05:59.48 | AgiNamu | and there are some commercial phones that look a lot better. |
| 05:59.56 | harryvv | i like xlite. 100% reliable so far |
| 06:00.05 | AgiNamu | anyways, im audi 5000 |
| 06:00.08 | moonwick | Nugget: I'm willing to bet it's still better than my analog uniden 900MHz phone. heh. |
| 06:00.19 | Nugget | once it's configured, x-lite isn't too awful. I hate the few areas they've intentionally crippled it though. |
| 06:00.35 | Nugget | I have my cow-orkers using sjphone. it's uglier, but it does more. |
| 06:00.42 | harryvv | nugget, what the next one pro? |
| 06:00.42 | Nugget | hah moon. |
| 06:00.44 | harryvv | how much |
| 06:01.21 | Nugget | I'd kill for a wireless headset that integrated with my cisco 7960. |
| 06:01.41 | moonwick | I'd kill for a cisco 7960 :) |
| 06:01.50 | Alric | Doesn't the CS50 do that? |
| 06:01.54 | PakiPenguin | Nugget, make one |
| 06:02.00 | Nugget | I'm too stupid to make one. |
| 06:02.07 | Nugget | all I know how to do is bitch on irc. |
| 06:02.08 | Alric | Plantronics |
| 06:02.25 | PakiPenguin | plantronics has AWESOME headphone |
| 06:02.29 | Nugget | Alric: you still can't answer/hang up without some whipdick handset lifter hardware. |
| 06:02.59 | Alric | Nugget: Yeah, its around $250 total, last I checked. |
| 06:03.23 | implicit | harryvv: because they are for developers :) |
| 06:03.31 | Alric | Still. Wireless w/ whipdick handset lifter > No wireless :) |
| 06:04.37 | cypromis | try GN Netcom |
| 06:04.48 | cypromis | they have nice headsets as well and Iprefer them over plantronics |
| 06:05.00 | liquidno2 | bluetooth is great... right up till the battery goes dead |
| 06:05.16 | cypromis | depends |
| 06:05.25 | cypromis | use a big battery and you allways have a spare weapon |
| 06:05.26 | cypromis | ;) |
| 06:06.33 | Alric | I hate HP laptops. |
| 06:06.41 | shido6 | why? |
| 06:07.07 | firestrm | do you really need a reason :P |
| 06:07.44 | *** join/#asterisk dwC- (~dwc@69.42.74.4) |
| 06:08.14 | dwC- | Hello |
| 06:08.17 | firestrm | hi |
| 06:08.28 | liquidno2 | pi = 3.141592653 |
| 06:08.36 | Alric | Why? Battery life: 45 minutes tops. This will be the 3rd time I've replaced a battery this year. Powers off for no reason. DVD player stopped working after 1 year. Touchpad is all messed up. |
| 06:09.05 | jerlique | paki: are the plantronics usb headsets compatible with linux/freebsd? |
| 06:09.19 | Alric | And apparently I'm not alone in my problems :) |
| 06:09.22 | firestrm | Alric, its a feature if you really think hard enough about it :) |
| 06:09.36 | dwC- | I have a moh extension setup but when i call it i can hear the music for about 1/2 second then i get silence until the pbx drops the call in 15000ms. any ideas whats going on here? |
| 06:09.44 | Alric | firestrm: A feature to make it go airborne out my 2nd story window faster...? |
| 06:09.44 | PakiPenguin | jerlique, i dont have one :p i have the standard ones |
| 06:09.56 | liquidno2 | if my boss would have just bought me the emachines laptop I wanted... none of this would have happened |
| 06:10.08 | liquidno2 | that widescreen jobby is ACE |
| 06:10.13 | PakiPenguin | emachines suck! |
| 06:10.20 | liquidno2 | not this laptop |
| 06:10.32 | harryvv | echeaphardware? |
| 06:10.33 | harryvv | ;) |
| 06:10.36 | Alric | I thought eMachines were only known for crappy low cost hardware? |
| 06:10.39 | Qwell | all emachines suck, no offense |
| 06:10.41 | liquidno2 | mate has one... it is ROCK SOLID |
| 06:10.44 | firestrm | lol, no 45 min battery keeps you from using it until 3:00 am and waking up tired for work, no dvd means you dont waste your time on p0rn :P |
| 06:10.49 | moonwick | dwC-: you're telling it to play the "Best of Silence" album? |
| 06:10.52 | liquidno2 | Here let me dig the link |
| 06:11.14 | dwC- | moonwick. yeah it almost seems that way lol |
| 06:11.19 | Alric | firestrm: Its a laptop from my college out of a promo. My smallest class is 50 minutes. This doesn't work too well... :) |
| 06:11.38 | liquidno2 | damn... looks like they discontinued them. |
| 06:11.52 | moonwick | Alric: so, um, shut it down for the last five minutes and learn something. :) |
| 06:11.54 | liquidno2 | these laptops were more expensive than the average laptop |
| 06:11.58 | liquidno2 | $1600USD |
| 06:12.00 | firestrm | Alric, kinda like its 1 shy of a sixpack |
| 06:12.12 | PakiPenguin | by the way , i am going to buy a new laptop pretty soon , what do you suggest , centrino or a p4? |
| 06:12.15 | liquidno2 | AMD64 3200, ATI 9600 mobile. |
| 06:12.23 | Nugget | PakiPenguin: PowerBook G4. |
| 06:12.25 | liquidno2 | it was ace |
| 06:12.30 | Alric | So yeah, HP is on my No buy list. For a very long time. |
| 06:12.39 | liquidno2 | Dell |
| 06:12.46 | PakiPenguin | Nugget, i already have one :p |
| 06:12.47 | firestrm | Alric, i remember the bad old days with my toshiba 286.. i had to carry 6 batteries, one for each class |
| 06:12.48 | dwC- | moonwick. any idea what the problem would be? it goes to play the music then it changes its mind :9 |
| 06:12.49 | liquidno2 | buy nothing else for a laptop |
| 06:12.51 | Alric | Dell or Apple, depending on what world you want. |
| 06:13.03 | liquidno2 | I think it is the Insperion 8000 |
| 06:13.04 | moonwick | dwC-: heh, nope. sorry. never played with musiconhold |
| 06:13.05 | sskyles | Shit, my Mac will go 4.5 hours on a battery... What's wrong with these PC's anyway? |
| 06:13.24 | dwC- | moonwick. k thanks |
| 06:13.34 | moonwick | my inspiron 8600 gets 4 hours to a charge, easily |
| 06:13.43 | Juggie | is it possible to force a zaptel card onto a certain irq with module parmaters? or do i have to go into the bios |
| 06:13.48 | Alric | I would have gotten an 8600 :( |
| 06:13.56 | Alric | But I like the new D610s too. |
| 06:13.59 | firestrm | Alric, look for a compaq evo.. not the most powerful things on the planet, but nice features and bullet proof.. Im a pilot, i drag it through every airport on the contenent, and no problems .. yet |
| 06:14.01 | Alric | For work. |
| 06:14.02 | PakiPenguin | i've heard centrinos are better |
| 06:14.12 | Juggie | centrino's are way better. |
| 06:14.22 | PakiPenguin | firestrm, my boss has one evo n610c |
| 06:14.32 | firestrm | PakiPenguin, same one i have |
| 06:14.55 | liquidno2 | as a general rule... avoid P4s for laptops... unless you need a way to heat sandwiches while you are on the road |
| 06:15.14 | Alric | So true |
| 06:15.15 | Qwell | This from the guy who wants an amd64 laptop :p |
| 06:15.28 | PakiPenguin | lol |
| 06:15.29 | liquidno2 | Qwell: I didn't say I practice what I preach :) |
| 06:15.47 | firestrm | Qwell, dont hold your breath.. that would be a heat dissapation nightmare |
| 06:15.49 | liquidno2 | I really just want the laptop so I don't have to lug my shit to lan parties :) |
| 06:16.45 | ph3nix | where do you live |
| 06:16.56 | firestrm | liquidno2, i have an old portable kaypro that i want to mod into a lan party machine |
| 06:16.58 | liquidno2 | next door to the middle of nowhere. |
| 06:17.01 | PakiPenguin | i want a laptop, so i can work without killing my back |
| 06:17.25 | liquidno2 | ph3nix: southern Alabama |
| 06:17.36 | liquidno2 | aka: nextdoor to the middle of no where |
| 06:18.08 | *** join/#asterisk Nukemizer (Nuke@66.237.85.58.ptr.us.xo.net) |
| 06:18.20 | firestrm | liquidno2, no galina, alaska.. now thats next door to the middle of nowhere |
| 06:18.28 | PakiPenguin | are factory refurb. lapis good? |
| 06:18.38 | liquidno2 | firestrm: adak island |
| 06:18.51 | liquidno2 | that is officially the middle of nowhere |
| 06:18.56 | liquidno2 | it is documented :) |
| 06:19.16 | firestrm | and fly there |
| 06:19.18 | liquidno2 | it is in the chain |
| 06:19.25 | liquidno2 | it is a military install |
| 06:19.29 | liquidno2 | or at least it was |
| 06:19.33 | liquidno2 | for the Coast Guard |
| 06:19.36 | postel | no mobile user has great needs of computing power, i dont know a lot of mob users that would need to serve 45000/sec queries on a db running locally on the lap, or anybody that gonna do CAM on a lap, so if you're a *true* mob user (no serfing slashdot from your living room over wifi to your bedroom does NOT qualify) you just need light weight and battery power, cxentrino or mac for me |
| 06:19.53 | liquidno2 | they might have shut it down because they are phasing out Loran C stations |
| 06:20.09 | firestrm | postel, amen |
| 06:20.28 | postel | thanks, we accept donations in any currency by the door |
| 06:20.30 | harryvv | Liq, talking avionics or boating? |
| 06:20.44 | liquidno2 | harryvv: both actually. |
| 06:20.48 | firestrm | liquidno2, i was close to there last summer.., next summer i want to fly across the balfort sea to russia.. |
| 06:21.07 | liquidno2 | I know that Loran C stations are being shutdown all over the place |
| 06:21.21 | harryvv | liq, I am a previos avionics grad long ago. yea probebly no need for them. |
| 06:21.22 | liquidno2 | but then I have been out of the military for too long |
| 06:21.33 | harryvv | what branch |
| 06:21.37 | liquidno2 | USCG |
| 06:21.41 | harryvv | usaf |
| 06:21.52 | liquidno2 | nice... |
| 06:21.53 | harryvv | what was your job? |
| 06:21.54 | firestrm | postel, compaq evo, 610c 4 hour battery.. reasonable power.. nuff said.. |
| 06:21.58 | liquidno2 | I actualyl consider those |
| 06:22.08 | liquidno2 | harryvv: Aviation Electronics |
| 06:22.22 | harryvv | :) we donated our Sikorsky H-3 to you guys. |
| 06:22.42 | liquidno2 | I swear I never touched it! |
| 06:22.45 | liquidno2 | ;) |
| 06:22.46 | harryvv | hehe |
| 06:22.51 | harryvv | sure you did |
| 06:23.05 | harryvv | firestrm, not in it anymore? |
| 06:23.29 | harryvv | I miss the h-3 good aircraft |
| 06:23.33 | firestrm | ame needs to be renewed, but i still do avionics design work on the side.. |
| 06:23.41 | harryvv | Thats cool |
| 06:24.05 | liquidno2 | I haven't touched anything avionics in... god... man... 11 years... |
| 06:24.21 | harryvv | I think other then the h53 the h-3 was th first helicopter to have a kneeling nose gear. |
| 06:24.33 | Nugget | I haven't flown in over a year. :( |
| 06:24.54 | harryvv | and the first for airial refuling. it was the first to fly across the atlantic non stop from ny to ireland I think. |
| 06:25.03 | firestrm | Nugget, i would rather be casterated than not be able to fly |
| 06:25.21 | PakiPenguin | why are hp's business notebooks costly? :( |
| 06:25.40 | postel | so many ppl in avionics, the next * meeting would be in JFK by the looks of it, better yet OVER JFK :P |
| 06:25.43 | firestrm | PakiPenguin, to cover all the replacement laptops they will have to send you for warrenty |
| 06:26.06 | netsurfer | lol |
| 06:26.53 | firestrm | postel, look for me in CYVR, or CYCG |
| 06:27.08 | harryvv | yea, I got out of aviation in the early 1990s when aviation companies were going belly up left and right and 1 million aviation releated jobs were shoved into the unemployment line. |
| 06:27.36 | firestrm | harryvv, times have changed.. they are begging for good avionics ppl now |
| 06:29.02 | firestrm | harryvv, i had one company offer me more than im making now flying, and they would pay to renew my AME.. |
| 06:29.16 | harryvv | firestrm, where in the states? |
| 06:29.19 | harryvv | or in canada |
| 06:29.48 | firestrm | harryvv, canada.. kelowna flightcraft (or Kelowna Frightcraft as they are know to locals :) |
| 06:30.07 | harryvv | Ohh are you in in that area? |
| 06:30.23 | firestrm | harryvv, they do convair and 727 conversions |
| 06:30.34 | firestrm | harryvv, im in victoria.. |
| 06:30.43 | harryvv | okay im in vancouver |
| 06:30.44 | harryvv | ;) |
| 06:30.48 | firestrm | lol |
| 06:31.17 | netsurfer | next u guys will be saying u know eachother :oP |
| 06:31.24 | harryvv | yea right heeh |
| 06:31.53 | firestrm | harryvv, want to car pool (or maybe plane pool) to von in cali? |
| 06:31.58 | netsurfer | small world, I bumped into a neighbour in #debian one night |
| 06:32.06 | harryvv | fire, you fly there? |
| 06:32.08 | postel | well even if they dont whats some gallons of fuel to fly over between friends |
| 06:32.29 | firestrm | harryvv, if i can find enough to fill the plane.. 4 seats avail.. |
| 06:32.31 | dr342346 | OK question: i am using SIP SJPhone when i park a call it works but it hangs the sjphone up before spitting the park extension back to me any ideas |
| 06:32.34 | harryvv | firestrm msg you |
| 06:32.38 | firestrm | ok |
| 06:32.55 | netsurfer | lol wikked |
| 06:33.30 | netsurfer | im beginning to think I should have went to bed after make modules |
| 06:33.54 | netsurfer | f'ing bloated kernel on tiny processor |
| 06:34.26 | postel | recompile and strip it down |
| 06:34.29 | firestrm | netsurfer, or bloated processor on tiny kernel :) |
| 06:34.55 | Juggie | anyone know if i can force a zaptel card to a certain irq from within linux? or do i need to make the changes in the bios? |
| 06:35.58 | Qwell | firestrm: Whats von? |
| 06:36.31 | netsurfer | postel - I could do, but every time I try to streamline the kernel, I break * |
| 06:36.39 | riksta | Juggie: enable apic |
| 06:36.39 | *** join/#asterisk sskeks (~Soulz-@cm51.epsilon168.maxonline.com.sg) |
| 06:36.43 | postel | Juggie: a chinese monk i once knew used to say.. "Move the cards around on the PCIs" |
| 06:37.03 | harryvv | juggie, having problems? |
| 06:37.04 | riksta | postel: you're nocturnal dude! |
| 06:37.34 | postel | riksta: hey dude, i gave adm a try |
| 06:37.57 | postel | heh, not so bad |
| 06:38.21 | riksta | man i feel like shit i just woke up, and im hungoverrrr |
| 06:38.26 | riksta | i gotta go to CREWE today |
| 06:38.30 | postel | it works, there is some prob in the forked child that causes the second run onwards to give an empty dialog box |
| 06:38.30 | sskeks | alo all, my asterisk hdd crashed today, so wanted to be brave and install a asterisk gui, wanted to find out if the astguiclient is better or i forgot the name of the other one |
| 06:38.36 | sskeks | which is supposed to be better? |
| 06:38.48 | riksta | postel: empty dialog box for what |
| 06:38.54 | Juggie | harryvv, i just noticed the machine i built friday in its infinite wisdom has decided it should place, usb,gigabit ethernet and zaptel on irq9 |
| 06:38.55 | postel | for new call |
| 06:39.07 | riksta | postel: hmm, i don't think that happens here |
| 06:39.07 | Juggie | in the meantime leaving, 5,7,8,13 free |
| 06:39.20 | Juggie | which is causing irq misses |
| 06:39.31 | Juggie | so i was just wondering if i could force someway in linux |
| 06:39.35 | postel | riksta: it only happens on second/third call onwards, i'll have a better look on it |
| 06:39.38 | riksta | postel: is that from pressing the middle mouse on the tray icon, to paste a number, or just doing a new call from the context menu |
| 06:39.43 | Juggie | or if i have to make a visit to the lab on monday |
| 06:39.49 | postel | riksta: from the menu |
| 06:40.05 | riksta | postel: it could be that the dialog boxes get destroyed, recursively |
| 06:40.48 | riksta | postel: does the volume code work nicely for you |
| 06:40.54 | postel | riksta: i also think the sub-dem routine needs some work |
| 06:41.14 | riksta | yeah like i said i don't know much about that stuff |
| 06:41.54 | postel | riksta: well, i had the system running over arts and wasnt happy when aumix was trying to change things, i forced output to a diff channel (i got more than one soundcards) and all plays smoothly |
| 06:42.23 | riksta | ahh i see, i don't use arts, i'll have a look at that |
| 06:42.56 | postel | but yeah, if you grab the perl modules works as it should (almost) |
| 06:43.07 | sskyles | How can I find out exactly what area codes and exchanges are within my local calling area? |
| 06:43.54 | postel | and i have some trouble with forcing out to speakerphone, but after enabling telnet_level on the configs and rebooting the ciscos i still couldnt get a prompt, i'll have a look on that too |
| 06:44.19 | riksta | postel: sec |
| 06:44.26 | postel | i mean manually get a prompt |
| 06:44.42 | riksta | you need telnet_level: "2" |
| 06:44.45 | postel | i know |
| 06:44.55 | postel | thats what i did |
| 06:44.56 | riksta | ah, and still no prompt |
| 06:45.04 | postel | its just timeouts on me |
| 06:45.15 | riksta | did you set a phone_password though |
| 06:45.20 | postel | yeap |
| 06:45.36 | riksta | strange |
| 06:46.00 | riksta | so you can't test the DND n stuff then? anyway....i'm yet to write the idle detection, etc |
| 06:46.06 | postel | i set the prompt to "Go Away" some time ago, maybe its keeping me out following instructions :P |
| 06:46.28 | riksta | hehe |
| 06:46.58 | liquidno2 | anybody use usedistinctiveringdetection? I need some help ni understanding where I should put that |
| 06:47.16 | postel | yeah, DND i cannot test and speaker-out either, but the aumix controls and dialout works |
| 06:47.36 | riksta | great |
| 06:47.44 | riksta | well, at the moment, they are the key features, so im happy enough with that |
| 06:47.45 | liquidno2 | does that go in the channel definition for the fxs_ks channel I am creating it on? |
| 06:48.10 | postel | strange things is i got more than one ciscos and i cant telnet to either one, may be the cisco router they're on. i'll have a go over the configs or drop them on a DES managed switch |
| 06:48.49 | riksta | liquidno2: it goes in the zapata.conf under [channels] afaik |
| 06:49.15 | liquidno2 | riksta: yeah... but how do I define it for one fxs_ks channel but not the other? |
| 06:49.27 | riksta | signalling=fxo_ks group=1 ? |
| 06:49.34 | liquidno2 | do I just override the default of no on the one channel I want it on? |
| 06:49.52 | riksta | i think so, although i don't have any means to find out |
| 06:50.14 | liquidno2 | it is okay... I just always feel retarded when I muck around in zapata.conf |
| 06:50.32 | riksta | haha |
| 06:50.58 | postel | riksta: by the way i think its just fine for a .9 release, drop it out there, get some ppl on it and coder pool |
| 06:51.25 | riksta | i'm planning on releasing in the new few days, i just want to clean some stuff up and add a few more features |
| 06:52.22 | riksta | anyway need to shower, gotta get ready 4 work |
| 06:52.24 | riksta | *sigh* |
| 06:52.31 | postel | today!! |
| 06:52.36 | riksta | yes |
| 06:52.40 | postel | ouch!!! |
| 06:52.43 | riksta | and i have to travel to god damn crewe |
| 06:52.52 | riksta | because our office is without power |
| 06:53.26 | riksta | i can't believe i am doing this shit, for such a SHIT job |
| 06:53.29 | riksta | anyway, laters |
| 06:53.36 | postel | later |
| 06:56.40 | *** join/#asterisk jesse_132 (~jdandr2@12-203-179-57.client.insightBB.com) |
| 06:57.39 | jesse_132 | from my AGI-BIN (in python using pyst) I sometimes get : COMMAND: stream file finance "" 0 |
| 06:57.42 | jesse_132 | <PROTECTED> |
| 07:02.53 | jesse_132 | <PROTECTED> |
| 07:03.04 | jesse_132 | changed to os.popen and I'm ok |
| 07:12.47 | riksta | postel, if you could hack away a bit at ADM it'd be greatly appreciated :) gotta dash |
| 07:13.53 | postel | riksta: you know my mail, pass me on the pre-release you're cooking and i have a look, i'll see what i can do with the empty window im getting |
| 07:14.25 | riksta | i havent made any changes since i sent it to you, been busy |
| 07:14.42 | postel | and drop the ciscos on the managed switch so telnet can handle DND and speaker-out |
| 07:15.17 | riksta | good luck, really gotta run |
| 07:15.25 | postel | k, later |
| 07:22.17 | tzafrir | good morning |
| 07:33.12 | brc_ | http://forum.woot.com/forum/viewtopic.php?t=277&postdays=0&postorder=asc&start=1420 |
| 07:49.02 | jerlique | why is it that I only get MOH, when I talk or blow into the mic? |
| 07:49.44 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.we.client2.attbi.com) |
| 07:49.49 | Wi_Fi | Ouch ... error while writing audio data: : Broken pipe |
| 07:49.50 | Alric | Silence suppression? |
| 07:49.56 | Wi_Fi | dbl ouch |
| 07:50.06 | Wi_Fi | my pbx broke |
| 07:50.24 | netsurfer | Wi_Fi - sounds like u need a plumber ;) |
| 07:51.25 | Wi_Fi | sure sure |
| 07:51.26 | Wi_Fi | hehe |
| 07:51.53 | jerlique | wifi - I spent 5 hours on this issue today! |
| 07:52.05 | jerlique | I upgraded to latest code to resolve it. |
| 07:52.18 | Wi_Fi | holy crap |
| 07:52.24 | Wi_Fi | it fixed itself |
| 07:52.58 | jerlique | no, the latest (cutting edge) code resolve it. My problem was specific to calls being answered into queues |
| 07:53.06 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 07:53.14 | jerlique | alric: why would silence supression stop hold music? |
| 07:53.29 | Pulu | is iaxtel not working anymore? |
| 07:53.40 | Pulu | i get alot of failed registrations, like 40% of them are rejected |
| 07:54.02 | jerlique | iaxtel was hopless for me too. I now use FWD (www.fwdnet.net) |
| 07:54.07 | Alric | jerlique: It'll stop all incoming audio, if I'm remembering correctly. And its not so much VAD, but rather the lack of implementation of VAD (If I understand correctly) |
| 07:54.41 | Alric | I did some testing on this w/ a 7905 a while back. Asterisk didn't like VAD being on. |
| 07:54.41 | Pulu | i tried to sign up for the fwd iax account but the howto on their site talks about a certificate, and I don't know where to get it from |
| 07:55.05 | Alric | Gives some NOTICE about an RFC being not fully implemented, turn off on client side if possible. |
| 07:55.12 | Juggie | yeah |
| 07:55.19 | Juggie | no need for vad on a lan anyways |
| 07:55.24 | Juggie | its a waste of time |
| 07:55.55 | Juggie | i'd prefer use more bandwidth then get the clipping which happens with vad |
| 07:56.07 | jerlique | whats VAD? |
| 07:56.22 | Alric | Silence suppression |
| 07:56.23 | implicit | voice activity detection |
| 07:56.26 | jerlique | ok. |
| 07:56.30 | Alric | Thats a better description. |
| 07:57.15 | Juggie | i created an infinite iax loop this evening ;) |
| 07:57.32 | Juggie | lesson learned, use a dif context for incomming iax calls |
| 08:02.19 | netsurfer | what happens if I call up an IVR type setup on asterisk and transfer it to its own extension ? |
| 08:03.44 | jerlique | GREAT work alric. I should have taken notice of the Transmit Silence"=YES) in my notes! |
| 08:04.15 | jerlique | Ok, so now my call is being dropped consisntely on the second announcement of being caller 1 in the queue.... |
| 08:11.18 | *** join/#asterisk jas_williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 08:11.58 | liquidno2 | mental note... always include a dring for 0,0,0 (normal ring cadance) on your channel that you have enabled usedistinctiveringdetection on |
| 08:12.40 | netsurfer | jerlique - Note that a timeout to fail out of a queue may be passed as part of application call |
| 08:12.53 | netsurfer | check its not set |
| 08:13.06 | netsurfer | Queue(queuename|[options]|[optionalurl]|[announceoverride]|[timeout]) |
| 08:14.27 | fa | hi |
| 08:14.47 | firestrm | is there a way to setup zap channel to only answer during the day? |
| 08:14.58 | implicit | yes |
| 08:15.25 | firestrm | implicit, that was quite an implicit answer ;) |
| 08:15.44 | liquidno2 | yeah, just put the zap channel in its own context |
| 08:15.44 | fa | firestrm use IF for time |
| 08:15.47 | implicit | :) |
| 08:15.55 | liquidno2 | and then set your tmie frames on the answer |
| 08:16.00 | netsurfer | firestrm - include => open|8:00-20:00|mon-fri|*|* where "open" is the context |
| 08:16.13 | jerlique | netsufer;i don't have any time out set exten => ${FWDNUMBER},2,Queue(tech|T) |
| 08:16.17 | firestrm | aahhhh.. i grock now... sure that makes sense.. |
| 08:16.45 | liquidno2 | firestrm: a tip |
| 08:16.45 | fa | netsurfer close|20:00-08:00|mon-fri|*|* will work? |
| 08:16.55 | liquidno2 | firestrm: run ntpdate at least once an hour. |
| 08:17.01 | netsurfer | fa - if u have a context called close yes |
| 08:17.10 | jas_williams | firestrm: or you can use gotoiftime look at http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime |
| 08:17.22 | firestrm | so if i do that zap wont answer the call at all, just let it ring? |
| 08:18.11 | firestrm | because ive noticed that zap picks up regardless is i use Answer() or not |
| 08:18.39 | netsurfer | firestrm - it will follow the context |
| 08:18.40 | jas_williams | firestrm: Rather than just letting it ring why not drop the call straight to voicemail and then offer to call back in hours or play a closed announcement and then hangup |
| 08:18.58 | netsurfer | include => open|8:00-20:00|mon-fri|*|* |
| 08:18.58 | netsurfer | include => closed|20:01-07:59|mon-fri|*|* |
| 08:18.58 | netsurfer | include => closed|20:00-19:59|sat-sun|*|* |
| 08:19.06 | netsurfer | oops that should have been a pastebin url :o\ |
| 08:19.15 | netsurfer | anyway.. u get the idea |
| 08:19.55 | firestrm | jas_williams, i need it to ring a night phone, not through asterisk.. part of the spec.. i know its stupid, i didnt write the spec.. |
| 08:20.12 | *** join/#asterisk coppice (~chatzilla@245.195.17.210.dyn.pacific.net.hk) |
| 08:20.32 | netsurfer | firestrm - surely if u Wait(999) it wont answer |
| 08:20.58 | netsurfer | im guessing, I dont have any zap hardware (yet) |
| 08:21.05 | firestrm | netsurfer, ive found wait is useless. it just jumps right over regardless of the number of sec you specify |
| 08:21.21 | netsurfer | firestrm - maybe something in zapata.conf then ? |
| 08:21.28 | Juggie | fire, wait works |
| 08:21.31 | firestrm | netsurfer, and it answers with Answer() |
| 08:21.33 | jas_williams | firestrm: Connect the night phone to an Sipura or similar but keep controll under asterisk do not use a phone connected through an x100p |
| 08:22.51 | netsurfer | well, 8.30am time for bed |
| 08:22.56 | netsurfer | g'nite |
| 08:23.00 | jerlique | cya |
| 08:23.03 | firestrm | jas_williams, nightphone must be connected un impeded (ie fallthrough) to an analog line.. in the origional system, they even bypassed the KSU unit for this |
| 08:23.29 | firestrm | netsurfer, gnite :) |
| 08:24.56 | firestrm | jas_williams, its for a womens center.. they dont want anything interfering with phones, at a time when they cant have a tech come and immediatly fix it.. |
| 08:25.28 | jas_williams | firestrm: OK I see] |
| 08:26.00 | firestrm | jas_williams, its a little paranoid, but i can see their point. |
| 08:27.32 | firestrm | Juggie, their must be something wrong with my version of asterisk, it just plows right through wait.. ive controlled tested it.. Ive had to resort to playing 5 sec of blank audio |
| 08:28.29 | crazyhick | finally... just got my sql server running again. The wife was not happy that the phone system has bee ndown for 2 days because of a drive failure :) The cool part is that she is going to let me build a ide hardware raid box :) |
| 08:29.26 | firestrm | crazyhick, is that the answer to getting wives to allow us to buy toys.. make the lack of toy an inconveience to her... interesting.. |
| 08:29.59 | crazyhick | i guess |
| 08:30.53 | firestrm | i'll have to try that.. "geez dear.. i cant figure out how to get to (insert important engagement here) i guess if i had that gps system....." |
| 08:31.00 | crazyhick | She has put up with my phone experimentation since linksys put out a voip router to net2phone 4 years ago. |
| 08:31.25 | brc_ | CONFIRMED: APPLE IS MAKING A g5 powerbook tomorrow! |
| 08:31.30 | brc_ | "I've got no doubt that a G5 could be in a PowerBook tomorrow if Apple were willing to trade design for functionality. They could whip up a three-inch thick computer, make it whine like a leaf blower and let run for an hour on a full charge." |
| 08:32.02 | firestrm | brc_,rotfl..that discribes my old ibm |
| 08:32.18 | crazyhick | the trick is to get the wife to need it. I installed a phone at her mothers so they could yack on an hourly basis. |
| 08:32.28 | brc_ | er |
| 08:32.33 | brc_ | wrong quote |
| 08:33.09 | crazyhick | g5 pb would be nice. I am still happy with my ti667. |
| 08:33.14 | brc_ | "I've got no doubt that a G5 could be in a PowerBook tomorrow" said AppleMatters, "[The powerbook g5 will be a] three-inch thick computer, make it whine like a leaf blower and let run for an hour on a full charge." |
| 08:33.19 | brc_ | there we go |
| 08:33.28 | firestrm | brc_ compaq evo rocks :) |
| 08:33.29 | brc_ | proper journalistic quoting |
| 08:33.30 | brc_ | much better |
| 08:33.47 | brc_ | http://lowendmac.com/bookrev/05/0204.html |
| 08:33.58 | brc_ | I can't decide if I should get a g4 pb or not |
| 08:34.58 | firestrm | brc_, i like the idea of an apple system, but every time ive tried one, its left me frustrated, and unable to do what i feel i should be able to accomplish on a computer |
| 08:35.14 | crazyhick | a friend of mine just bit the bullet. think of it this way. If a g5 came out now, the current g4 will still be better. I can remember that a pismo made a tibook look like junk until the 667 came out. |
| 08:35.26 | brc_ | everybody's probably seen this...but: http://channels.lockergnome.com/news/archives/20050125_happy_birthday_macintosh_the_lost_1984_mac_video.phtml |
| 08:35.39 | brc_ | wrong link, http://www.trunkmonkey.com/content/view/52/51/ |
| 08:35.50 | firestrm | brc_, although i havent checked out the *nix ish OS yet.. |
| 08:36.02 | brc_ | firestrm, yeah |
| 08:36.04 | *** join/#asterisk _LM_ (foobar@cleopatra.jogback.se) |
| 08:36.10 | brc_ | asterisk compiles out of the box |
| 08:36.14 | brc_ | (almost) |
| 08:40.27 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 08:40.33 | silik0n | only this you need is yo upgrade bison |
| 08:40.44 | brc_ | the master hath spoken |
| 08:40.59 | brc_ | now get it working with xcode! |
| 08:41.01 | brc_ | :p |
| 08:41.04 | firestrm | i will have to go find a good mac emulator :) |
| 08:41.11 | brc_ | firestrm, pearpc is all there is |
| 08:41.20 | brc_ | very slow |
| 08:41.49 | firestrm | surly a a64 should be able to emulate a g5 realtime;) |
| 08:41.55 | brc_ | no. |
| 08:41.59 | firestrm | lol |
| 08:42.04 | brc_ | x86 teh sux |
| 08:42.18 | brc_ | read >>> http://arstechnica.com/articles/paedia/cpu/amd-hammer-1.ars |
| 08:43.39 | brc_ | very good read |
| 08:43.48 | brc_ | read it! |
| 08:44.56 | firestrm | k |
| 08:48.58 | *** join/#asterisk DaTrueLion (anon@Toronto-HSE-ppp3884464.sympatico.ca) |
| 08:49.32 | DaTrueLion | woo ever stole DalIon from me u made a point yourfuciked for life |
| 08:49.58 | DaTrueLion | now after this xmas liner im onto you |
| 08:50.22 | DaTrueLion | anyone need billling for voip with fraud prroof system ? |
| 08:50.24 | DaTrueLion | btw |
| 08:51.27 | DaTrueLion | anyhow let me know |
| 09:03.40 | *** join/#asterisk bjohnson_ (~bjohnson@ip226-181.tor.istop.com) |
| 09:09.54 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 09:10.58 | DaTrueLion | yo |
| 09:17.36 | silik0n | http://www.suspicious.org/~krice/vase1.jpg |
| 09:17.41 | silik0n | misfire |
| 09:37.41 | silik0n | http://www.wetdesign.com/portfolio.html |
| 09:49.46 | silik0n | http://www.suspicious.org/~krice/NW1-Stamping.mov |
| 09:53.56 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:00.58 | sskyles | How do I indicate a pause in the middle of a number when dialing? |
| 10:07.38 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 10:08.55 | tzafrir | does it work? |
| 10:09.03 | tzafrir | ~xorcom rapid |
| 10:09.04 | jbot | methinks xorcom rapid is at http://www.xorcom.com/rapid.html . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir |
| 10:09.13 | tzafrir | ~xorcom rapid |
| 10:09.14 | jbot | well, xorcom rapid is at http://www.xorcom.com/rapid.html . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir |
| 10:09.20 | tzafrir | ~xorcom rapid |
| 10:09.21 | jbot | well, xorcom rapid is at http://www.xorcom.com/rapid.html . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir |
| 10:09.35 | tzafrir | not the right effect. nm |
| 10:11.56 | djin | tzafir, Rapid is your project, isn't it? |
| 10:12.27 | tzafrir | I'm just an employee. I maintain it |
| 10:12.58 | djin | Oh, ok. It looks pretty neat, but download of iso fails? |
| 10:13.30 | djin | "Sorry can't allow you access today" |
| 10:13.40 | djin | same as yesterday ;) |
| 10:14.04 | SexyKen | Listen guys, I need something that will monitor servers. Nagios aint workin' too well. |
| 10:15.34 | *** join/#asterisk justinnnnnn (~justinnnn@c211-28-200-67.eburwd1.vic.optusnet.com.au) |
| 10:15.35 | djin | http://argus.tcp4me.com/ |
| 10:15.42 | djin | http://www.voip-info.org/tiki-index.php?page=Example%20Argus%20Config |
| 10:15.49 | djin | http://www.voip-info.org/tiki-index.php?page=Asterisk%20monitoring |
| 10:15.54 | justinnnnnn | hehe |
| 10:15.56 | justinnnnnn | we used to use nagios |
| 10:15.58 | justinnnnnn | its got issues |
| 10:16.03 | justinnnnnn | i mean argus.. |
| 10:16.06 | justinnnnnn | now we use nagios. |
| 10:16.13 | justinnnnnn | its also got issues :( |
| 10:16.27 | sskyles | How do I indicate a pause in the middle of a number when dialing? |
| 10:16.29 | *** join/#asterisk tla (~tl@almestien.com) |
| 10:18.28 | djin | I don't use Argus myself. Just bookmarked it. |
| 10:19.10 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 10:37.09 | Poincare | justinnnnnn: what issues you have with nagios? |
| 10:38.08 | *** join/#asterisk DaFuckingLion (anon@Toronto-HSE-ppp3771214.sympatico.ca) |
| 10:41.32 | *** join/#asterisk SeaForth (~SeaForth@c-24-1-126-202.client.comcast.net) |
| 10:41.56 | SeaForth | hi there guys. What would be consider the best price for performance headset under 20 USD that I can use for VoIP? |
| 10:44.02 | DaTrueLion | ich |
| 10:44.04 | DaTrueLion | try |
| 10:50.26 | justinnnnnn | u no... |
| 10:50.31 | justinnnnnn | if u all gave me $1 in paypal |
| 10:50.34 | justinnnnnn | id have $233 |
| 10:50.41 | justinnnnnn | shall we start the transfer ? |
| 10:58.04 | SeaForth | no |
| 10:58.22 | SeaForth | sounds like the beginning of yet another Telephony Ponzi Scheme. |
| 10:58.40 | tzafrir | djin, the ISO is now avilable |
| 10:59.32 | tzafrir | But I think I'll upload a new version soon: I've just noticed that I need to protect the passwordless IAX extensions just as I protext the passwordless SIP extensions |
| 11:00.20 | djin | tzafrir, thanks. I'll give it a try. |
| 11:04.04 | SeaForth | Is there an emerging (or existing) protocol for VoIP... Say one maker's client could chat with another manufactures? Ex: skype with say some client X |
| 11:07.46 | djin | SeaForth, uuuh SIP? |
| 11:16.27 | tzafrir | SeaForth, SIP? |
| 11:16.33 | *** join/#asterisk NoRemorse (~me@202.161.68.2) |
| 11:16.37 | NoRemorse | hello all |
| 11:16.40 | tzafrir | not exactly emerging, though |
| 11:26.09 | Yoda-BZH | Le bonjour je vous souhaite / Hi ppl |
| 11:40.30 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 11:40.54 | markit | hi :) anyone using the native attended transfer? (cvs head) |
| 11:45.07 | SeaForth | thanks guys. |
| 11:45.46 | SeaForth | so if I use the OSX Skype client, it will allow me to connect and work with another providers VoIP client if both clients use SIP? |
| 11:46.06 | tzafrir | SeaForth, skype doesn't use SIP |
| 11:46.07 | JerJer | um no |
| 11:46.11 | JerJer | skype is not sip |
| 11:46.19 | tzafrir | skype is not a standard client |
| 11:47.05 | djin | skype=closed protocol (and, for now, network) |
| 11:47.37 | SeaForth | ok |
| 11:47.42 | SeaForth | that is clear to me now. |
| 11:47.44 | SeaForth | thanks. |
| 11:47.47 | SeaForth | skype = blows. |
| 11:49.40 | djin | technology is quite nice. |
| 11:56.01 | *** join/#asterisk florz (nobody@odnb-d9baa5b5.pool.mediaWays.net) |
| 11:59.44 | jerlique | why is it that the AgentCallbackLogin allows anyone to login regardless of the password? |
| 12:01.51 | jerlique | Actually to be more precise its not actually loggin the person in, it just reports success. |
| 12:03.55 | *** join/#asterisk wasim (~wasim@203.81.200.8) |
| 12:04.15 | markit | justinnnnnn: ... |
| 12:09.42 | *** join/#asterisk ctooley_mobile (~chris@65.166.25.111) |
| 12:10.09 | ctooley_mobile | good morning folks |
| 12:10.27 | djin | morning. |
| 12:10.43 | wasim | 'eve |
| 12:13.16 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 12:15.01 | *** join/#asterisk eivindtr (~Eivind@062016241059.customer.alfanett.no) |
| 12:16.40 | ctooley_mobile | How long does a sip registration last in Asterisk? |
| 12:16.55 | justinnnnnn | 3.2 seconds |
| 12:17.23 | djin | that depends on your settings in Asterisk. |
| 12:17.42 | wasim | justinnnnnn: no, no, not an erection, a registration! |
| 12:18.13 | ctooley_mobile | so I can change the defaultexpirey setting or the default is 3200ms |
| 12:18.14 | ctooley_mobile | ? |
| 12:19.48 | djin | ;maxexpirey=3600 ; Max length of incoming registration we allow |
| 12:19.48 | djin | ;defaultexpirey=120 ; Default length of incoming/outoing registration |
| 12:20.06 | djin | check /etc/asterisk/sip.conf |
| 12:20.12 | ctooley_mobile | djin: I don't need to worry about inbound registrations, just outbound |
| 12:20.29 | ctooley_mobile | djin: so I'm guessing that means that I need to only change my defaultexpirey |
| 12:20.51 | justinnnnnn | can u do 3way calling in asterisk yet ? |
| 12:20.56 | ctooley_mobile | djin: The next questions is, how large can that number be? 3600000? |
| 12:21.03 | justinnnnnn | were u can join 2 calls 2gether type thing ? |
| 12:21.31 | djin | ctooley_mobile, not sure what maximums are. |
| 12:21.31 | wasim | justinnnnnn: threewaycalling=yes |
| 12:22.05 | justinnnnnn | wasim.. how do uactualy do it tho ? |
| 12:22.30 | wasim | justinnnnnn: flash hook |
| 12:22.39 | justinnnnnn | hook ? |
| 12:24.54 | shadebob | Hi, How can I plug many regulars phones (more than 20) on an asterisk server? |
| 12:25.11 | djin | threewaycalling: If enabled, you can place a call on hold by pressing a hook flash, whereupon you get a dialrecall tone and can make another call. Default: no. |
| 12:25.45 | justinnnnnn | and then how do u link the 2 calls ? |
| 12:26.08 | djin | shadebob, ATA-devices or FXS channels |
| 12:26.51 | djin | justinnnnnn, would the new call be included in the current call? |
| 12:27.23 | djin | transfer: This option has effect only when threewaycalling=yes. If threewaycalling=yes and transfer=yes, then once you've placed a call on hold with a hook flash, you can transfer that call to another extension by dialing the extension and hanging up. Default: no. |
| 12:27.28 | djin | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20zapata.conf |
| 12:27.46 | shadebob | djin : How can I install many Fxs port in a single asterisk server? A channel bank can do this job? |
| 12:28.22 | ctooley_mobile | shadebob, depends on how mobile the phones are. ATA's are going to be in the 60-80 dollar range and you have to have on per phone. If you get a T1 card and a channel bank everything is handled in the telephony room and provide service to existing telephone cabling. |
| 12:29.54 | djin | shadebob, I agree with ctooley_mobile. ATA was suggested, as I don't know the physical location of the phones. The might be spread throughout the country ;) |
| 12:30.15 | shadebob | ctooley_mobile : exactly ctooley. I want to use existing cabling and phones. I can do this <t1>------<Asterisk>----Channel Bank (Adit 600)-----<regular phones>? |
| 12:30.40 | ctooley_mobile | shadebob, you're going to need a T1 card between Asterisk and your channelbank. |
| 12:30.54 | shadebob | Asterisk servers will be used to link differents sites, but existing phones system will be the same as now |
| 12:31.33 | shadebob | ok ctooley_mobile |
| 12:32.20 | ctooley_mobile | shadebob, so it's more like: <T1>-----<T100P--Asterisk--T100P>----<Adit 600>----<regular phones> with the T100P cards in the Asterisk server |
| 12:33.10 | djin | or use a TE400P with two extra port for future expansion. |
| 12:33.13 | shadebob | ctooley_mobile : ok :) it will be so easy ;) |
| 12:33.18 | ctooley_mobile | You might find it more beneficial to buy a 400 (not the TDM400) as you're already making a large enough commitment to justify just getting the expansion room. |
| 12:33.27 | ctooley_mobile | djin: I agree with djin. |
| 12:34.13 | shadebob | thanks djin and ctooley_mobile |
| 12:34.15 | ctooley_mobile | shadebob: Easy is a relative term. It's easier for me to design and complete a skyscraper than it is to make SIP work. |
| 12:36.15 | shadebob | ctooley : just with SIP and E110P I haven't any problem.... But maybe with an Adit 600 or anything else it's an another question... |
| 12:36.40 | shadebob | I will see in a near future ;à |
| 12:41.09 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 12:45.47 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-6-3.d4.club-internet.fr) |
| 12:48.33 | *** join/#asterisk kaitseb (~sadie@jutrzenka.firma.o2.pl) |
| 12:51.09 | ctooley_mobile | Ok, so is there someone here that answer a basic |
| 12:51.19 | ctooley_mobile | "How SIP Works" questino |
| 12:53.22 | ctooley_mobile | If I register with a provider to be able to recieve calls via SIP at my proxy load balancer and then try to make calls with my Asterisk box without spoofing my address shouldn't I be allowed to use that SIP information as long as I have the correct username and secret? |
| 13:02.11 | ariel_ | Morning all |
| 13:02.39 | djin | morning, ariel. |
| 13:04.54 | *** join/#asterisk SirPrize (~blah@83.146.62.73) |
| 13:06.07 | ariel_ | ctooley_mobile, your proxy load balancer is natting your asterisk box is it not? |
| 13:06.08 | SirPrize | What command could I use to be able to allow the user to enter for example a duration specifier via the keypad, which I could then use to dynamically set a timeout for their account? |
| 13:06.22 | SeaForth | out of bizarre curiosity, when was the area code introduced? |
| 13:06.57 | Chuji | SirPrize : what kind of timeout? |
| 13:07.18 | SirPrize | Chuji: for example, the amount of time asterisk would ring an extension, before going to voicemail |
| 13:07.19 | ctooley_mobile | ariel_: no |
| 13:07.30 | markit | hi :) anyone using the native attended transfer? (cvs head) |
| 13:07.37 | Chuji | SirPrize : Permanently? or per call? |
| 13:08.10 | ctooley_mobile | ariel_: That would mean that all fo the traffic would come from one box. We're going to have more calls than the network connection can handle. |
| 13:08.11 | ariel_ | ctooley_mobile, you should be able to use the sip info. But use sip debug to see what you actually get. |
| 13:08.14 | Chuji | SirPrize : So it's like a call forward no answer time on a traditional pbx? |
| 13:08.28 | SirPrize | Chuji: either way. The main point of this exercise for me is to understand how to accessa a number that a user enters in via the keypad, as a variable from within Asterisk |
| 13:08.43 | Chuji | SirPrize : Look at dbput and dbget |
| 13:08.44 | tzafrir | bah, why doesn't iax.conf have a context option in the general section? :-( |
| 13:08.46 | ctooley_mobile | ariel_: Well with Voicepulse I get "503 Service Unavailable", but I'm not sure if htat's Voicepulse or SIP. |
| 13:08.49 | ariel_ | ctooley_mobile, you might need to setup a stun server for the different ip address. |
| 13:08.52 | Chuji | SirPrize : You can store whatever you like in there. |
| 13:09.06 | Chuji | SirPrize : So you create a macro that uses a Read() command |
| 13:09.19 | Chuji | The Read() sets a variable and you dbput it |
| 13:09.23 | ariel_ | ctooley_mobile, sip debug will tell allot of information about your call. |
| 13:09.32 | Chuji | SirPrize : Then, you can retrieve that any time |
| 13:09.34 | SirPrize | Chuji: Brilliant - that's what I was looking for. Thanks! |
| 13:09.38 | ctooley_mobile | ariel_: So what would be the best way to set up 10 Asterisk servers to handle the load when there's only one Termination and one Origination account from the provider? |
| 13:10.25 | ctooley_mobile | ariel_: We'll be getting 3-5000 concurrent calls on that channel in the beginning moving to 50-100,000 within 3 months. |
| 13:12.26 | ariel_ | ctooley_mobile, I have not done a system that large. But I can tell you that I have been able to get 2 main servers to load balance for inbound PSTN lines and outbound but we had to use Lucent TNT servers to supply the PRI's to us. |
| 13:12.36 | tzafrir | I had such a lovely workaround on sip, and it doesn't work on iax. I want to prevent password-less IAX extensions in the default configuration from being abused |
| 13:12.50 | justinnnnnn | hey ppls |
| 13:12.53 | justinnnnnn | if u do call parking.. etc.. |
| 13:12.58 | justinnnnnn | how do u cdr the call :) ? |
| 13:13.14 | ariel_ | ctooley_mobile, asterisk is not really made for that many sip clients. |
| 13:13.16 | wasim | justinnnnnn: ResetCDR |
| 13:13.35 | tzafrir | The workaround with sip was to give them all a bogus context , in the general section. And then I only need to change one place when the user feels it is OK |
| 13:13.38 | markit | damn, native assisted transfer support was one of the most requested features... now seems that no one is using it :( |
| 13:13.48 | *** part/#asterisk SirPrize (~blah@83.146.62.73) |
| 13:14.08 | ctooley_mobile | ariel_: It's only one SIP client, that's the only real problem |
| 13:14.10 | ariel_ | ctooley_mobile, but I feel that maybe the people at fwd could help in getting you the redundent load balance system. |
| 13:14.29 | justinnnnnn | wasim, but when u reconnect hows it going to no to add on the previous |
| 13:14.46 | justinnnnnn | and when u reconnect hows it going to no '700' or woteva.. is whoever u dialed internationaly.. |
| 13:15.51 | wasim | justinnnnnn: set account code and follow it from there, i have done parking with CDR, so i might be barking up the wrong tree |
| 13:16.00 | wasim | justinnnnnn: s/have/haven't |
| 13:20.14 | tzafrir | Is there any other simple way of keeing the extensions there but "disabling" them in a "user-visible" way? |
| 13:20.35 | tzafrir | IAX extensions, I mean |
| 13:20.39 | djin | what is visible to the user? |
| 13:21.33 | tzafrir | making calls is not possible. The name of the context is disabled-sip-insecure-read-getting-started |
| 13:22.13 | djin | I guess comments is not what you're lookign for? |
| 13:22.30 | djin | lol @ wasim. |
| 13:22.40 | tzafrir | And they have a menu option that changes the context back to "default" |
| 13:22.55 | tzafrir | which is a minimal change: 1 file, 1 line |
| 13:23.21 | *** join/#asterisk Jedirl (irm22@154.Red-217-127-168.pooles.rima-tde.net) |
| 13:23.25 | Jedirl | Hello |
| 13:24.49 | *** join/#asterisk sd-tux (~vnvbnvbn@athalle2.informatik.tu-muenchen.de) |
| 13:25.04 | tzafrir | I can remove all the IAX extensions (unload iax module or remove an include line) but then the users won't know that they need to do something to enable iax extensions |
| 13:25.54 | tzafrir | djin, I'm alway looking for comments |
| 13:26.05 | djin | you can comment them out? |
| 13:26.42 | djin | ; #include <customer/iax.conf> |
| 13:27.26 | tzafrir | I can rem-out the line that includes all of the extensions. |
| 13:27.46 | tzafrir | But will the users know that the system should have IAX users/extensions? |
| 13:29.18 | tzafrir | well, seems like no better alternative |
| 13:30.07 | *** join/#asterisk nirs (~nirs@62.90.49.115) |
| 13:30.17 | nirs | good morning everybody |
| 13:30.21 | nirs | how are we today ? |
| 13:30.33 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 13:30.40 | djin | Hi Zeeek. |
| 13:30.47 | Zeeek | djin hullo |
| 13:31.36 | Zeeek | Holy convention, Batman... VON 2005 discount price, $2495 |
| 13:31.58 | Zeeek | no wonder all those guys are speakers! They won't have to pay :) |
| 13:32.19 | djin | For what, 100x100 stand? |
| 13:32.26 | Zeeek | to attend! |
| 13:32.51 | djin | wow. |
| 13:33.18 | Zeeek | one day reg = $100 |
| 13:33.22 | Zeeek | $1100 |
| 13:33.41 | djin | why that much? Does it include a female escort? |
| 13:33.45 | Zeeek | exhibits only $200 |
| 13:33.53 | Zeeek | no joke - FOR LIFE ! |
| 13:34.19 | Zeeek | for $200 that must include something better than the "snack" mentioned |
| 13:34.33 | Zeeek | http://pulver.com/von/register.html |
| 13:34.35 | *** join/#asterisk zbysio (~zbysio@chello084010031149.chello.pl) |
| 13:34.38 | djin | define snack. |
| 13:35.13 | djin | well, it includes all meals. |
| 13:35.15 | Zeeek | Charley Theron? |
| 13:35.27 | Zeeek | yeah $2200 is some great food! |
| 13:35.43 | Zeeek | I guess it's a limited audience... |
| 13:35.46 | Zeeek | so high price |
| 13:36.22 | Zeeek | must be only reserved for the brightest lights on voIP |
| 13:36.25 | djin | it also defines the type of visitors. |
| 13:36.27 | Zeeek | because |
| 13:36.29 | Zeeek | " Our secure serveruses an advanced encryption technology called SSL" |
| 13:36.30 | wasim | Zeeek: thats just a little over 2 farfon beta evals |
| 13:36.43 | Zeeek | heh two? more like one! |
| 13:37.16 | Zeeek | an advanced encryption technology called SSL... Hmmmm wow, that's a new one on me! |
| 13:37.26 | tzafrir | hi nirs |
| 13:38.12 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 13:38.14 | Zeeek | hi |
| 13:40.26 | djin | As you're mentioning Farfon. Did they release their IAX-phone by now? |
| 13:41.41 | Zeeek | any second! |
| 13:42.09 | jerlique | what would this refer to: |
| 13:42.12 | jerlique | app_queue.c:374 changethread: Can't change device '**Unknown**' with no technology! |
| 13:42.21 | djin | ?? |
| 13:44.19 | djin | when does this occur? |
| 13:44.45 | Zeeek | I'm being dragged outside against my will.... help |
| 13:44.57 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 13:50.31 | jerlique | when I've just finished leaving a message |
| 13:51.19 | wasim | who, what ... farfon! |
| 13:51.36 | wasim | djin: yeah, they did, late as ever |
| 13:52.38 | wasim | djin: 60% of the phones are now in .EU, some should make their way to US |
| 13:52.45 | djin | does it work well? I personally thinks it looks kinda cheap. |
| 13:53.01 | djin | well, I'm in Europe. |
| 13:53.23 | wasim | djin: its a donor casing, these are beta eval units to test the core OS and IAX stack and other niche stuff like encryption etc |
| 13:53.45 | djin | ah, ok. |
| 13:54.35 | Mw3 | hm, what ? iax phone ? where ? |
| 13:54.54 | djin | http://www.farfon.com |
| 13:55.07 | Mw3 | thanks |
| 13:55.41 | Mw3 | i'd need a phone which can do sip and iax2 :) |
| 13:55.59 | djin | why is that? |
| 13:58.00 | Mw3 | because i'd like to test sip and iax2 with the same device :) |
| 13:59.19 | wasim | no SIP, no H.323, no MGCP, no CRAP |
| 13:59.50 | wasim | hail IAX2! |
| 14:00.08 | Mw3 | i'm working for an isp and we're planning to give voip to our customers, but we haven't decided what protocol/device to use |
| 14:00.30 | djin | Mw3, there isn't much to decide. |
| 14:00.56 | Mw3 | others using sip |
| 14:01.00 | djin | Sip is more widely supported by both hard- and software. |
| 14:01.19 | djin | Iax is a great protocol, but it limits you to Asterisk. |
| 14:01.32 | wasim | yate does IAX too |
| 14:01.52 | djin | ok, * and Yate ;) |
| 14:03.39 | lohelle | A couple of useful (I think) scripts at www.tech-support.no/asterisk please comment! |
| 14:05.30 | mbranca_home | Mw3: use sip AND iax for customers, and iax for server2server backbones |
| 14:05.44 | netsurfer | lohelle - hmm nice, generatecall looks useful :) |
| 14:05.53 | Mw3 | djin: we are planning to make a hardware for us, which support 2 telephone lines with iax or sip and a s-video out for mpeg2 |
| 14:05.55 | mbranca_home | if you want only 1 proto for the customers, go with sip |
| 14:06.05 | Mw3 | djin: set-top-box device or what :) |
| 14:06.10 | mbranca_home | iax, so :) |
| 14:06.29 | mikegrb | lohelle: you forgot to copy the .call file in to the outgoing spool in your script |
| 14:06.37 | lohelle | changed now! :) |
| 14:06.39 | lohelle | tnx |
| 14:06.45 | mikegrb | lohelle: also, /tmp would be a much better place to build it then /root |
| 14:07.18 | lohelle | mikegrb: I know.. just an example |
| 14:07.36 | djin | Mw3, sound very interesting. |
| 14:07.56 | lohelle | but it actually works! I can call in and join whoever I want to a conference etc. |
| 14:08.07 | djin | Mw3, is there some site where I can check updates on this proejct? |
| 14:09.38 | mikegrb | lohelle: for your ping script, you might look at asterisk-sounds in cvs, there are some recordings you may be interested in, such as "ping" |
| 14:09.40 | Mw3 | djin: no, we just contacted a hardware making company in taiwan and sent the specifications |
| 14:09.59 | Mw3 | djin: we're waiting for their answer |
| 14:10.51 | djin | Mw3, cool. Looking forward to progress. |
| 14:10.55 | lohelle | mikegrb: ok. tnx. I was thinking about creating more sounds for this one.. "please input ip address to ping" etc. |
| 14:12.52 | lohelle | ne1 know of a place to download scripts like this? |
| 14:12.58 | lohelle | useful.. |
| 14:18.13 | *** join/#asterisk dasv (~chatzilla@ua-83-227-224-57.cust.bredbandsbolaget.se) |
| 14:20.17 | markit | what does "disconnect" does in features.conf? it does not hung up :( |
| 14:20.44 | dasv | Looking for IAXy in Europe... Anyone know where to get one? |
| 14:21.01 | wasim | dasv: a digium distributor or reseller |
| 14:21.39 | Mw3 | which is the best isdn bri card for asterisk by the way ? |
| 14:21.42 | dasv | wasim: been searching, but haven't found it in Europe yet... |
| 14:21.45 | Mw3 | dasv: www.beronet.com |
| 14:21.48 | ariel_ | markit, I did not know there was a disconnect in features.conf |
| 14:22.01 | wasim | Mw3: octobri from junghanns.net |
| 14:22.02 | lohelle | has anyone integrated bristuff into latest cvs.. There are features I "need" in both! :) |
| 14:22.15 | *** join/#asterisk adnans (~adnans@linuxgoeroe.demon.nl) |
| 14:22.32 | markit | ariel_: cvs head |
| 14:22.38 | dasv | Mw3: Thanks. I'll check that out. |
| 14:22.45 | markit | ariel_: under [featuremap] |
| 14:22.59 | Mw3 | wasim: ah, i need only 1 or 2 ports |
| 14:23.15 | wasim | Mw3: thats ok, they are the best |
| 14:23.24 | djin | Mw3, I like the EIcon's, but they're expensive (I have them from Hylafax projects), but you can check the quadbri from junghanns |
| 14:23.25 | ariel_ | markit, thanks I I don't use head. (I stay with stable it's safer). |
| 14:24.06 | Mw3 | wasim: and where can i buy it ? |
| 14:24.06 | ariel_ | I wish we had more access to isdn lines here in the US. |
| 14:24.19 | wasim | Mw3: junghanns.net |
| 14:24.21 | *** join/#asterisk sudhir492 (~sudhir@4.7.59.167) |
| 14:24.24 | sudhir492 | hi all |
| 14:24.40 | Mw3 | wasim: hm, i cant find it there |
| 14:24.53 | Mw3 | djin: i've heard that eicon diva is not well supported |
| 14:25.05 | sudhir492 | anyone doing IVR with asterisk here? |
| 14:25.07 | markit | Mw3: try http://www.beronet.com/?PageID=3017 also |
| 14:25.08 | djin | multiBRI (zaptel) |
| 14:25.15 | netsurfer | sudhir492 yes |
| 14:25.21 | sudhir492 | I need some tips for doing a litte complicated IVR |
| 14:25.33 | netsurfer | sudhir492 shoot |
| 14:25.43 | djin | Mw3, never had problems with Eicon (server) cards. |
| 14:25.47 | lohelle | I use el cheapo asuscom cards.. (only single..). Is these quadbri cards stable? |
| 14:27.01 | *** join/#asterisk BBRodriguez_ (~BBRodrigu@pD956386B.dip.t-dialin.net) |
| 14:27.03 | *** join/#asterisk ToyMan (~konversat@204.8.82.238) |
| 14:27.14 | sudhir492 | In the process of IVR, when we collect responses, how to branch on various responses, and how to collect save all collected responses |
| 14:27.39 | netsurfer | sudhir492 - go to wiki and search on asterisk read cmd |
| 14:27.54 | sudhir492 | which one? |
| 14:27.59 | netsurfer | 'read' |
| 14:28.15 | sudhir492 | thanks. |
| 14:28.17 | netsurfer | yw |
| 14:28.28 | Mw3 | djin: is eicon works with capi driver ? |
| 14:28.49 | sudhir492 | I have been doing simpler ones with Play and Background |
| 14:28.49 | netsurfer | heh dont u just hate it when a kernel install goes wrong on a remote box |
| 14:29.02 | djin | Mw3, I use chan_capi for them. |
| 14:29.24 | sudhir492 | netsurfer: Oh boy ! I hope the box is not more than 3000 miles :-) |
| 14:29.27 | *** part/#asterisk justinnnnnn (~justinnnn@c211-28-200-67.eburwd1.vic.optusnet.com.au) |
| 14:29.30 | ariel_ | sudhir492, someone here lohelle has done something might help you with what your doing www.tech-support.no/asterisk |
| 14:29.54 | Mw3 | djin: ah, nice. someone told me eicon is sux it works with chan_modem which is not the best |
| 14:30.05 | sudhir492 | ariel_: thanks. I will look into that too |
| 14:30.06 | Mw3 | djin: but he wasn't right then ... |
| 14:30.16 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 14:31.15 | netsurfer | sudhir492 na.. not even 300 steps |
| 14:31.51 | djin | Mw3, everyone has different experience. I use Eicon for years for Hylafax configurations (PRI-cards). Chan_capi works very well for me. |
| 14:32.18 | Mw3 | ah, you use PRI cards ... i need bri cards :) |
| 14:32.57 | djin | Mw3, I use 2BRI and 8Bri cards as well, same CAPI driver ;) |
| 14:36.32 | Mw3 | nice |
| 14:39.46 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 14:46.22 | Jedirl | anyone writes AGI scripts ? |
| 14:47.12 | wasim | Jedirl: anybody who uses AGI, does |
| 14:47.32 | Jedirl | i'd like to know if there's any way to execute the AGI scripts in a different machine from the one which runs asterisk |
| 14:47.45 | Jedirl | I mean, having a dedicated "agi" server for this purpose |
| 14:47.49 | wasim | Jedirl: why not? just ssh -x blah |
| 14:48.00 | Jedirl | ssh -x? |
| 14:48.00 | liquidno2 | ssh is the way to go |
| 14:48.12 | liquidno2 | man ssh |
| 14:48.14 | wasim | err ... no -x ssh host command |
| 14:48.15 | Jedirl | I know ssh |
| 14:48.24 | liquidno2 | apparently not |
| 14:48.44 | Jedirl | your perceptions are not correct, then |
| 14:48.44 | liquidno2 | because you would know how to remotely execute commands with ssh then |
| 14:50.45 | djin | Jedirl, did you check fastAGI? |
| 14:50.55 | Jedirl | no |
| 14:50.57 | Jedirl | what's it |
| 14:51.01 | djin | http://www.sineapps.com/news.php?rssid=142 |
| 14:51.08 | Jedirl | thankx |
| 14:51.10 | djin | It's what you're looking for. |
| 14:51.29 | Jedirl | thankyou :) |
| 14:57.28 | *** join/#asterisk Dibblah (~Dibblah@82-41-243-74.cable.ubr02.dund.blueyonder.co.uk) |
| 14:58.11 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com) |
| 15:01.35 | *** join/#asterisk hundra (hundra@xtc.df.lth.se) |
| 15:01.50 | hundra | hello! |
| 15:02.11 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:03.10 | djin | welcom back Zeeek. |
| 15:03.15 | Zeeek | heh |
| 15:03.15 | djin | hundra, hi. |
| 15:03.22 | Zeeek | it's an addiction |
| 15:03.43 | djin | :> |
| 15:04.21 | hundra | using sql to read extensions and voicemail information.. i noticed that there are a couple of different solutions, is there a safe bet which of the solutions to use? (i'm using 1.0.5 atm) |
| 15:04.33 | zyke | is there anyway u can convert your exsting agi scripts into FastAgi scripts ? |
| 15:06.58 | *** join/#asterisk dca (~teliax@c-67-166-37-218.client.comcast.net) |
| 15:12.05 | *** join/#asterisk zoa (~zoa@ip-212-239-162-34.dsl.scarlet.be) |
| 15:12.09 | zoa | yo |
| 15:13.10 | djin | whazzup? |
| 15:15.11 | Zeeek | quiet sunday |
| 15:16.20 | djin | Zeek, wouldn't want it any other way ;) |
| 15:16.54 | Zeeek | I dunno... could be a distor war or two |
| 15:17.15 | Zeeek | It's been at least a few hours since the s word has appeared |
| 15:17.34 | djin | erhm, 's'-word? |
| 15:18.01 | Zeeek | speaking of which... I tried yet another new firmware, 5.22 for BT100 series and still doesn't work |
| 15:18.04 | *** join/#asterisk christo (~chris@212.18.226.160) |
| 15:18.09 | Zeeek | the s word is sucks |
| 15:18.12 | christo | afteroon |
| 15:18.21 | Zeeek | so to reiterate.... |
| 15:18.35 | Zeeek | asterisk on a dynamic ip, BT102 on a fixed one. |
| 15:18.46 | Zeeek | the BT102 need not register since it's fixed ip |
| 15:19.00 | Zeeek | but it has to be able to do DNS to find the asrerisk (and time) server |
| 15:19.13 | Zeeek | all firmware after 5.11 does not do DNS |
| 15:19.15 | djin | ah, well I don't read those s-remarks. |
| 15:19.31 | Zeeek | it won't find the time server (so eliminates my asterisk install as a cause) |
| 15:19.51 | markit | anyone using "automon" feature? |
| 15:19.54 | Zeeek | it just plain does NOT do any DNS lookups of anything with firmware after .11 |
| 15:20.12 | Zeeek | GS says "no one else has this problem!" which I'm sure is not a lie |
| 15:20.24 | Zeeek | yet, I see no explanation |
| 15:20.42 | Zeeek | if I use ip addresses everywhere, the phone works normally in a ll versions |
| 15:20.57 | Zeeek | if I want DNS, only <= 5.11 works |
| 15:20.58 | netsurfer | why did they remove it ? |
| 15:21.03 | christo | I'm trying to get an IVR system to query a database for a code provided by the user.. are there any good examples of this in the voip-info wiki, or someplace else? |
| 15:21.21 | Zeeek | what database christo? Where? |
| 15:21.26 | netsurfer | christo - I did this last week, hang on i'll get u the code |
| 15:21.40 | christo | mysql |
| 15:21.47 | Zeeek | on same asterisk box? |
| 15:21.51 | christo | the database is on the same box yeah |
| 15:22.03 | Zeeek | should be easy |
| 15:22.05 | christo | I have the odbc stuff installed and the odbcinst.ini setup for mysql |
| 15:22.09 | christo | hopefully :) |
| 15:22.24 | Zeeek | does asterisk have an Mysql() app loaded ? |
| 15:22.25 | djin | odbc for mysql? |
| 15:22.40 | christo | Zeeek - not sure |
| 15:22.45 | christo | djin yup |
| 15:22.56 | Zeeek | well let's see what netsurfer has to say |
| 15:23.00 | wasim | don't do odbc for mysql, do native, i think it'll be faster |
| 15:23.25 | wasim | use citats perl AGI handlers, and mysql perl etc |
| 15:23.30 | christo | wasim - can I do that in my dialplan? I figured I'd have to use ODBCget type commands |
| 15:23.35 | Zeeek | go native, wasim! I can see you in a grass skirt already! |
| 15:23.43 | christo | lol |
| 15:24.40 | Zeeek | nice site design: http://iwxchange.com/ |
| 15:24.56 | netsurfer | damn this celeron 700 sux |
| 15:25.07 | Zeeek | I told you it would be long, djin! |
| 15:25.31 | netsurfer | ;) |
| 15:25.31 | christo | :) |
| 15:25.32 | netsurfer | christo - any questions in particular? I cant get to pastebin |
| 15:26.29 | christo | well - It doesn't really matter how I do it.. I mean if I could just call a little perl script or something, that'd be just as good.. hmm perhaps I should just do that instead |
| 15:26.35 | djin | Well Zeeek, I'm not picking up anything from the floor around you! |
| 15:26.52 | netsurfer | christo - use the MYSQL cmd |
| 15:27.02 | christo | netsurfer - didn't know there was one :) |
| 15:27.08 | netsurfer | christo - now u do ;) |
| 15:27.13 | christo | can I put that straight into my dialplan? |
| 15:27.16 | netsurfer | yes |
| 15:27.26 | netsurfer | check it out on the wiki |
| 15:27.38 | christo | got it.. |
| 15:27.39 | christo | ta |
| 15:27.40 | Zeeek | where is this mysql cmd? In Head or addon? |
| 15:27.50 | christo | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MYSQL |
| 15:28.08 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 15:28.12 | netsurfer | BUT note that I had problems getting it to work with the hostname, use 127.0.0.1 instead of localhost |
| 15:28.15 | Zeeek | addon - ok - I'd seen it but never looked where |
| 15:28.26 | markit | how to activate automon besides features.conf? |
| 15:28.51 | netsurfer | Zeeek - its a great feature :) |
| 15:29.12 | Zeeek | If you need db lookup sounds perfect |
| 15:29.52 | Zeeek | for yopu FireFox fans: http://www.xitimonitor.com/etudes/equipement3.asp |
| 15:30.34 | christo | are comments expressed with ';' in * ? |
| 15:30.34 | Zeeek | in .conf yes except for zaptel |
| 15:31.29 | christo | cool |
| 15:32.58 | *** join/#asterisk danfrey (user@24.229.228.66) |
| 15:36.32 | Zeeek | did I remember correctly that the X100P FXO usues only two wires? |
| 15:37.04 | wasim | Zeeek: yep, pins 4/5 |
| 15:37.14 | danfrey | Hello all, I am install asterisk as we speak. 2 machines, 1st machine has 2 x100p, second machine has 2 quicknet phonejacks. I plan to do TDMoE between them to send 2 pstn lines about 1/2 mile over multimode fiber. I would like to disable everything that I don't need. (sip, voicemail, codecs, etc.) Could somebody point me to a good resource on how to disable them? |
| 15:37.32 | wasim | danfrey: modules.conf |
| 15:37.47 | Zeeek | wasim and the same would be the case for the FXS from TDM400P ? and |
| 15:37.53 | danfrey | thank you |
| 15:38.06 | wasim | Zeeek: affirmative |
| 15:38.18 | Zeeek | so that can't be why callerid is not working |
| 15:38.22 | Zeeek | on one phone |
| 15:38.24 | djin | danfrey, http://www.voip-info.org/tiki-index.php?page=Asterisk%20Slimming |
| 15:38.57 | danfrey | I thought I had seen that somewhere. Thanks |
| 15:39.10 | Zeeek | anyone know anything at all about the ARM codec? |
| 15:39.15 | sudhir492 | exit |
| 15:39.17 | sudhir492 | exit |
| 15:39.18 | wasim | AMR ? |
| 15:39.25 | Zeeek | errr, yes |
| 15:39.29 | bjohnson_ | anyone had this win98 problem before (troubleshooting wife's laptop) .. boots into desktop, flashes panel, then nothing (no desktop icons or panel .. just desktop colour and working mouse cursor). Smells like a corrupted system file but not sure how to correct. |
| 15:39.31 | sudhir492 | oops, sorry, wrong window |
| 15:40.07 | Zeeek | bjohnson format c: |
| 15:40.10 | Jedirl | xD |
| 15:40.18 | wasim | bjohnson_: gentoo.org |
| 15:40.20 | Zeeek | but maybe boot in safe mode first |
| 15:40.29 | bjohnson_ | same thing in safe mode |
| 15:40.52 | *** join/#asterisk nassy (~mark@24-193-228-118.nyc.rr.com) |
| 15:41.02 | bjohnson_ | wasim: thanks for the tip .. I've run linux for a number of years but gentoo on her laptop = no nookie |
| 15:41.19 | tzafrir | any way to cheage the shell on that win98? |
| 15:41.40 | wasim | bjohnson_: my wife has totally converted to gentoo, she wouldn't at first, but now she keeps telling all her coffee friends about firefox and how to use multiple desktops and tabs etc |
| 15:41.41 | djin | yes, DOS. |
| 15:41.42 | bjohnson_ | change the shell? |
| 15:42.06 | tzafrir | IIRC you can change the shell to something other than explorer. It seems some parts of explorer fail to load |
| 15:42.16 | danfrey | isn't the shell defined in system.ini |
| 15:42.28 | bjohnson_ | I can boot into DOS .. I've also looked at bootlog.txt and browesed around using knoppix but don't see a specific problem |
| 15:46.31 | Zeeek | can you re-install over the Win98? |
| 15:46.48 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfnid.dialup.mindspring.com) |
| 15:46.54 | Zeeek | as a last resort sometimes all the apps and everything stay and the os is fixed |
| 15:46.56 | bjohnson_ | found this http://www.duxcw.com/dcforum/DCForumID7/307.html |
| 15:46.59 | bjohnson_ | checking it now |
| 15:47.05 | sd-tux | hi, I have a question: I not a PBX expert but i know that some PBXes like siemens HiPath can show the name of called person to the caller. for example i'm calling number: 101 and i see the name of person i have called. is it possible to have this function with asterisk and SIP ? does anybody know if there is support for this in SIP RFC... ? |
| 15:47.47 | Zeeek | sd-tux if your phone supports it, yes |
| 15:48.05 | bjohnson_ | sd-tux: I think so but you'd have to read the data from your own db (and your phone would have to support being backfed the callerid) |
| 15:49.02 | sd-tux | bjohnson_: i'm interested in SIP only solution ... no ldap directorys etc... |
| 15:50.04 | fa | bo |
| 15:50.07 | sd-tux | Zeeek: what sip header have i to use? |
| 15:50.27 | Zeeek | if your talking about a SIP phone connected to asterisk, it just hgappens |
| 15:50.59 | Zeeek | or happens even |
| 15:53.40 | sd-tux | Zeeek: hmm i see that my phone sends my name in "contact:" header... |
| 15:53.54 | sd-tux | Zeeek: thank you :) |
| 15:54.15 | Zeeek | not me, Mark Spencer! |
| 15:54.26 | Zeeek | (or maybe the SIP people?) |
| 15:55.15 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 15:57.52 | *** join/#asterisk djMax (~djMax@dsl093-190-107.nyc2.dsl.speakeasy.net) |
| 15:59.25 | djMax | joining two *'s with IAX2. Can I put "friend" in both iax.conf entries? |
| 16:01.36 | djMax | I'm getting "no authority found" |
| 16:02.14 | Zeeek | you need to get some authority |
| 16:02.33 | wasim | djMax: use user/peer while debugging |
| 16:02.50 | djMax | so the caller * is user, and the callee is peer right? |
| 16:02.55 | wasim | djMax: start with a IAX2/user:pass entry in extensions.conf |
| 16:03.00 | djMax | ok |
| 16:03.01 | wasim | djMax: then gradually get rid of the pass etc |
| 16:03.29 | wasim | djMax: they are both both, use user when someone registers to you, use peer when you want to make a call to them |
| 16:03.51 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 16:05.26 | djMax | ok. user/pass in extensions also gets no authority found |
| 16:08.17 | djMax | one strange thing is that it seems to be sending reject's long after I hang up |
| 16:08.21 | djMax | (5-10 seconds) |
| 16:09.55 | bjohnson_ | djMax: friend in both works for me |
| 16:10.14 | bjohnson_ | djMax: I use same username & secret in both iax.conf sections |
| 16:10.41 | djMax | yeah, same here. |
| 16:11.21 | djMax | do you have context settings? |
| 16:11.55 | bjohnson_ | btw .. was able to change my win98 shell to winfile.exe and it boots up. When I run iexplore.exe I get a dll error .. checking the file system reveals no such file by the name shown in the error .. downloading a fresh copy now |
| 16:12.13 | bjohnson_ | djMax: context settings? |
| 16:12.27 | djMax | which context the IAX call gets dumped into |
| 16:12.27 | *** join/#asterisk blitzrage (~blitzrage@d141-234-145.home.cgocable.net) |
| 16:12.40 | bjohnson_ | djMax: one is dynamic and it registers .. the other just uses host=(enter dns name here) |
| 16:13.05 | *** join/#asterisk Nivex (kjotte@user-0ce2jqe.cable.mindspring.com) |
| 16:13.08 | bjohnson_ | djMax: oh yes .. same as every iax incoming config |
| 16:13.44 | Zeeek | o iwho is using the asterisk iax2 provision command with templates? |
| 16:14.06 | christo | hmmmm poo |
| 16:14.21 | christo | has anybody got an example of using the Mysql() command? |
| 16:16.31 | Zeeek | take a quick look at this: |
| 16:16.32 | Zeeek | http://lists.digium.com/pipermail/asterisk-users/2004-November/072713.html |
| 16:17.55 | mtqh | zeek: yes? |
| 16:18.04 | Zeeek | yes what? IAXy ? |
| 16:18.34 | mtqh | sorry, wrong window, I say your nick and thought I was in another window |
| 16:18.47 | Zeeek | my question was that in the old provision method you have "register" and in the new I see "flags=register" |
| 16:19.47 | markit | I've updated the Wiki with some more info about "one touch recording" feature |
| 16:19.54 | djMax | so if I do Dial(IAX2/user:pass@1.1.1.1/1001) it should dial 1.1.1.1 using user and pass, and try to go to extension 1001 right? |
| 16:20.18 | djMax | is the name of the section in the "called *" important? |
| 16:20.20 | markit | anyone here can help me to make "disconnect" feature work, so I can then provide info to the community? |
| 16:20.22 | christo | Zeeek - which makefile is he referring to? That's exactly the error I'm getting |
| 16:20.35 | Zeeek | that error means you haven't added mysql |
| 16:21.09 | christo | what do you mean by 'added' mysql - I mean the mysql server is running on that machine, the db is installed and I can connect in the normal fashion |
| 16:21.16 | christo | is this something to to with the way asterisk was built? |
| 16:21.28 | Zeeek | You need to download the asterisk-addons to have mysql support now. It was |
| 16:21.28 | Zeeek | only moved to its own project due to licensing changes with MySQL. |
| 16:21.39 | bjohnson_ | djMax: with the username, secret, and host in iax.conf I just do Dial(IAX2/iaxsectionname/1001) |
| 16:21.48 | Zeeek | http://lists.digium.com/pipermail/asterisk-users/2004-November/072727.html |
| 16:22.07 | christo | ooh mmm ok |
| 16:22.13 | Zeeek | it's an addon |
| 16:22.13 | djMax | weird... So it seems that the name of the section is important, and basically the username doesn't matter. |
| 16:22.15 | Zeeek | ya see |
| 16:22.26 | Zeeek | np |
| 16:22.32 | djMax | so when I do dial(IAX2/section:secret@IP/exten) it works. |
| 16:22.39 | bjohnson_ | djMax: it uses the username FROM the section |
| 16:22.45 | Zeeek | I think there's a big future in Querying db with asterisk - it's a cool area |
| 16:23.09 | bjohnson_ | djMax: try Dial(IAX2/iaxsectionname/1001) |
| 16:23.16 | djMax | yeah, that's failing so far. |
| 16:23.22 | Zeeek | especially if you had like toll free numbers and could query a distant server and just rent that service out to a company |
| 16:24.15 | djMax | yep, so the username in the client iax.conf has to match the section name in the server iax.conf |
| 16:24.16 | netsurfer | bbl |
| 16:27.54 | djMax | more subtle than that I guess. Hrmph. |
| 16:28.23 | sd-tux | Zeeek: maybe you can help ?: i found out that only one of my phones is sending my name to caller in "Contact:" header... asterisk don't forward thes header to caller :( .. now i want to test if my phone can schow the name of called person ... for this i need to add a string in Contact header from asterisk .SipAddHeader seems to add a header to called phone but not calling phone :( .. how can i add a header in 180 ringing reply ? |
| 16:29.02 | file | the Contact head isn't used for that |
| 16:29.08 | file | callerid information is sent in the From header |
| 16:29.09 | Zeeek | sd-tux what os the callerid= stuff in you configs? |
| 16:29.49 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 16:31.16 | sd-tux | Zeeek: i need something like calledid not callerid :) |
| 16:31.37 | djMax | yeesh, I had it working and now I can't get back to it. :) |
| 16:31.43 | file | that's what the To field is for |
| 16:32.02 | Zeeek | sd-tux what exactly is missing? |
| 16:32.32 | mtqh | sd-tux search did on the wiki there is a long section on it |
| 16:32.47 | |Vulture| | you can pull it from a database |
| 16:33.38 | file | sd-tux: as for your initial query, I highly doubt your phone can do that... and the Contact header isn't what you want |
| 16:34.32 | sd-tux | file: thank you ... |Vulture| i searched the wiki but i don't know of this feauture .... can you send me the link.. |
| 16:34.43 | file | if you go mucking with it, weird things might happen |
| 16:34.47 | sd-tux | file i have Optipoint 400 |
| 16:35.05 | |Vulture| | sd-tux: I have never done it just know its possible |
| 16:36.07 | sd-tux | mtqh: cn you send me the link ? or maybe the name of this "feauture" :) ? |
| 16:37.51 | sd-tux | file: the contact header is the only place where i can see my name being sent to the caller Sip UA |
| 16:38.38 | file[laptop] | I don't quite understand what you want to do |
| 16:38.42 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 16:39.01 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 16:40.16 | sd-tux | file[laptop]: i want to send the name of called person to caller... if i have to change "To:" header for this how can i do this ? |
| 16:40.28 | file[laptop] | will your phone do anything with it? |
| 16:42.30 | sd-tux | file[laptop]: i want to test it :) i hope.... |
| 16:42.39 | file[laptop] | I doubt it, never seen a phone that will |
| 16:43.22 | lohelle | has anyone tried avaya 4630 screenphone => asterisk.. looks good! |
| 16:43.24 | sd-tux | file[laptop]: cisco phones ? |
| 16:43.42 | file[laptop] | cisco phones don't... but because you have the separate line appearances, you know what line it came in on |
| 16:44.11 | Zeeek | some kind of scheme where asterisk instigates the calls and the person doesn't know who he's calling? |
| 16:44.35 | file[laptop] | Zeeek: I think he wants it to say exactly what was called, not who is calling... rather confusing |
| 16:45.06 | Zeeek | for example I call 0102030405 ans I see that on my phone? |
| 16:45.30 | Zeeek | ironically, I think I do on the BT102 :) |
| 16:45.47 | Zeeek | but I guess this is not a dialed number |
| 16:45.52 | file[laptop] | really confusing |
| 16:46.01 | sd-tux | file[laptop]: this is very popular feauture in the company where i work :) you dial the number and you kno the name of the person you are speaking with... |
| 16:46.07 | Mother_ | hi all |
| 16:46.08 | file[laptop] | sd-tux: oh |
| 16:46.28 | file[laptop] | sd-tux: but which end... the person you called gets your name, or you get their name when you dial the number? |
| 16:46.41 | file[laptop] | if you mean the latter, no - SIP doesn't work like that... 'nor do any of the SIP phones |
| 16:46.48 | sd-tux | file[laptop]: the caller sees my name |
| 16:46.55 | Mother_ | I have a problem with fax: it's detecting the incoming fax, but even though I have an exten => fax,1,.... it says "Fax detected, but no fax extension" |
| 16:47.16 | file[laptop] | your phone might have the option of looking it up in the address book/phonebook and displaying it though |
| 16:47.25 | Zeeek | the call left the conext where fax was defined? |
| 16:47.39 | *** join/#asterisk voipjet (JJ@modemcable166.107-80-70.mc.videotron.ca) |
| 16:47.40 | Mother_ | does the fax extension need to be in the same context? |
| 16:47.56 | voipjet | Hi! |
| 16:47.56 | WifiFred | Hi! |
| 16:48.16 | Mother_ | i.e. I have four PSTN lines, each with it's own context, and then at the end of extensions.conf I added the fax extension |
| 16:48.31 | Zeeek | I had that fax probelm and I'm trying to rememebr when I got that - because I did |
| 16:48.31 | sd-tux | file[laptop]: maybe but I don't know this... i hoped to find a sip header which is used for this ... |
| 16:48.45 | file[laptop] | sd-tux: nope |
| 16:48.48 | *** join/#asterisk coool (~ghjhghjgh@63.168.168.207) |
| 16:48.57 | Mother_ | OK |
| 16:49.06 | file[laptop] | sd-tux: but just because you added it to a SIP header (which might freak out asterisk OR your phone) doesn't mean your phone would use it |
| 16:49.07 | Zeeek | let me take a look at my files |
| 16:49.11 | Mother_ | I'll try to add the fax extension inside the context |
| 16:49.14 | coool | i need help pls with fwdOUT |
| 16:50.01 | sd-tux | file[laptop]: :) ok ... I was told that cisco can do this with skinny and CCM and wanted to test this with asterisk :) |
| 16:50.03 | Mother_ | thanks |
| 16:50.11 | file[laptop] | sd-tux: skinny isn't SIP |
| 16:50.12 | Zeeek | Mother I think it may be that you want fax extension to be in the same context as when the call is answered |
| 16:50.18 | Mother_ | HA! |
| 16:50.19 | file[laptop] | sd-tux: skinny gives much more control of the phone to the server |
| 16:50.33 | Mother_ | Zeeek: you're right, just tried that and it works, thanks a bunch! |
| 16:50.34 | Zeeek | Answer() 'ed |
| 16:50.35 | coool | i need help pls with fwdOUT plssss |
| 16:50.41 | Zeeek | cool |
| 16:50.44 | file[laptop] | coool: just ask your question, we aren't psychic |
| 16:51.04 | Zeeek | Mother_ cool |
| 16:51.09 | Zeeek | cool ask away |
| 16:51.15 | Zeeek | coool ask away |
| 16:51.16 | sd-tux | file[laptop]: I know this... it would be nice to have this in SIP too |
| 16:51.34 | file[laptop] | sd-tux: it's reality, deal with it :p |
| 16:52.02 | coool | file[laptop] sorry. its simple Q. i did sign in fwdOUT.and im downloading now asterisk to use it. so i can later make free calls from this software to land phones FREE without paying anything? |
| 16:52.11 | coool | Zeeek |
| 16:52.16 | Zeeek | coool |
| 16:52.35 | Zeeek | nothing in this life is free |
| 16:52.36 | lohelle | ne1 have example of web page that can issue commands(command line) to the asterisk server. I was thinking about something like a text input field and an OK buttom. When inputting phone number and click OK then a script shold run and generate call file (the script I allready have). |
| 16:52.37 | Zeeek | ever |
| 16:52.46 | Alric | coool: To other FWD users, but you're still paying for bandwidth :) |
| 16:52.55 | Mother_ | well, the first few lines of the fax were OK, then after that it's all condensed into a blur :D |
| 16:52.55 | Zeeek | lohelle manager interface |
| 16:52.58 | file[laptop] | Alric: he's talking about fwdOUT, something different |
| 16:53.05 | coool | Zeeek so what is fwdOUT for? |
| 16:53.07 | Alric | fwdOUT? |
| 16:53.13 | coool | yeah |
| 16:53.14 | Zeeek | what is fwdOUT? |
| 16:53.20 | coool | fwdOUT.com |
| 16:53.20 | file[laptop] | http://www.fwdout.net/ |
| 16:53.21 | Alric | hrm, bellster... |
| 16:53.23 | file[laptop] | formerly bellster |
| 16:53.34 | Mother_ | did Bell get it's way with the name thing? |
| 16:53.37 | coool | sorry .. yeah fwdout.net |
| 16:53.37 | Mother_ | I presume it did :) |
| 16:53.40 | Zeeek | oh? Well bellster works but many think it's a bad idea |
| 16:54.16 | coool | in fwdout site they said: Call anywhere in the world with Fwdout.Net's |
| 16:54.17 | coool | Phone Sharing Service! |
| 16:54.29 | coool | Freely call any ordinary phone with fwdOUT |
| 16:54.38 | Zeeek | you can if you share your own local phone line coool |
| 16:55.03 | Zeeek | you have a phone line and hardware to hook up to asterisk? |
| 16:55.18 | coool | Zeeek so i still need hardware ? not only software |
| 16:55.27 | file[laptop] | coool: read the FAQ |
| 16:55.33 | Zeeek | you need to give time on your phone line to make calls |
| 16:55.36 | file[laptop] | specifically, "How does it work?" |
| 16:55.46 | Zeeek | read this: NOTHING is FREE |
| 16:55.48 | coool | Zeeek pls visit http://www.fwdout.net/web/ToSignup |
| 16:55.49 | Zeeek | ever |
| 16:56.04 | Zeeek | I'm already signed up but I stopped using it after a day :) |
| 16:56.05 | coool | ic |
| 16:56.16 | *** join/#asterisk sandnigg0r (~trilluser@66-55-197-254.gwi.net) |
| 16:56.35 | Nugget | you guys just aren't speaking coool's language. Try: "nthn z free 2 u nless u cn give bk" |
| 16:56.36 | Zeeek | when you sign up you need to provide a route for others to call using YOUR phone line |
| 16:56.37 | file[laptop] | coool: needless to say you have to give calls to make calls :p |
| 16:57.06 | coool | bad./. i used to use pulver.Communicator .. its was FREE to PSTN in USA and canada |
| 16:57.09 | coool | not anymore :( |
| 16:57.12 | Zeeek | anyway it's kind of a dead topic - asterisk won't help make free calls except through FWD etc |
| 16:57.33 | Nivex | I miss the FWD<->Vonage gate. Any idea if that's coming back? |
| 16:58.19 | Godsey | I use ipkall for free did and nufone for outbound |
| 16:58.26 | Godsey | I have yet to make an outbound call :) |
| 16:58.34 | Zeeek | Nugget a worthy suggestion |
| 16:58.57 | coool | Zeeek any advice pls to make free call from pc to phone in usa and canada? |
| 16:59.27 | Zeeek | FWD has holiday specials which are free calling |
| 16:59.36 | djMax | Dial(IAX2/myname:mysecret@1.1.1.1/1001)... that will look for a section named myname on the remote server right? |
| 16:59.43 | Zeeek | you can call US/CAN for 1.4c/minute |
| 16:59.50 | file[laptop] | djMax: yes. |
| 16:59.54 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 17:00.06 | Zeeek | That's less than $1 an hour |
| 17:00.12 | coool | Zeeek i mean free like pulver.Communicator |
| 17:00.23 | Zeeek | nothing is free. ever. |
| 17:00.27 | file[laptop] | nowhere is free, so deal with it |
| 17:00.34 | Zeeek | except when it is. for a minute. |
| 17:00.40 | file[laptop] | just a minute. |
| 17:00.59 | coool | i mean for few minutes, not all time |
| 17:01.03 | Nivex | Access to the PSTN will never be free. IP to IP is only the cost of the IP connection, which most people consider free because they are already paying for that bandwidth to do other things. |
| 17:01.18 | Nivex | coool: You want a few free minutes a day, check out sipphone.com. |
| 17:01.19 | *** join/#asterisk lilneon (~tj_r3@cuscon11838.tstt.net.tt) |
| 17:01.22 | lilneon | hi |
| 17:01.24 | lilneon | good morning all |
| 17:01.39 | coool | Nivex aha .. thnx |
| 17:01.40 | Zeeek | but seriously: anyone that has enough bandwidth can afford 1.5c a minute for a call - unless you are stealing the bandwidth |
| 17:02.07 | Zeeek | entropy means that nothing is ever free and never will be |
| 17:02.12 | coool | pls check www.mobilecaller.com <-- offer 10min FREE world calls .. i need somthing like that |
| 17:02.32 | file[laptop] | trial error. |
| 17:02.35 | Zeeek | www.tradeFreeCallsForSex.com |
| 17:02.58 | |Vulture| | lol |
| 17:03.03 | Zeeek | gee, the site is down |
| 17:03.03 | djMax | and the iax section on the server should be type=friend, same secret, and that's it right? |
| 17:03.14 | Zeeek | or going down :) |
| 17:03.15 | file[laptop] | djMax: sure. |
| 17:03.24 | |Vulture| | Zeeek: harharhar :P |
| 17:03.30 | Mother_ | Zeeek: did you have problems with the page being cut short (very short in my case) |
| 17:04.01 | Zeeek | Mother_ faxing is not ready for primetime business - it's a wonderful bit of work but has many problems |
| 17:04.04 | Zeeek | so, yes |
| 17:04.06 | coool | <PROTECTED> |
| 17:04.11 | Mother_ | hrrrrmmmmmmggggrrrmffff |
| 17:04.20 | Mother_ | oh well |
| 17:04.22 | lilneon | hey guys, i setup an sshtunnel with my machine at wrk and my asterisk box home.. how do i get a softphone to use it now? |
| 17:04.24 | Zeeek | alwasy either down or going down :( |
| 17:04.47 | Zeeek | Mother_ I received exactly two perfect faxes - one spam and the other by accident |
| 17:04.52 | Mother_ | LOL! |
| 17:04.57 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 17:05.10 | Zeeek | and I HATE SPAM esp spam faxes which are clearly illegal in Eu |
| 17:05.18 | Mother_ | yeah, we get them quite often |
| 17:05.18 | coool | www.mobilecaller.com <-- offer 10min FREE world calls .. i need somthing like that plss |
| 17:05.37 | Mother_ | that's why I wanted to use callerid to kill those, only whitelisted numbers can send |
| 17:05.53 | file[laptop] | Zeeek: as is mine |
| 17:05.56 | Mother_ | but if it's going to be this iffy...I need at least *some* reliability |
| 17:05.58 | |Vulture| | * + Fax == Problems |
| 17:06.03 | file[laptop] | coool: then use them and be quiet |
| 17:06.04 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 17:06.19 | file[laptop] | we aren't going to give you free calling, we aren't going to find you free calling, we don't know where to get free calling |
| 17:06.19 | Zeeek | Mother_ but a $40 fax machine, it'll be way more reliable |
| 17:06.34 | Mother_ | Zeeek: I have that one already, I picked it up just in case :D |
| 17:06.40 | coool | file[laptop] ok ok :) clam down. thanks anyway for help |
| 17:06.46 | tzanger | |Vulture|: untrue. voip+fax = problems |
| 17:06.49 | Alric | The most reliable faxing I've seen w/ * is T100P + channel bank -> Fax. |
| 17:06.51 | coool | Zeeek thanks 2 for ur help. and sorry 4 bothering |
| 17:06.53 | file[laptop] | NEXT!!! |
| 17:07.09 | Mother_ | the problem in this particular setup is that they want to route 4 PSTN lines to a remote * on a new office, as they cannot have the numbers changed to the new address |
| 17:07.13 | tzanger | Alric: agreed |
| 17:07.14 | file[laptop] | somebody ask me a question that isn't already answered somewhere |
| 17:07.17 | Mother_ | and the line is shared between voice and fax |
| 17:07.21 | file[laptop] | one that makes me think |
| 17:07.27 | Zeeek | Mother_ here's the prob. I have asked customrs to fax my asterisk and the first one I asked has a nice fax and it won't work AT ALL with spandsp - yet a $30 shareware on WIn gets every fax ever thrown at it |
| 17:07.38 | djMax | argh, I'm still getting "no authority found", no clue why. |
| 17:07.40 | |Vulture| | tzanger: going from fax--voipgateway--*--voip provider--pstn--fax has lots of problems |
| 17:07.54 | Zeeek | so, either the X100P can't do it or the software can't do it. Result is the same |
| 17:07.58 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 17:08.00 | tzanger | |Vulture|: as I said, voip+fax = problems |
| 17:08.10 | Zeeek | djMax you're reloading each time of course? |
| 17:08.17 | Mother_ | Zeeek: agreed, I think the problem is that they try to do everything in low-level DSP, which is tricky at best |
| 17:08.24 | djMax | yeah, restarting even. |
| 17:08.35 | file[laptop] | djMax: are you going to get me to write up examples? |
| 17:08.41 | Zeeek | I don't doubt that what is working now is already brilliant, only hey, I can't use it for business |
| 17:08.49 | djMax | The super annoying thing is that it was working for like 5 seconds. |
| 17:08.51 | Mother_ | agreed |
| 17:09.24 | Zeeek | I have a $15/month acount with jfax and I want to be able to replace it within one year - so I do hope for imporvement |
| 17:09.32 | djMax | so on the server, the section name is "mysection", type friend, with a secret, a host, and a context. Nothing else. |
| 17:09.39 | Zeeek | imporvement, yes that's it |
| 17:09.44 | Mother_ | will * still catch the crappy ring signal those fax discriminators? I may have to install one |
| 17:09.51 | file[laptop] | djMax: one sec |
| 17:10.01 | Zeeek | "this is to imporve you that your license has expired" |
| 17:10.08 | djMax | on the client, I'm just dialing with no "section", IAX2/mysection:secret@IP/1001. |
| 17:10.14 | djMax | ok, I wait |
| 17:10.28 | Zeeek | Mother_ I'm sure you can still route the fax to a fax machine using detection that seems to work well |
| 17:10.32 | file[laptop] | djMax: http://pastebin.ca/5350 |
| 17:11.02 | Mother_ | ah OK, problem is I have four FXO boards on this TDM, I'll have to move things around a bit |
| 17:11.25 | Zeeek | djMax note that nufone uses two section named [nufone] in their suggested setup |
| 17:11.26 | Mother_ | I could swap one for an FXS and divert to that |
| 17:11.45 | Zeeek | Mother_ that's what I meant - sorry |
| 17:11.53 | Zeeek | I forgot not everyone has FXS |
| 17:11.54 | Mother_ | Zeeek: OK thanks |
| 17:11.57 | file[laptop] | if you want a peer on the server... add a peer entry, don't use a friend... less headaches |
| 17:12.52 | Zeeek | nufone did uses user, secret context - that's it |
| 17:13.32 | Zeeek | peer, host, secret for outgoing |
| 17:13.46 | djMax | nope, the example fails. |
| 17:14.01 | file[laptop] | then you've got something else mucked up |
| 17:14.43 | Zeeek | djMax, do you have any iax entities in iax.conf that you can copy and modify? |
| 17:14.45 | djMax | yeah, in iax2 debug I don't see rejects now, I see "new", but still doesn't connect. |
| 17:15.08 | file[laptop] | what does it do? |
| 17:16.05 | djMax | client says IAX2/mybox/2 is circuit-busy |
| 17:16.25 | file[laptop] | what does the server say? |
| 17:16.46 | djMax | just shows a "NEW" and a bunch of auth requests, but no rejects. |
| 17:16.57 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 17:17.09 | file[laptop] | turn off iax2 debug |
| 17:17.11 | file[laptop] | and try again |
| 17:18.11 | djMax | same |
| 17:18.35 | file[laptop] | can I have access? and if this is something stupid I'm going to thwap you |
| 17:18.42 | Zeeek | heh |
| 17:18.52 | djMax | let me make an IAX entry for you |
| 17:18.57 | Zeeek | has to be one of those forest for the trees deals |
| 17:19.05 | djMax | absolutely, especially since it worked. |
| 17:19.10 | file[laptop] | nah I mean SSH access, I have a sneaking suspicion you're overlooking something |
| 17:19.14 | fa | where can i find hfc card for 4x BRA ? |
| 17:19.17 | djMax | ok |
| 17:19.19 | Zeeek | Mother_ 4 FXO? Not a TDM? |
| 17:19.51 | lohelle | is there an example web page/script to issue commands to linux CLI? (just not an asterisk question). Must login first... apache webserver.. |
| 17:22.22 | Mother_ | Zeeek: TDM400P with 4 FXO boards |
| 17:22.48 | Zeeek | or ak |
| 17:22.53 | Zeeek | k |
| 17:23.18 | *** join/#asterisk Nukemizer (Nuke@66.237.85.58.ptr.us.xo.net) |
| 17:25.54 | *** join/#asterisk jeofrey (~jeofrey@202.160.45.29) |
| 17:26.40 | wasim | TDM40, i.e. |
| 17:26.47 | jeofrey | hi all |
| 17:26.57 | jeofrey | anyone can help please |
| 17:26.59 | jeofrey | http://www.pastebin.com/238490 |
| 17:27.35 | slePP | file[laptop]: my channel bank still doesn't work right. silly asterisk. |
| 17:27.48 | file[laptop] | okay people, WHEN I GIVE EXAMPLES FOLLOW THEM TO THE DOT |
| 17:29.47 | file[laptop] | slePP: :( |
| 17:30.57 | file[laptop] | NEXT!!! |
| 17:31.39 | file[laptop] | I'm afraid not dear |
| 17:31.54 | bjohnson_ | I have this bump on my foot. file[laptop] .. could you tell me what it is? |
| 17:32.10 | file[laptop] | bjohnson_: it's where I injected you with a sedative |
| 17:32.36 | bjohnson_ | WHAT ?? You shouldn't ha zzzzzzzzzzzzzzz |
| 17:32.46 | file[laptop] | anyone want a foot? brain? liver? |
| 17:32.59 | bjohnson_ | no brains here |
| 17:33.04 | file[laptop] | didn't think so |
| 17:33.07 | mikegrb | yes, how many miles on the brain? |
| 17:33.14 | bjohnson_ | none |
| 17:33.18 | bjohnson_ | I walk with my feet |
| 17:33.25 | mikegrb | I'll give $1 |
| 17:33.33 | file[laptop] | I have $1, do I hear $1.25? |
| 17:33.49 | file[laptop] | eh? EH? $1.25? it's a steal! |
| 17:34.01 | file[laptop] | Sold to mikegrb for $1! |
| 17:34.06 | mikegrb | Ld |
| 17:34.37 | mikegrb | :D |
| 17:35.00 | wasim | there's one born every minute ... |
| 17:35.04 | djMax | Well, there's no question the thwack was worth making the problem go away. Still scratching my head because I could swear I was in this config before. Probably missed a reload or something. |
| 17:35.47 | file[laptop] | wasim: what, an asterisk user? |
| 17:35.57 | djMax | whoa! Got it to fail again. To a different extension. |
| 17:36.21 | wasim | does the # get commission on sales? |
| 17:37.04 | file[laptop] | djMax: You have exhausted your one question for today. Please try your question again tomorrow, thank you! |
| 17:37.09 | Nukemizer | is ther a way to go straight to voice mail. lets say i want to leave a message for a buddy but not ring his phone when it is late. Just go directly to his mailbox |
| 17:37.27 | djMax | no more questions now, just notices in case I find something interesting. |
| 17:37.38 | file[laptop] | Nukemizer: use the dialplan! |
| 17:37.42 | djMax | now that I have a working base I can handle any non-bug. :) |
| 17:37.58 | lohelle | ne1 know when automon and atxfer (featurs.conf) is going to STABLE release? (then maybe bristuff will be patched into that one) |
| 17:37.59 | wasim | Nukemizer: Voicemail(25) |
| 17:38.06 | lohelle | features.conf |
| 17:38.17 | file[laptop] | lohelle: stable is for BUG FIXES, repeat after me BUG FIXES |
| 17:38.30 | file[laptop] | lohelle: the next major release may make the latest cvs head stable... who knows |
| 17:38.46 | file[laptop] | I swear I've turned into bkw |
| 17:38.53 | file[laptop] | anywhoser |
| 17:38.55 | file[laptop] | NEXT!!! |
| 17:39.07 | lohelle | OK! I learn something new every day here! |
| 17:39.41 | Nukemizer | thank you both, trying :) |
| 17:42.08 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 17:42.37 | bjohnson_ | errr .. terminology. Eventually HEAD will get branched off to a new version .. after a beta period, THAT will become STABLE |
| 17:43.38 | bjohnson_ | that branching of HEAD to ceate a new version is not scheduled to happen .. but it may be a sudden thing .. all of a sudden .. new version branch for testing |
| 17:43.38 | blitzrage | More than likely 1.2 stable |
| 17:44.00 | blitzrage | I suspect 1.2 will be released at Astricon Europe (my educated guess) |
| 17:44.14 | Jedirl | when will it be? |
| 17:44.20 | blitzrage | there is no date |
| 17:44.29 | lohelle | My point was that I need bristuff because I use two ISDN cards in NT-mode.. But I would also like to have (at least) the atxfer feature.. :) |
| 17:45.42 | blitzrage | file[laptop]: I can add a -r v1-0-x version tag to download the particular version from CVS right? (as opposed to the latest stable) |
| 17:47.17 | macTijn | did anyone see http://www.aefirion.org/ ? |
| 17:47.28 | blitzrage | file[laptop]: just tried it... yep, I can! :) |
| 17:47.51 | file[laptop] | tags are nifty |
| 17:48.32 | blitzrage | file[laptop]: sure are. Gotta add that information to the installation chapter regarding CVS. I just have how to checkout -r v1-0 |
| 17:48.41 | file[laptop] | you can do date too |
| 17:48.55 | blitzrage | file[laptop]: how do you format that again? I should put that in there. |
| 17:49.14 | blitzrage | haha, yah I do that too |
| 17:51.16 | file[laptop] | Sunday, how about that Sunday |
| 17:51.18 | file[laptop] | yup yup yup |
| 17:51.26 | blitzrage | file[laptop]: will look it up and add it to the docs |
| 17:54.04 | *** join/#asterisk [alex] (~[alex]@201.137.118.84) |
| 17:54.31 | file[laptop] | toot toot yeah beep beep |
| 17:58.01 | djMax | that base eroded fast. Same config not working anymore. Must be a net or other problem i guess. |
| 17:59.02 | *** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net) |
| 18:00.53 | *** part/#asterisk lilneon (~tj_r3@cuscon11838.tstt.net.tt) |
| 18:01.57 | djMax | bah, I think it's a routing problem. Stupid linux q: how do you control routing pref order for default gateways on dhcp interfaces? |
| 18:02.14 | Syncros | :) |
| 18:04.33 | bjohnson_ | you should only have 1 default gateway |
| 18:04.47 | bjohnson_ | everything else should be a pattern match |
| 18:05.23 | djMax | yeah, problem is the local interface feeds back a gateway in DHCP since there is a firewall, but this is a dual-homed machine |
| 18:06.17 | djMax | clearly a bad setup, so you're right, let me fix the setup instead. |
| 18:11.47 | *** join/#asterisk Rick_Hunter (~rhunter@03-193.008.popsite.net) |
| 18:12.46 | JohnAB | you can have more than one default router in linux, but that would be unusual |
| 18:12.57 | Qwell | You sure? |
| 18:13.02 | JohnAB | yeah |
| 18:13.09 | JohnAB | but you need to set up rules for which one to use |
| 18:13.12 | silik0n | *yawn* |
| 18:13.15 | JohnAB | you can do that with iproute2 |
| 18:13.30 | JohnAB | you might want to do that if you're load balancing between interfaces |
| 18:13.32 | Qwell | Then it wouldn't really be a default route, if you have to setup rules. |
| 18:13.34 | djMax | yep, was a routing problem. Maybe I can sidestep file's thwack ever so slightly. |
| 18:13.55 | JohnAB | well that's a matter of semantics |
| 18:14.35 | JohnAB | as far as linux is concerned you effectively have 2 default routes that you use alternately or whatever, but you also need rules to say if it comes in on interface A, it goes out on interface A etc. |
| 18:15.10 | Jedirl | that's called policy routing |
| 18:15.25 | *** join/#asterisk netsurfer (netsurfer@81-6-248-21.dyn.gotadsl.co.uk) |
| 18:15.53 | Jedirl | you can mark packets with iptables and route them as you want depending not only on destination but on source, port, or anything |
| 18:16.10 | JohnAB | yes indeed |
| 18:16.16 | Qwell | marking packets is fun |
| 18:16.55 | shido6 | ZZzzzZz |
| 18:17.22 | mikegrb | I like to mark them with sharpies[tm] |
| 18:17.49 | Qwell | mikegrb: I figured out my wiring, heh |
| 18:18.09 | mikegrb | excellent |
| 18:18.29 | Qwell | mikegrb: I guess its an RJ12 plug, and cat3(with 3 pairs), where only one pair is used on each wire. It comes in as orange, leaves as blue, and in the other room, it comes in blue, and doesn't leave |
| 18:18.34 | mikegrb | I'm sitting here trying to widdle 1.25 gb of photos down to 700 mb for burning on cd |
| 18:18.51 | mikegrb | excellent |
| 18:19.13 | mikegrb | I took almost 500 photos yesterday at a mardi gras parade |
| 18:19.30 | Qwell | well... |
| 18:20.02 | Qwell | for i in `find . | grep -v breasts`; do rm $i; done |
| 18:20.17 | mikegrb | heh |
| 18:20.34 | mikegrb | this was a family parade |
| 18:20.42 | lycklaybeeuh | grep "breasts" * | wc -l |
| 18:20.46 | Qwell | s/breasts/float |
| 18:20.55 | mikegrb | though it was nice to be able to drink beer on the street |
| 18:21.02 | mikegrb | Qwell: heh, just about all of them |
| 18:21.33 | mikegrb | this parade was frik'n long too, we left at three hours and it was still going! |
| 18:21.47 | Nukemizer | when using SIP phones can "all call Page" be accomplished ? |
| 18:21.51 | blitzrage | does anyone know if there is anything to worry about when checking out Asterisk from CVS with the -D flag? Do I need to use the -f flag? |
| 18:22.24 | mikegrb | oh, and when the news channel went buy they had a video and still photographer walking along, the still photographer was using same camera as me so I took a picture of him :D |
| 18:22.42 | *** join/#asterisk jeofrey (~jeofrey@202.160.45.29) |
| 18:23.00 | jeofrey | any one can help me please |
| 18:23.06 | Qwell | mikegrb: Yours must be fairly pricey then, eh? |
| 18:23.36 | jeofrey | i always ahve rpoblem in chan_sip |
| 18:23.51 | jeofrey | http://www.pastebin.com/238499 |
| 18:23.53 | ariel_ | jeofrey, what is the problem that your having? |
| 18:24.05 | jeofrey | i have paste it in pastebin |
| 18:24.40 | ariel_ | jeofrey, What are you trying to do? |
| 18:25.02 | ariel_ | what you posted is that there is no channel |
| 18:25.18 | jeofrey | i have sip ipphone and i want to connect it to asterisk |
| 18:25.31 | jeofrey | but i always got that notice |
| 18:25.36 | ariel_ | which phone. |
| 18:25.48 | jeofrey | grandstream |
| 18:26.02 | jeofrey | and sipsoftware |
| 18:26.17 | ariel_ | ah ok what is your sip.conf setting for the phone and what are you dialing rules for it? Pastebin it please |
| 18:26.22 | jeofrey | im using sipphone elite |
| 18:26.29 | shido6 | are you using friend |
| 18:26.33 | shido6 | or users and peers in sip.conf? |
| 18:26.43 | jeofrey | ok |
| 18:26.53 | shido6 | sip show peers |
| 18:26.58 | shido6 | can u see your peer? |
| 18:27.03 | shido6 | sip show users |
| 18:27.06 | shido6 | can u see your user? |
| 18:27.15 | jeofrey | please hang on |
| 18:27.27 | shido6 | I will eventually fall off , my grip is slipping |
| 18:27.39 | shido6 | :) |
| 18:31.16 | jeofrey | i dont have anything any setup inside of the sip conf but because its using bluesip_sip.agi |
| 18:31.54 | ariel_ | what is bluesip_sip.agi? |
| 18:31.54 | shido6 | then smack bluesip_sip.agi and add a user in your sip.conf |
| 18:32.01 | jeofrey | then i have to register in webpage before i can have the account |
| 18:32.08 | blitzrage | shido6: are you in Vancouver or Toronto? Pretty sure you're in Canada somewhere right? |
| 18:32.18 | blitzrage | file[laptop]: lol |
| 18:32.25 | file[laptop] | don't make me hurt you Mr. Madsen |
| 18:32.43 | shido6 | a user .. then a peer |
| 18:32.43 | shido6 | and issue a sip reload at the CLI |
| 18:32.43 | file[laptop] | I'll do it!!! |
| 18:32.54 | jeofrey | can you give me please a sample of sip conf setup |
| 18:33.15 | shido6 | im somewhere in Canada not in Toronto nor in Vancouver, I have a quad span in Toronto lit up at Front Street |
| 18:33.15 | blitzrage | ~user |
| 18:33.16 | jbot | hmm... user is currently detached. Talk to this user upon their return. You will now be ignored. [HackFactor Elite 2.0], or a synonym for moron |
| 18:33.16 | blitzrage | hrmmmmm, that really should return something |
| 18:33.22 | blitzrage | bah! |
| 18:33.28 | shido6 | jeofrey ...... look in /usr/src/asterisk/configs for your sample configs |
| 18:33.50 | jeofrey | ok ok |
| 18:34.55 | *** join/#asterisk Fanguin (~Fanguin@p5081907B.dip0.t-ipconnect.de) |
| 18:37.31 | Qwell | jeofrey: http://pastebin.ca/4810 |
| 18:37.31 | Qwell | file++ |
| 18:37.57 | file[laptop] | dejavu |
| 18:38.17 | Qwell | hmm, lag is fun |
| 18:38.49 | shido6 | uhhh |
| 18:38.50 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 18:38.54 | shido6 | peers dont need contexts |
| 18:38.57 | shido6 | only users |
| 18:39.11 | shido6 | everything else is great |
| 18:39.11 | ariel_ | users are for incoming peers are for outgoing.... |
| 18:40.46 | implicit | hi |
| 18:41.51 | jeofrey | ok thanks for the sip conf sample |
| 18:42.16 | Qwell | jeofrey: thank file |
| 18:42.58 | ariel_ | so is anyone going to be watching the game tonight? Any picks? |
| 18:45.16 | *** join/#asterisk benno2 (~benno2@host178-117.pool80117.interbusiness.it) |
| 18:45.40 | benno2 | hi, anyone knows if you can set the volume of the budgetone bt101 ringer |
| 18:46.01 | Zeeek | shido6 I think you wanted to say something about IAXy the other day - I have a question: where do I find the parameters for the asterisk provisioning template ? |
| 18:46.13 | Zeeek | benno2 yes down and up arrows |
| 18:46.21 | *** join/#asterisk niZon (~ilt@S0106deadbeef6977.wp.shawcable.net) |
| 18:46.22 | Zeeek | with the phone ON hook |
| 18:46.38 | Zeeek | I see flags=register |
| 18:46.54 | Zeeek | where would dhcp or the ip go ? |
| 18:47.10 | Zeeek | in the provision template iaxyprov.conf? |
| 18:47.11 | *** join/#asterisk Strom_TM (~Strom_TM@office4.tmcs.net) |
| 18:48.19 | Strom_TM | yow. I upgraded from stable to cvs-head and it broke the monitor command |
| 18:48.57 | blitzrage | probably something in head is much different now |
| 18:49.11 | Zeeek | head is head is head |
| 18:49.19 | implicit | nope |
| 18:49.22 | silik0n | head is good |
| 18:49.34 | Zeeek | it can be |
| 18:49.41 | Zeeek | hello blitz |
| 18:49.46 | *** join/#asterisk shepherd (~matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 18:50.38 | ctooley | I think I've finally gotten a SIP Proxy to handle inbound Termination for me. |
| 18:51.58 | ctooley | I wish I had some way to make Origination do something similar. Make all calls come from one proxy but then do some kind of redirect so the proxy gets taken out of the middle. |
| 18:52.00 | *** join/#asterisk itnomad (~jackal@199.89.146.110) |
| 18:52.09 | ctooley | Hi jack |
| 18:52.42 | blitzrage | wow... had a power failure last night, and not Asterisk won't start.... |
| 18:52.44 | blitzrage | now* |
| 18:53.00 | ctooley | what's it say it can't start on? |
| 18:53.00 | Qwell | If I were to make a dpdt relay (http://www.voip-info.org/tiki-index.php?page=Asterisk%20failover), what voltages should I be looking for? |
| 18:53.01 | itnomad | hey c, what up? |
| 18:53.33 | *** join/#asterisk pepepedo (~poop@OL67-166.fibertel.com.ar) |
| 18:53.47 | pepepedo | hi , someone may help me please |
| 18:54.50 | ctooley | pepepedo, since I don't know what's wrong it's impossible to help you |
| 18:55.16 | wasim | pepepedo: you were supposed to put the cream in a little bit at a time |
| 18:55.42 | ctooley | wasim, I don't know how to control the flow of cream, it all comes out at onc |
| 18:56.05 | ctooley | OH, you meant coffee... yeah, stir it in slowly |
| 18:56.07 | pepepedo | Hi , I recently install mysql for dip-friend |
| 18:56.30 | ctooley | no, he meant dipping a friend, I was right the first time. |
| 18:57.06 | pepepedo | everithing work fine, i can do calls from my sip phone perfect , but when I call the sip-friends from dial plan |
| 18:57.41 | pepepedo | whith something like exten => 8XXX,1,Dial(SIP/${EXTEN},20) |
| 18:57.55 | pepepedo | say extention unknown |
| 18:58.37 | pepepedo | if I do somemething like this exten => 8130,1,Dial(SIP/8130,20) |
| 18:58.43 | pepepedo | work perfect |
| 18:58.49 | Qwell | try _8XXX |
| 18:58.58 | pepepedo | but I dont wnat put all my sip cliente in the dial plan |
| 18:59.10 | pepepedo | ok ... |
| 18:59.12 | pepepedo | let me see. |
| 18:59.13 | Qwell | pepepedo: read up on what _ does |
| 18:59.22 | Qwell | There is an explanation in the sample config |
| 19:02.17 | pepepedo | thanks _8XXX woark fine |
| 19:02.23 | pepepedo | thanks a lot!! |
| 19:02.35 | Qwell | now tell me why it worked |
| 19:02.38 | pepepedo | (sorry for my poor english) |
| 19:02.54 | pepepedo | where i can read about it? |
| 19:03.03 | Qwell | in the sample extension config |
| 19:06.14 | blitzrage | ctooley: pbx_dundi fails for some reason |
| 19:06.21 | pepepedo | I found it .. thanks again!! |
| 19:08.42 | ctooley | blitzrage, do you have a dundi.conf? |
| 19:09.11 | wwalker_ | I can originate a call via the manager, how via manager, do I spawn an AGI from tat call? |
| 19:09.50 | Moc | Anyone use Hylafax in here ? |
| 19:09.53 | ctooley | wwalker_, I suppose you want to do something silly like have the manager call someone, and fire off an AGI to deal with the channel after they answer |
| 19:12.09 | *** part/#asterisk Corvin (~zbysio@chello084010031149.chello.pl) |
| 19:13.04 | liquidno2 | anybody here get spandsp to compile under Debian sarge? |
| 19:13.42 | markit | anyone here can help me to make "disconnect" feature work (features.conf), so I can then provide info to the community? |
| 19:15.17 | blitzrage | ctooley: yah, the computer just rebooted, nothing changed ;) |
| 19:15.28 | blitzrage | ctooley: just going to reinstall anyways, my CVS version was from November |
| 19:16.01 | ctooley | blitzrage, did you read the FAQ? |
| 19:16.12 | blitzrage | ctooley: ummmmm, yah, I write docs.... I know what I'm doing :) |
| 19:16.16 | ctooley | blitzrage, if you had read the FAQ, the answer to your question si plainly there. |
| 19:16.28 | ctooley | it's "upgrade to the latest version of HEAD" |
| 19:16.29 | ctooley | :) |
| 19:16.30 | blitzrage | ctooley: what FAQ are you talking about? |
| 19:16.33 | blitzrage | oh :) |
| 19:16.55 | ctooley | that's the number one answer to all questions relating to Asterisk |
| 19:17.32 | wwalker_ | ctooley You think? I said I would originate the call, and HOW do I fire an AGI against the call channel that was just originated, via the Manager? |
| 19:18.20 | *** join/#asterisk vandenk (~root@mn-69-69-105-79.dyn.sprint-hsd.net) |
| 19:18.22 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 19:18.50 | blitzrage | wwalker_: why don't you just use a call file? |
| 19:19.33 | blitzrage | ctooley: funny enough, upgrading CVS head worked :D |
| 19:19.56 | ctooley | blitzrage, there you go, the FAQ works again |
| 19:20.05 | blitzrage | lol |
| 19:20.52 | blitzrage | no idea why it would have failed on a system that had been running find for months though |
| 19:22.16 | *** join/#asterisk musimi (jens@sip-proxy.gratissip.dk) |
| 19:22.50 | bjohnson_ | anyone have an example of using logrotate with asterisk? My version is acting funny |
| 19:24.09 | tla | bjohnson_: I think you can just rotate the files using logrotate, asterisk will open/close files properly anyway |
| 19:24.36 | *** join/#asterisk wasim (~wasim@203.81.200.8) |
| 19:25.13 | bjohnson_ | not for me. when logrotate runs, it moves/renames the files as expected but * starts writing to messages.1 instead of messages .. then next logrotate run, * writes to messages.2, etc |
| 19:25.48 | ctooley | bjohnson_, you need to use cron to do a "asterisk -rx 'logger rotate'" instead of using logrotate |
| 19:26.31 | ctooley | Asterisk has to know that the log file has been changed otherwise it continutes to write to the same filehandle |
| 19:26.42 | tla | bjohnson_: I only do this for the cdr logfile, so you are probably right |
| 19:26.49 | ctooley | even though you've changed the name of the file you haven't changed it's inode (which the filehandle points to) and so it keeps writing to the same place |
| 19:27.24 | ctooley | the logger app in asterisk has a rotate function. |
| 19:27.51 | blitzrage | I love sheperd's quit message ;) |
| 19:27.53 | ctooley | Otherwise as part of logrotate you need to call "asterisk -rx 'logger reload'" which should close and reopen the file handle |
| 19:28.01 | bjohnson_ | ok .. but logrotate still used for cdr csv files? |
| 19:28.08 | *** join/#asterisk mikes2277 (~mike@12-221-249-232.client.insightBB.com) |
| 19:28.10 | *** join/#asterisk ennuyeux72 (~ennuyeux7@62.53.79.208) |
| 19:28.21 | *** join/#asterisk skel_home (~andrew@ip68-230-51-68.ph.ph.cox.net) |
| 19:28.29 | bjohnson_ | ctooley: you can get logrotate to do that? |
| 19:28.55 | tla | bjohnson_: yes, check postrotate command |
| 19:28.57 | ctooley | bjohnson_, sure, look at the way it handles apache (it has to HUP apache for the same reason) |
| 19:29.24 | skel_home | does anyone know of any asterisk hosting? I want to have voip conferencing available to my Linux users group memebers that are disabled and can't attend meetings |
| 19:29.42 | musimi | any have time to help me with international callerid? :-) (ie. someone with foreign number could call into my number so i can verify that it shows up correct? :-) (danish pstn number) |
| 19:29.45 | wwalker_ | People with weed AND guns make holes in people while giggling over Cheetos |
| 19:29.48 | skel_home | I was going to make an asterisk server myself.. but our group can't afford to colo it yet. |
| 19:30.15 | bjohnson_ | ctooley: are there logrotate files available somewhere? I couldn't google any |
| 19:30.38 | bjohnson_ | skel_home: FWD? |
| 19:30.40 | musimi | skel_home: how about using fwd's conference server? or sipphones? |
| 19:30.54 | riksta | voipuser down? |
| 19:30.58 | bjohnson_ | or a few other voip providers |
| 19:30.59 | ctooley | That HUP in apache has gotten me more than once, make changes to the config and not restart because I'm waiting on something else, and logrotate goes and and reloads the config for me anyway |
| 19:31.01 | skel_home | what is fwd? |
| 19:31.20 | ctooley | bjohnson_, dunno |
| 19:31.21 | bjohnson_ | skel_home: free world uhhh dialup? |
| 19:31.35 | skel_home | bjohnson_: cool thanks =] I'll google that :) |
| 19:31.39 | mikes2277 | I was looking at the realtime code and it seems to me like its very inefficient... I hope I'm just interpreting it wrong but it looks like it loads the full sipfreinds, extentions, etc. tables for each query instead of using the SQL server to do the query... can anyone confirm this? |
| 19:33.08 | skel_home | bjohnson_: now that allows the remote users to call in.. but I don't see how to setup a bridge call.. will I need something seperate for that? |
| 19:33.34 | Moc | Anyone use Hylafax in here ? |
| 19:34.00 | ctooley | mikes2277, that might be a #asterisk-dev question |
| 19:34.28 | mikes2277 | no one in there is reponding :( |
| 19:34.28 | |Vulture| | Anyone know who is favored in the super bowl? |
| 19:34.33 | skel_home | bjohnson_: nm I found a conference calls section |
| 19:34.37 | Qwell | |Vulture|: The Vegas bookies |
| 19:34.42 | Moc | mikes2277, using store procedure aint a very fun pratice |
| 19:35.01 | |Vulture| | Qwell: well I just made a bet with someone and I have no clue what my chances are |
| 19:35.05 | |Vulture| | I bet on the Pats |
| 19:35.10 | mikes2277 | i dont understand what you mean |
| 19:35.31 | Moc | well it different for every DB, also alot of DB dont support them |
| 19:35.44 | Moc | and working with store procedure will just open more problems |
| 19:36.07 | mikes2277 | oh, but they all support SELECT statements just fine and realtime only seems to work with odbc anyway |
| 19:36.28 | Moc | mikes2277, yes, the idea is to support all the Database type |
| 19:36.34 | mikes2277 | so why not SELECT for name = X or IP = X instead of loading ALL of the list into memory |
| 19:36.48 | Moc | well I didnt look at that part of code thought |
| 19:36.50 | mikes2277 | ODBC does that for you, no sense in having 2 abstract layers |
| 19:37.01 | Moc | my guess is not to make multiple SQL call, maybe not |
| 19:37.38 | Moc | I personally dont like the basic behind realtime anyway |
| 19:37.50 | file[laptop] | Alert: Parental units detected. |
| 19:37.57 | file[laptop] | Approaching. |
| 19:37.59 | Moc | it was a quickfix to have something 'realtime' between multiple system. But not the good longterm solution |
| 19:38.20 | ennuyeux72 | Moc: i get probems with realtime and odbc connecting to mysql db |
| 19:38.41 | markit | someone has more info about the phones in http://www.iaxtalk.com/ ? do they have 10 or 100 mb/s lan ports? |
| 19:38.50 | ennuyeux72 | Moc: from time to time asterisk just give sql execute errors and the only soln is to restart asterisk |
| 19:39.04 | mikes2277 | what I cant figure out is why they didnt use the ast_data patch since it does work correctly |
| 19:39.18 | Moc | ennuyeux72, I try to stay away from realtime. I ratter think of a distributed * Network config style |
| 19:39.39 | Moc | with master and slaved. |
| 19:39.53 | ennuyeux72 | Moc: thing is we have 10000 iax clients registered to one of our boxes |
| 19:40.08 | Moc | ennuyeux72, use text config |
| 19:40.30 | ennuyeux72 | Moc: reload took 2 mins and killed the box while reloading |
| 19:40.39 | Moc | lol, your having issues hehe |
| 19:40.47 | ennuyeux72 | Moc: also call quality was awful |
| 19:40.57 | ennuyeux72 | Moc: I know thats why we used realtime |
| 19:41.15 | ennuyeux72 | Moc: sorted the quality/deadlock on reload txt file issue |
| 19:41.18 | bjohnson_ | so I shouldn't touch the files created by asterisk logger at all? just use asterisk -rx 'logger rotate'? |
| 19:41.33 | ennuyeux72 | Moc: but introduced realtimes own special problems |
| 19:41.36 | mikes2277 | ast_data scales indefitily because it actually queries the DB live for just the info it needs.... |
| 19:41.40 | skel_home | how can fwd afford to do that? |
| 19:41.58 | mikes2277 | works for us pretty well, I have over 350,000 routes in it and its doesnt slow down at all |
| 19:42.32 | ennuyeux72 | mikes2277: ur using realtime rigth? |
| 19:42.46 | Chuji | Moc : You checked out the new soundpoint 4000 yet? |
| 19:42.51 | Moc | ennuyeux72, I see, I dlike to get some dev to agree on a real manager interface. That you can issue add/mod/del of anything, that will be distributed between multiple box |
| 19:43.12 | mikes2277 | no, ast_data cause realtime istn working |
| 19:43.20 | Moc | Chuji, well the specs look GREAT. They had the bright idea to use the same hardware as the IP 300,500,600 |
| 19:43.27 | Moc | so it same firmware, alot less work to manage |
| 19:43.38 | Moc | but the price, I mean I can't pay that for fun... hehe |
| 19:43.52 | Moc | But now Im stuck with a SoundStation IP 3000 :( |
| 19:44.01 | Chuji | Moc : Yeah, Waiting to see if they get on Ebay any time soon |
| 19:44.05 | ennuyeux72 | mikes2277: ok i'll check ast_data out |
| 19:44.14 | Moc | Chuji, I dont think so ;) |
| 19:44.19 | mikes2277 | http://svn.asteriskdocs.org/res_data/ |
| 19:44.28 | Moc | ok project for tonight, build a * Fax Server |
| 19:44.30 | mikes2277 | res_data ast_data same thing |
| 19:44.49 | Chuji | Moc: rxfax or Hylafax? |
| 19:45.02 | Moc | well hylafax seem very limited |
| 19:45.13 | ennuyeux72 | mikes2277: do you still need to do reloads using res_data/ast_data? |
| 19:45.28 | Moc | It was based on very basic need, and had being hacked up to do more stuff, but it still limited |
| 19:45.34 | mikes2277 | nope, ast_data is 100% live |
| 19:45.39 | Moc | Chuji, you know RightFax ? |
| 19:45.55 | mikes2277 | see http://bugs.digium.com/bug_view_page.php?bug_id=0002980 for my MWI patch for it |
| 19:45.58 | Chuji | Moc : Yeah, we have RightFax 8 at our office |
| 19:46.01 | ennuyeux72 | mikes2277: so there's nothing realtime does that ast_data doesn't do |
| 19:46.03 | Chuji | Moc : 12 Channels |
| 19:46.14 | Moc | Chuji, my idea is to build a rightfax style server + win client |
| 19:46.24 | blitzrage | anyone know what voltage the TDM400P card uses from the Molex connector? |
| 19:46.25 | mikes2277 | basically except ast_data is WAY more efficient and faster |
| 19:46.31 | Moc | with * and spandsp as backend for a start |
| 19:46.33 | Chuji | Moc : Nice, I'm interested. I don't have much of a backup for Rightfax |
| 19:46.39 | *** part/#asterisk skel_home (~andrew@ip68-230-51-68.ph.ph.cox.net) |
| 19:46.46 | Chuji | Moc : The damn Brooktrout hardware is soooo expensice |
| 19:46.48 | Chuji | expensive |
| 19:46.54 | mikes2277 | if you have 10,000 sipfriends entries asterisk with realtime will poop on itself |
| 19:47.00 | Moc | Chuji, yes, it why SpanDSP make a very interesting solution ;) |
| 19:47.13 | Moc | you could sell a Fax server for very cheap.. |
| 19:47.24 | Moc | I want to include billing code also in the software |
| 19:47.25 | mikes2277 | plus Asterisk will get progressive slower without my MWI patch (which works for non realtime stuff too) |
| 19:47.25 | markit | Moc: write a web-based inerface instead of win client, so other users (linux) can benefit |
| 19:47.30 | djMax | pointer for "unable to create channel of type 'IAX2' (cause 3)? google yields nothing. |
| 19:47.38 | Moc | markit, web interface suck |
| 19:47.45 | Chuji | Moc : Yeah, but I don't think you are going to get the reliability that Brooktrout boards have |
| 19:47.57 | Moc | it what Ive learn. For alot of thing, web is fine, but there is some stuff that can't be web based |
| 19:48.04 | ennuyeux72 | mikes2277:pooping is about right |
| 19:48.06 | DaTrueLion | yo |
| 19:48.10 | markit | Moc: I know, but windows sucks also... |
| 19:48.44 | Moc | markit, yep, but buisness use it, and alot of end user use it. It will be done in C# so people will be welcome to port it using mono |
| 19:48.59 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 19:49.21 | Moc | Chuji, agreed, but it could be modified to use it. I mean everything from client to server storage will be in postscript format |
| 19:49.39 | Moc | except incoming fax, and generated outgoing fax result |
| 19:49.58 | Moc | so will have server based frontpage creator ... |
| 19:50.30 | Moc | my basic idea rightnow, is build the Database structure, and build basic windows GUI to capture printing of document |
| 19:50.41 | *** join/#asterisk rfc1918 (~Gentoo@pool-151-205-67-15.char.east.verizon.net) |
| 19:50.46 | Moc | then build the server in C with db access, and make them communicate with each other |
| 19:51.15 | Chuji | Moc : If I were much of a coder, I'd help |
| 19:51.28 | *** join/#asterisk kaitseb (~sadie@jutrzenka.firma.o2.pl) |
| 19:52.02 | dan2 | hmm |
| 19:52.06 | Moc | Chuji, I need help with idea after ;) I feature I think would be great, is when you print a word doc, it ask for the # name blah... but have a hold for more document flag |
| 19:52.06 | dan2 | I have some really bad echo problems |
| 19:52.20 | wwalker_ | blitzrage Thank you!!! Call file is exactly what I needed! |
| 19:52.25 | Moc | then if you print from excel, it will prompt if you want to attach this print to a previous prepared fax |
| 19:52.41 | Moc | so you can easily put multiple page together from different software/document |
| 19:52.43 | blitzrage | wwalker_: glad I could help! |
| 19:52.55 | Moc | I dont think rightfax allow you to do it easily |
| 19:53.09 | Moc | you can attach document from rightfax util, but it failed most of the time |
| 19:53.29 | Moc | also I hate that the billing info and routing code isnt verifyed rightaway on the client side |
| 19:53.32 | *** join/#asterisk shidan (david_sarr@CPE000625dbadc2-CM014280007905.cpe.net.cable.rogers.com) |
| 19:56.19 | DaTrueLion | anyone know if Hyperthreading: 2 logical CPUs from dmesg |
| 19:56.32 | DaTrueLion | i mean if dual is enabled in freebsd 5.3 ? by default |
| 19:56.33 | DaTrueLion | smp |
| 19:57.03 | danfrey | is anyone familiar with tundo hardware? |
| 19:58.36 | lohelle | ne1 have an example on how to run a bash script from a webpage? |
| 19:59.06 | Moc | lohelle, you need to have your bash script in the cgi-bin folder |
| 19:59.29 | Moc | and you just call that file in the cgi-bin and it should be executed |
| 19:59.41 | Strom_TM | lohelle, i cn tel u bt u nd 2 tk n prpr nglsh k? |
| 20:00.01 | DaTrueLion | so no one know if dual cpu SMP enabled by default on Freebd 5.3 ? |
| 20:00.22 | Moc | DaTrueLion, it should be, Ive stop using freebsd at v4. something |
| 20:00.27 | DaTrueLion | k |
| 20:00.31 | DaTrueLion | tough 5.3 had a problem |
| 20:00.39 | DaTrueLion | but diffrenet pages say sdiffernet things on site |
| 20:00.52 | Moc | DaTrueLion, I never liked my freebsd experience |
| 20:00.57 | DaTrueLion | and has far has apache and php any speiclal building ? |
| 20:01.40 | Moc | DaTrueLion, dont think so, read the readme of each |
| 20:02.00 | Moc | search for freebsd or just bsd to see special action needed by then |
| 20:02.17 | bjohnson_ | hmm .. my sip through double nat project now gets sound from the * server .. but no sound from sip device (spa 2000) gets through to * |
| 20:02.28 | DaTrueLion | Intel(R) Xeon(TM) CPU 2.80GHz (2800.11-MHz 686-class CPU) |
| 20:02.29 | DaTrueLion | ;) |
| 20:02.32 | DaTrueLion | 2 cpu |
| 20:03.32 | Moc | i got a dell dual 2.8 too nice machine |
| 20:03.51 | DaTrueLion | yeah im reading all the apache tuning performance options right now |
| 20:03.55 | DaTrueLion | need to tune this box |
| 20:04.10 | Moc | my time cost more than buying more box ;) |
| 20:08.32 | Nukemizer | where would I find the text file to edit, that gets emailed to a user when a voice mail is sent to email? |
| 20:09.48 | file | maybe, voicemail.conf? |
| 20:10.30 | Nukemizer | did not look like it |
| 20:10.37 | DaTrueLion | so apache 2 or 1.3 is the best ? |
| 20:10.44 | kaitseb | 1.3 |
| 20:11.00 | DaTrueLion | shy do they keep 2 of them ? any reason |
| 20:11.01 | DaTrueLion | ? |
| 20:11.36 | kaitseb | DaTrueLion: depends what are you doing with it, no need to go 2.0 if you are running php |
| 20:11.44 | DaTrueLion | k |
| 20:11.49 | DaTrueLion | why go 2 then |
| 20:13.03 | file | Nukemizer: yes, it's there... it's a single line... read and you shall find it |
| 20:13.03 | DaTrueLion | o last question |
| 20:13.05 | DaTrueLion | php 4 or 5 |
| 20:13.05 | DaTrueLion | lol |
| 20:13.27 | Nukemizer | no, i mean the actual text that gets sent |
| 20:14.00 | *** join/#asterisk PTG1234 (~sdf@ip68-106-19-249.ph.ph.cox.net) |
| 20:14.05 | Nukemizer | when an email gets sent , I am wnating to modify the body of the message |
| 20:14.48 | file | Nukemizer: I'm telling you it's there |
| 20:15.07 | file | look at emailsubject and emailbody you twat :p |
| 20:18.42 | *** join/#asterisk mindCrime (~mindCrime@math00249.dhcp.unc.edu) |
| 20:18.50 | *** join/#asterisk dizzydiffi (~diseyi@adsl-70-240-241-42.dsl.hstntx.swbell.net) |
| 20:18.54 | dizzydiffi | hello |
| 20:19.04 | dizzydiffi | anyone here |
| 20:19.09 | *** part/#asterisk mikes2277 (~mike@12-221-249-232.client.insightBB.com) |
| 20:20.24 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 20:21.25 | DaTrueLion | hey anyone got RsaRef i need that |
| 20:22.15 | Nukemizer | File: - I dont mean to disbelieve you, but I don't see how the text that is sent to a user in email can be a one liner in voicemail.conf perhaps my voicemail.conf is missing this. Either way I still do get a four line message body in the email message that I want to modify. |
| 20:22.35 | file | Nukemizer: it's called... new lines! ala: \n |
| 20:22.47 | file | and tab is \t |
| 20:23.08 | file | so if you look in there... you will find, what you want... in your mind! |
| 20:23.13 | silik0n | unless its windows then newline is \r\n |
| 20:23.22 | dizzydiffi | hello peps trying to install OpenH323 |
| 20:23.37 | dizzydiffi | any one that can help would be great |
| 20:24.54 | danfrey | I can try to help, I just got done installing it |
| 20:25.03 | danfrey | what's the question |
| 20:25.22 | dizzydiffi | okay well iam trying to install thePWLBDIR |
| 20:25.35 | dizzydiffi | but i get a no makefile found |
| 20:25.51 | dizzydiffi | thanks buddy |
| 20:26.27 | danfrey | what os are your using? |
| 20:27.20 | dizzydiffi | linux |
| 20:28.28 | danfrey | sorry, I was mistaken, I just installed the quicknet drivers from openh323 not the openh323 itself |
| 20:28.45 | dizzydiffi | okay no probs |
| 20:28.57 | shido6 | hehe |
| 20:29.07 | shido6 | so do you have quicknet gear? or did u install the wrong stuff? :) |
| 20:29.13 | shido6 | got pwlib and openh323 ? |
| 20:29.18 | dizzydiffi | really |
| 20:29.30 | dizzydiffi | did u have any probs installing |
| 20:29.32 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 20:29.33 | dizzydiffi | on linux |
| 20:29.39 | shido6 | get those and then download the NuFone H.323 channel driver if you dont have it in /usr/src/asterisk/channels/h323 |
| 20:30.14 | shido6 | do you have any Digium (http://www.digium.com) gear? |
| 20:30.41 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 20:31.23 | dizzydiffi | no i dont |
| 20:31.42 | danfrey | I have a couple of 100p and a couple of phonejacks |
| 20:31.50 | shido6 | x100ps ? |
| 20:32.05 | danfrey | yes |
| 20:32.08 | ph3nix | lordy |
| 20:32.28 | ph3nix | is there a way to read incoming calls other than using AGI through extensions.conf? |
| 20:32.29 | shido6 | Good, need to support Digium |
| 20:32.44 | shido6 | read them how? |
| 20:32.45 | dizzydiffi | doing that right now |
| 20:32.54 | shido6 | be sure to only make opt |
| 20:33.02 | ph3nix | i want to use a client on desktops |
| 20:33.11 | ph3nix | to open a call manager program based on incoming calls |
| 20:33.22 | shido6 | and to add their locations in /etc/profile |
| 20:33.37 | shido6 | write a module :) |
| 20:34.02 | ph3nix | so, use AGI? :) |
| 20:34.14 | shido6 | screenpop.so |
| 20:34.24 | shido6 | an asterisk module |
| 20:34.43 | JerJer | screen pop won't make any sense as a module |
| 20:35.02 | JerJer | make the windows app connect to the asterisk manager port |
| 20:35.05 | shido6 | dont u have a screenpop app written already? |
| 20:35.19 | JerJer | i've written a couple different ones |
| 20:35.24 | danfrey | when defining channels with tdmoe can I use less than 24? |
| 20:35.30 | dizzydiffi | installing pwlib on linux |
| 20:35.30 | ph3nix | you have one i can use? |
| 20:35.43 | shido6 | would you release any or are you charging something affordable for users like ph3nix |
| 20:36.18 | JerJer | don't have the rights to release them |
| 20:36.25 | JerJer | i was paid to develop them for others |
| 20:37.14 | Moc | JerJer develop what ? |
| 20:37.22 | Moc | what are they looking for.. |
| 20:37.44 | *** join/#asterisk [alex] (~[alex]@201.137.118.84) |
| 20:37.53 | Moc | btw hi JerJer ;) |
| 20:37.59 | Moc | what up ? |
| 20:39.09 | *** join/#asterisk [alex] (~[alex]@201.137.118.84) |
| 20:39.13 | JerJer | lots of things |
| 20:39.23 | Moc | same here.. |
| 20:39.31 | JerJer | i have been solving everyone else's problems except NuFone's |
| 20:39.32 | ph3nix | humm.. is there uh, documentation on the call manager api |
| 20:39.40 | JerJer | vi manager.c |
| 20:39.45 | file[laptop] | po-tate-oh |
| 20:39.52 | Moc | ph3nix, check the wiki |
| 20:39.55 | Moc | JerJer : ( |
| 20:40.05 | Moc | I got too many idea rightnow :( |
| 20:40.18 | Moc | I got to focus, but im excited by all of them ;) hehe |
| 20:42.45 | Moc | I only wish I won the lotery and could spend all my time just on those idea.. |
| 20:48.41 | *** join/#asterisk nassy (~mark@24-193-228-118.nyc.rr.com) |
| 20:52.38 | Godsey | I paid $20 for nufone yesterday or day before :) |
| 20:52.56 | Godsey | sooner or later I'll order an SPA-2000 or something and be ready to play |
| 20:55.21 | Moc | Chuji, I found the name for my faxing solution ;) |
| 20:55.58 | *** join/#asterisk postel (~canonical@host217-42-82-130.range217-42.btcentralplus.com) |
| 20:58.58 | PoWeRKiLL | hi |
| 20:59.10 | PoWeRKiLL | why when I create prompt for asterisk the sound is very bad |
| 20:59.34 | PoWeRKiLL | I use sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql to convert the 44khz 16bit stereo wav file |
| 20:59.47 | Strom_TM | because gsm sucks |
| 20:59.56 | file[laptop] | Strom_TM!!!!!!!!! |
| 21:00.30 | Strom_TM | use a wav file; let the codec compress that down to gsm in realtime if it needs to |
| 21:00.52 | Strom_TM | file: clarity is more important than bandwidth for me. ergo, ulaw > * |
| 21:01.03 | file[laptop] | Strom Strom Storm! |
| 21:01.08 | file[laptop] | where have you been hiding? |
| 21:01.22 | Strom_TM | at work :) |
| 21:01.36 | file[laptop] | this 'work' is unhealthy for you! |
| 21:01.37 | bjohnson_ | IS there a way to config an external SIP device to handle double nat WITHOUT using the external ip in the config of either * OR the SIP device |
| 21:01.57 | *** part/#asterisk iMediax (lklk@00045a809589.click-network.com) |
| 21:02.07 | Strom_TM | file: true, but it pays the bills |
| 21:02.19 | file[laptop] | silly bills |
| 21:02.23 | Strom_TM | i know |
| 21:02.28 | Strom_TM | everything should be freeee |
| 21:02.32 | file[laptop] | indeed |
| 21:04.29 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 21:05.22 | Strom_TM | anyway, ive got one eye permanently affixed on the super bowl's router - it's going to be a baaaaaaad thing if that goes down |
| 21:06.20 | JerJer | bjohnson_: use IAX |
| 21:06.21 | file[laptop] | is it flashing madly?!? |
| 21:06.22 | *** join/#asterisk Inv_arp (junya@adsl-8-230-175.mia.bellsouth.net) |
| 21:06.40 | Inv_arp | bah xchat never autojoins this chan |
| 21:07.04 | file[laptop] | IAX is so nifty |
| 21:07.11 | Strom_TM | file, no, it's stable at the moment |
| 21:07.36 | Inv_arp | file[laptop]: love it cause it just works |
| 21:07.59 | file[laptop] | well it's easy to understand, easy to modify |
| 21:08.17 | Inv_arp | ok got paid gonna buy handy tone 486, what site should i go to |
| 21:09.04 | liquidno2 | heard decent things about voipsupply.com |
| 21:09.15 | JerJer | get a sipura 3k |
| 21:09.24 | JerJer | from voxilla |
| 21:09.31 | liquidno2 | not sure if they carry sipura though |
| 21:09.44 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlgrv.pa.sed6.net) |
| 21:10.08 | Inv_arp | sipura 3000 more than my bidget atm |
| 21:10.18 | JerJer | pretty small budget then |
| 21:10.22 | Inv_arp | and my budget |
| 21:10.29 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 21:10.48 | Inv_arp | heh 7.5 an hr part time |
| 21:10.59 | Inv_arp | you tell me |
| 21:11.41 | rfc1918 | has anyone setup a STUN server? |
| 21:11.51 | JerJer | stun is not necessary |
| 21:12.48 | rfc1918 | need it for Internet users to reach my local extensions |
| 21:13.05 | rfc1918 | Calls connect no audio |
| 21:13.25 | JerJer | no you don't |
| 21:13.34 | rfc1918 | explain |
| 21:13.35 | *** join/#asterisk blaisen1 (~blaisen1@tightcode.ofpower.net) |
| 21:13.40 | JerJer | don't NAT your asterisk box |
| 21:13.47 | JerJer | ie give your asterisk box a public ip |
| 21:14.03 | rfc1918 | Don't have that option |
| 21:14.06 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 21:14.08 | Mother_ | JerJer: it exposes your box a helluva lot more than with NAT |
| 21:14.10 | blaisen1 | hmm, i have a call incoming by IAX from a DID provider, I have asterisk to a dial(SIP/blah,40,r) but it doesn't give a ringback to the person coming through the iax link |
| 21:14.14 | JerJer | then you don't use SIP then |
| 21:14.24 | JerJer | um its called a firewall people |
| 21:14.28 | *** join/#asterisk ardor (~ardorgof@ip68-227-38-164.lv.lv.cox.net) |
| 21:14.31 | Mother_ | JerJer: yeahright :D |
| 21:14.33 | JerJer | iptables |
| 21:14.39 | rfc1918 | SIP is where the industry is going |
| 21:14.45 | shido6 | HAH! |
| 21:14.49 | Inv_arp | rfc1918: heh |
| 21:15.06 | Inv_arp | prob if they revise it |
| 21:15.11 | blaisen1 | SIP kinda sucks but if your customers' CPE is using it you don't have to use up all your bandwidth and add latency by "relaying" it to the customer |
| 21:15.17 | shido6 | really? sip? you think so? not with these nat problems |
| 21:15.26 | Mother_ | until the next bug/vuln/exploit is found, then you find the kiddies making calls on your PBX to Jamaica |
| 21:15.38 | rfc1918 | STUN will fix it |
| 21:15.45 | blaisen1 | I have grandstream handytone's working perfectly behind consumer broadband routers connecting to my * box with no problem whatsoever |
| 21:15.49 | JerJer | there wasn't an exploit found in the first place |
| 21:15.51 | blaisen1 | and no stun |
| 21:16.05 | rfc1918 | Is it using SIP? |
| 21:16.21 | blaisen1 | yes |
| 21:16.27 | rfc1918 | explain |
| 21:16.44 | Mother_ | JerJer: I get linux exploits on the vuln lists every day, I'm not saying *I* wouldn't run a PBX in the open, so-to-speak, it's just that I wouldn't leave one at a client's location just like that |
| 21:16.46 | danfrey | what is cdr_csv.so? |
| 21:16.56 | JerJer | Mother_: go away |
| 21:17.04 | Mother_ | why? |
| 21:17.08 | blaisen1 | i just have it set up nat=yes all that stuff, and the handytones seem to handle it fine |
| 21:17.11 | rfc1918 | I am using x-ten and th esniffer capture clearly show internal address used instead of NAT |
| 21:17.12 | file[laptop] | why do you build me up, build me up, buttercup baby |
| 21:17.16 | file[laptop] | just to let me down! |
| 21:17.28 | liquidno2 | danfrey: call detail records in csv module |
| 21:17.29 | rfc1918 | so how to do it without STUN? |
| 21:17.46 | danfrey | safe to disable in modules.conf? |
| 21:17.47 | Mother_ | JerJer: tell me you *never* got a call from a client who installed whatnot and he now finds his box taken over? |
| 21:17.47 | blaisen1 | rfc1918 i don't know... it just works |
| 21:17.48 | JerJer | get it out of your head, stun does nothing |
| 21:17.57 | file[laptop] | rfc1918: nat=yes would tell asterisk to not use the IP address in the SIP message, and use the IP that it was received fun |
| 21:18.01 | liquidno2 | danfrey: if you don't need cdr in csv |
| 21:18.01 | JerJer | nat=yes |
| 21:18.05 | file[laptop] | I use it for my stuff here at home and it works peachy |
| 21:18.06 | JerJer | problem solved |
| 21:18.17 | Inv_arp | rfc1918: think u have to put real ipaddress in sip.conf |
| 21:18.17 | blaisen1 | rfc1918: i have a handytone i take around and demo to people, just plug it into their broadband router it gets dhcp and away we go it sends and receives calls to/from my * box no sweat.. |
| 21:18.30 | file[laptop] | nat=yes makes it all happy |
| 21:18.50 | JerJer | but the serving asterisk box cannot be natted |
| 21:18.59 | JerJer | ie it has to have a public IP address |
| 21:19.06 | liquidno2 | how would I make all calls on a specifc channel not have to dial a 9? |
| 21:19.11 | file[laptop] | unless you're bloody insane |
| 21:19.23 | Inv_arp | i had to put externip=ipaddress for mine |
| 21:19.35 | rfc1918 | NAT=yes works, but the nat prevents the audio stream from working cause it is using private IP instead of public |
| 21:19.42 | JerJer | liquidno2: make an exten that doesn't match on a 9 |
| 21:19.48 | JerJer | rfc1918: no |
| 21:19.51 | file[laptop] | rfc1918: no no no |
| 21:19.56 | JerJer | rfc1918: your edge device is blocking udp |
| 21:20.37 | file[laptop] | that's why your RTP stream isn't working |
| 21:21.07 | JerJer | register thru the nat |
| 21:21.13 | JerJer | problem solved |
| 21:21.30 | file[laptop] | qualify helps keep the hole open too... I have to do that for a friend's D-Link POS |
| 21:21.46 | danfrey | what would use Raw signed linear audio? |
| 21:21.53 | *** part/#asterisk _LM_ (foobar@cleopatra.jogback.se) |
| 21:21.54 | file[laptop] | asterisk internally. |
| 21:22.19 | blaisen1 | i have qualify to yes as well |
| 21:22.44 | blaisen1 | so does anyone know why my incoming calls on iax aren't getting a ring tone when they dial an extension with dial(sip/blahblah,30,r) ? |
| 21:22.50 | JerJer | just lower the registration timeout then |
| 21:23.00 | danfrey | in modules, what is the relationship of codecs to formats? |
| 21:23.24 | PoWeRKiLL | Strom_TM what bitrate do you use for your wav file ? |
| 21:23.47 | Strom_TM | powerkill: 8000khz 16bit wav |
| 21:23.53 | Strom_TM | mono |
| 21:23.59 | rfc1918 | hmm |
| 21:24.33 | rfc1918 | phones are already registered throught the NAT - no issue there |
| 21:24.38 | wwalker_ | what's the best way to convert normal wav files into .gsm files for the sound directory? |
| 21:24.50 | Strom_TM | wwalker_, dont use gsm |
| 21:24.58 | Strom_TM | downsample to 8000khz 16bit wav |
| 21:25.05 | Strom_TM | it'll sound much better |
| 21:25.11 | danfrey | tmdoe uses no codec, is this correct? |
| 21:25.26 | JerJer | rfc1918: then your edge device is not stateful |
| 21:25.53 | rfc1918 | Cisco Pix |
| 21:26.02 | JerJer | i'm sorry |
| 21:26.04 | rfc1918 | one one side |
| 21:26.04 | *** join/#asterisk BigCanOfTuna (~chatzilla@dsl-macn-66-18-205-30-cgy.nucleus.com) |
| 21:26.08 | rfc1918 | on one side |
| 21:27.28 | rfc1918 | STUN, I think, will adjust the SDP message and replace it witht the NAT address instead of private |
| 21:27.56 | rfc1918 | so the audio can be correctly recieved |
| 21:28.03 | JerJer | that has no bearing |
| 21:28.09 | JerJer | nat=yes does not look at SDP |
| 21:28.10 | JerJer | at all |
| 21:28.11 | JerJer | what soever |
| 21:28.12 | JerJer | never |
| 21:28.12 | JerJer | does not |
| 21:28.15 | JerJer | clear? |
| 21:28.27 | rfc1918 | the x-Ten phone does |
| 21:28.29 | BigCanOfTuna | If I want to have an external application issue commands to Asterisk, should I be looking at AGI? Or is AGI one way ....Asterisk -> Ext App? |
| 21:28.58 | JerJer | do not NAT the asterisk box |
| 21:29.01 | JerJer | plain and simple |
| 21:29.38 | JerJer | BigCanOfTuna: use the Asterisk manager API |
| 21:29.52 | rfc1918 | asterisk only sets up the call, the clients carry the audio |
| 21:30.12 | rfc1918 | that part works |
| 21:30.26 | rfc1918 | clients conncet, but no audio |
| 21:30.29 | lohelle | Is 2 vs 6 lines the only difference between cisco 7940 and cisco 7960? |
| 21:30.38 | JerJer | then your edge device is blocking UDP |
| 21:30.39 | rfc1918 | extra lines |
| 21:30.40 | Strom_TM | lohelle, yes |
| 21:30.52 | lohelle | ok.. |
| 21:33.09 | JerJer | don't use SIP in NAT envrionments |
| 21:33.15 | JerJer | IAX or nothing |
| 21:33.16 | rfc1918 | JerJer: Ok, I will check hte Pix again |
| 21:33.55 | rfc1918 | What do you recommend instead of SIP? |
| 21:34.02 | rfc1918 | oh |
| 21:34.11 | JerJer | IAX |
| 21:34.12 | Strom_TM | yay for paying attention |
| 21:34.47 | rfc1918 | What softphone do you use? |
| 21:34.51 | markit | anyone here can help me to make "disconnect" feature work (features.conf)? seems not to work here |
| 21:35.20 | JerJer | i don't |
| 21:35.33 | rfc1918 | waht ATA? |
| 21:35.37 | rfc1918 | what ATA |
| 21:35.44 | JerJer | IAXy |
| 21:36.25 | Mother_ | you need a fairly nice pipe with those from what I've heard |
| 21:36.26 | *** join/#asterisk [hC] (~turnerd@8.10.2.4) |
| 21:37.11 | [hC] | Im trying to figure out the best way to handle a tech support queue in asterisk. The agents and queues seems a bit limited right now. I cant figure out a way to force all of the members to be logged in all the time |
| 21:37.24 | JerJer | dont' use agents |
| 21:37.36 | [hC] | What ive done, on our cisco 7960's, is create another line for 'support' - people are support1, support2, support3, etc.. |
| 21:37.37 | Mother_ | speaking of which anyone knows of an IC that has a built-in GSM or similar bitrate codec built-in? I'm working on a dsPIC but would rather offload this task for now |
| 21:37.43 | *** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net) |
| 21:37.49 | [hC] | That way they can tell the incoming call is for support |
| 21:38.18 | [hC] | I was going to just do a dial multiple-sip thing to each of them.. |
| 21:38.56 | [hC] | JerJer: how do you suggest handling a situation like this, where you have say, 10 phones that need to be rang, and also the potential for say, someone from home to be able to add themselves to the dialdown queue |
| 21:39.24 | JerJer | a queue |
| 21:39.38 | [hC] | so, a queue without agents then? |
| 21:39.59 | [hC] | Just all dynamic extensions in the queue? |
| 21:41.47 | *** join/#asterisk talkwebhosts (~admin@c-24-130-183-95.we.client2.attbi.com) |
| 21:42.11 | [hC] | even then, how can i ensure that people are always logged into the queue? I have 10 phones in the office that i always want to ring, but i dont necessarily want them to have to log in and out all the time |
| 21:43.01 | *** join/#asterisk YoussefAssad (~yassad@62.114.36.156) |
| 21:43.24 | JerJer | you are still thinking agents |
| 21:43.28 | JerJer | stop |
| 21:43.50 | [hC] | heh |
| 21:43.58 | [hC] | ok, so how should i be thinking about it? |
| 21:44.08 | JerJer | member => SIP/bob |
| 21:44.48 | [hC] | ok, so i define the queue, with a bunch of members. those members still need to log themselves into the queue, dont they? |
| 21:44.52 | YoussefAssad | Evening folks |
| 21:44.55 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
| 21:44.58 | JerJer | GRR |
| 21:45.01 | JerJer | THAT IS AGENTS |
| 21:45.01 | [hC] | or do they not, if they are SIP/ |
| 21:45.49 | JerJer | if member => SIP/bob is in your queues.conf SIP/bob is always going to be a part of that queue |
| 21:45.53 | JerJer | very simple |
| 21:45.59 | [hC] | Thats what i was trying to find out |
| 21:45.59 | *** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl) |
| 21:45.59 | [hC] | Thanks |
| 21:46.01 | [hC] | thats all i needed. |
| 21:46.01 | YoussefAssad | General question (apologies if it is too general): ho mature is openss7? |
| 21:46.06 | YoussefAssad | how* |
| 21:46.12 | JerJer | hoe |
| 21:46.24 | YoussefAssad | :) |
| 21:50.37 | blaisen1 | any suggestions on my lack of ringback to dialing a SIP extension from an incoming IAX call? |
| 21:51.17 | RoyK | ~lart blaisen1 |
| 21:51.38 | YoussefAssad | oh my; teh nasty |
| 21:51.43 | blaisen1 | royk: what do you mean? ~lart |
| 21:51.49 | RoyK | ~lart? |
| 21:51.50 | jbot | methinks lart is Luser Attitude Re-adjustment Tool |
| 21:51.57 | sjaak538 | Can any body check if www.forumvoip.vcom is down !!, for a few day's I can't visit there website |
| 21:52.14 | blaisen1 | i'm doing the dial(sip/blahblah||r) but asterisk is not generating a ringback sound... |
| 21:52.20 | RoyK | sjaak538: never heard of .vcom domains before :P |
| 21:52.30 | blaisen1 | it does when a call comes in from pstn on an fxo card, it does from phone to phone... |
| 21:52.42 | blaisen1 | but not from an incoming iax call |
| 21:52.42 | Mother_ | it's the new Venus TLD |
| 21:52.51 | sjaak538 | sorry www.forumvoip.com (typo) |
| 21:52.56 | *** join/#asterisk BigCanOfTuna (~chatzilla@dsl-macn-66-18-205-30-cgy.nucleus.com) |
| 21:52.58 | djin | sjaak, it works. |
| 21:53.00 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 21:53.04 | ardor | anyone knwo how to take a.wav and b.wav and make ab.wav on the command line? |
| 21:53.07 | sjaak538 | ?? |
| 21:53.15 | djin | sjaak538 = dutch? |
| 21:53.57 | sjaak538 | Yes dutch djin |
| 21:54.20 | djin | Ah, Sjaak, het werkt hier wel, maar een beetje langzaam ;) |
| 21:54.34 | ChatWeazl | hier ook niet |
| 21:54.37 | YoussefAssad | works there well, but a bit slow |
| 21:54.45 | sjaak538 | I can't also ping them on forumvoip.com and not on www. |
| 21:54.47 | YoussefAssad | Did I translate that right? |
| 21:55.09 | djin | Youssef is correct. |
| 21:55.11 | ChatWeazl | you did great YoussefAssad |
| 21:55.15 | sjaak538 | Thats correct translating |
| 21:55.22 | YoussefAssad | haha cool :) |
| 21:56.29 | sjaak538 | So from the Netherlands it isn't up !! |
| 21:56.37 | djin | yes, it is. |
| 21:56.44 | ChatWeazl | no it's not :D |
| 21:56.48 | djin | (xs4all) |
| 21:56.54 | ChatWeazl | cambrium ... |
| 21:57.08 | sjaak538 | KPN |
| 21:57.12 | djin | Cambrium, funny . . . |
| 21:57.36 | ChatWeazl | why funny? |
| 21:57.46 | djin | Because Cambrium isn't a provider. Either TweakDSL, TruDSL or MyADSL. |
| 21:57.58 | Bentley | Hi all, Does anyone here use 'Hint' in extensions.conf? I'd like to know how to implement 'Hint' if I am dialing multiple extensions (ie: Dial(sip/200&sip/202)) |
| 21:58.31 | ChatWeazl | that's true, but all route throught the same hardware... |
| 21:58.51 | djin | Cambrium = djin's Business Partner . . . That's why I thought it's funny to see their name. |
| 21:58.59 | ChatWeazl | hehe |
| 21:59.06 | djin | You're right, same hardware/routes. |
| 22:00.00 | *** join/#asterisk Luke-Jr (~luke-jr@207.192.221.172) |
| 22:00.13 | Nukemizer | Would I want to change the text in app_voicemail.c |
| 22:00.13 | Nukemizer | <PROTECTED> |
| 22:00.30 | Luke-Jr | Can anyone recommend a POTS connection provider (like Vonage) that would work with Asterisk? |
| 22:00.48 | djin | Nuke, why not change voicemail.conf? |
| 22:01.31 | Nukemizer | is that where you put the text ? because there is no such "text" message in the file |
| 22:02.19 | djin | what do you want to change, the contents of the mail carrying the voicemail? |
| 22:02.36 | Nukemizer | yes |
| 22:02.54 | djin | Then check /etc/asterisk/voicemail.conf. It's all there. |
| 22:03.47 | Nukemizer | I have been told that. let me patbin my file .. It must be that I do not have it in mine which is why i am confused |
| 22:04.33 | djin | I'll pastebin the default as well. |
| 22:04.36 | *** join/#asterisk jterrero (~some@ool-43576e0d.dyn.optonline.net) |
| 22:05.14 | djin | Nuke, this is from de default file: http://pastebin.ca/5360 |
| 22:06.26 | Nukemizer | Wow,, see i knew something was wrong.. none of that is in my file |
| 22:06.40 | Nukemizer | that explains it |
| 22:06.49 | Nukemizer | Thank you so much :) |
| 22:07.00 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 22:07.05 | djin | Shall I paste the full .conf? |
| 22:07.30 | djin | There is a setting for the 'from'-address as well. |
| 22:08.12 | Nukemizer | man that would be great. Thank you so much Not sure whymine didi not have the text in it |
| 22:08.36 | djin | did you do a 'make samples' after compiling? |
| 22:08.57 | Nukemizer | yes |
| 22:10.51 | *** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net) |
| 22:11.01 | djin | strange, well here it is: http://pastebin.ca/5363 |
| 22:11.58 | Nukemizer | Thanks djin, I appricate your help |
| 22:12.16 | jterrero | someone wanna help me out? I have an account with a IAX provider who provides me with my DID, i setup my account on my asterisk box for home. now, when i make or receive a call i can hear the person clearly on my end, no lag, distortion or breaking up, but on the other hand they cant hear me too well, alot of breakup (like if i was using a cell phone) |
| 22:12.18 | mikegrb | Qwell: I have a nikon d70 |
| 22:12.38 | Qwell | eh? |
| 22:12.45 | Qwell | oh, right |
| 22:15.28 | markit | mikegrb: my compliments |
| 22:15.43 | *** join/#asterisk imagmo (~imagmo@ip-64-250-232-158.lasvegas.net) |
| 22:15.46 | mikegrb | yes, I'm very happy with it |
| 22:16.01 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 22:16.12 | mikegrb | one of the biggest things that sold me on it vs the canon rebel was 0.2 sec from power off to on |
| 22:16.22 | mikegrb | and < .5 sec shutter lag |
| 22:16.26 | markit | I've a minolta A2, but true dslr sensor is another planet |
| 22:16.37 | mikegrb | yes |
| 22:16.55 | markit | I'm tempted by canon 20d, but too expensive |
| 22:17.07 | markit | and dslr are too heavy also |
| 22:17.11 | rfc1918 | JerJer: Sniffer shows private addresses being used for audio between clients |
| 22:17.13 | Luke-Jr | I'm seeing stuff about 'pri' support a lot, but I can't seem to figure out what PRI is... :/ |
| 22:17.17 | mikegrb | I want a nikon d2h |
| 22:17.30 | Qwell | Luke-Jr: primary rate interface - a T1 basically |
| 22:17.33 | mikegrb | Luke-Jr: like a voice t1 |
| 22:18.55 | Luke-Jr | ah... something a telco would use |
| 22:19.36 | mikegrb | well an office too |
| 22:20.06 | mikegrb | depending on the area and the telco there, once you get to 5-10 phone lines it is cheaper to have PRI then individual business lines |
| 22:20.27 | *** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com) |
| 22:20.43 | Qwell | You can get a portion of a T1, can't you? |
| 22:20.55 | heath__ | anyone know about EAGI? does the audio data come in on stdin or something? |
| 22:21.20 | mikegrb | Qwell: yes but sometimes they are expensive too |
| 22:21.26 | Luke-Jr | mikegrb: I'm just setting up our apartment |
| 22:21.31 | silik0n | PRIs in TN usually work out cheaper around the 10 line mark due to the fact they dont hit you with crap like universal service fund and E911 etc for EVERY channel... they only get you for it on 4 or 5 channels |
| 22:21.36 | Qwell | Luke-Jr: You probably don't need a pri then :p |
| 22:21.50 | mikegrb | heh |
| 22:22.10 | mikegrb | I've heard people say it is just barely less at the 5 line point in thier area |
| 22:22.11 | Qwell | mikegrb: You know anything about line voltages? |
| 22:22.22 | mikegrb | Qwell: some, shoot |
| 22:22.27 | kFuQ | grrrrrrrrr....... does anyone have any clue whatsoever on getting call waiting to work on a x100p ? --- *0 doesn't do anything, Flash command doesn't seem to work properly either... |
| 22:22.31 | mikegrb | just depends on telco |
| 22:22.34 | Qwell | I'm thinking about making a relay, but wouldn't have any idea what voltages I would need for anything |
| 22:22.59 | mikegrb | kFuQ: try flash and then *0 |
| 22:23.09 | *** part/#asterisk YoussefAssad (~yassad@62.114.36.156) |
| 22:23.24 | mikegrb | Qwell: relay for switching phone between asterisk box and not? |
| 22:23.34 | Qwell | yeah, this basically http://www.voip-info.org/tiki-index.php?page=Asterisk%20failover |
| 22:24.02 | Qwell | tzanger was talking about it a bit last night, and I got some more info from a friend, but we didn't discuss voltages much, just how to set it up |
| 22:24.11 | Qwell | (night before last? whenever it was) |
| 22:24.17 | kFuQ | mikegrb: nothin |
| 22:24.25 | Qwell | ..5 nights ago according to my logs. heh |
| 22:25.21 | mikegrb | heh |
| 22:25.33 | mikegrb | kFuQ: sorry, was just a suggestion, I don't have a x100p |
| 22:25.41 | Qwell | basically, I don't want to blow up my relay or my other phones. ;) |
| 22:26.57 | kFuQ | mikegrb: have this loud noise in my ears.. "THE DAM CALL WAITING DOESN'T WORK!@#!" (wife) |
| 22:27.20 | mikegrb | well the 5v or 12v suggestion relates to the coil that switches back and forth |
| 22:27.35 | Qwell | yeah, I got that part, and I can manage that with a parallel port |
| 22:27.46 | Juggie | sigh |
| 22:27.55 | Juggie | i broke my collar bone today |
| 22:28.00 | Qwell | Juggie: ouch |
| 22:28.08 | mikegrb | well, you can power it fro one of the drive connectors |
| 22:28.09 | Juggie | some kid cut me off skiing |
| 22:28.18 | mikegrb | would be a bit easier to get it from there |
| 22:28.25 | Juggie | we hooked skis and i went flying |
| 22:28.28 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 22:28.30 | mikegrb | telephone ring voltage is about 90 vac |
| 22:28.32 | mikegrb | Juggie: :/ |
| 22:28.35 | Qwell | mikegrb: I'm gonna go ahead and use an npn transistor, and control it with the parallel |
| 22:28.41 | mikegrb | Qwell: oh |
| 22:28.47 | Qwell | So, I would probably need a good 125vac? |
| 22:29.06 | mikegrb | yes, any relay that can do linve voltage should be fine |
| 22:29.31 | mikegrb | line as in electrical outlet |
| 22:29.32 | Qwell | ring voltage is like the "most" you'll get, right? |
| 22:29.36 | mikegrb | right |
| 22:29.36 | firestrm | mikegrb, ever been leaning over a punchdown rack, and accidently contact wires with bare(usually sweating) skin when a call comes in.. YOUCH!! |
| 22:29.55 | Qwell | excellent, lets see if I can convince my wife to stop at radio shack, heh |
| 22:29.56 | mikegrb | firestrm: yes indeed ;) |
| 22:30.09 | firestrm | mikegrb, gets your attention :) |
| 22:30.17 | mikegrb | my parents house has beautiful telephone and network wiring |
| 22:30.27 | mikegrb | with patch panels and everything |
| 22:30.31 | Qwell | nice |
| 22:30.35 | Qwell | excessive, but nice :p |
| 22:30.40 | mikegrb | when I was hooking up a patch panel for telco got a call |
| 22:30.50 | mikegrb | Qwell: levitron makes some nice panels for residential use |
| 22:31.07 | Qwell | I always use levitron for my wall jack needs |
| 22:31.11 | mikegrb | they are like $12 for a panel with 24 positions |
| 22:31.23 | mikegrb | then they use the snap in keystone jacks |
| 22:31.32 | mikegrb | great deal |
| 22:31.41 | firestrm | mikegrb, i once had a pare come loose, and sneak up into my armpit while i was leaning behind a panel, and wouldnt you know it.. a call came through on that particular pair.. |
| 22:31.51 | mikegrb | and I used some $3 mointing kits they have that allow you to mounth them to wall |
| 22:31.58 | mikegrb | firestrm: :/ |
| 22:32.12 | Qwell | mikegrb: Those snapin plates are great |
| 22:32.30 | mikegrb | Qwell: yes, parents have them all through thier house |
| 22:32.32 | firestrm | the guy i was with told me he could almost see the sparks shooting out my eyeballs.. |
| 22:33.13 | Qwell | hmm, a 125vac dpdt relay is fairly large, isn't it? Need to find a place to hide it |
| 22:33.19 | mikegrb | last time we were home they wanted more ethernet jacks in home office so I taught my wife to do it |
| 22:33.45 | mikegrb | Qwell: about 2, 2.5 inches wide, inch deep and about 3 inch tall |
| 22:33.49 | firestrm | of course the first jolt caused me to get stuck.. so i had to endure 2 more jolts before i could get free, swearing like a sailor |
| 22:33.56 | Qwell | ouch, yeah, pretty big |
| 22:34.04 | mikegrb | well you can get some reed relays that are very small and can handle 125 vac |
| 22:34.07 | mikegrb | but spst |
| 22:34.18 | mikegrb | you don't nevessarily need a hell of a lot of current |
| 22:34.45 | firestrm | mikegrb, why not use a bridge, resistor and a common 12vdc reed relay? |
| 22:35.16 | mikegrb | I dunno |
| 22:35.19 | firestrm | mikegrb, voltage doesnt matter.. current does |
| 22:35.34 | mikegrb | ja |
| 22:35.35 | firestrm | mikegrb, so use a resistor to limit current |
| 22:35.59 | firestrm | bridge makes it dc.. much easier to work with.. |
| 22:36.16 | mikegrb | but then the phone won't ring |
| 22:36.28 | Qwell | ringer? we don't need no stinking ringer :p |
| 22:36.40 | mikegrb | using 12vdc coil |
| 22:36.47 | mikegrb | ac is for the relay contacts |
| 22:37.04 | firestrm | mikegrb, why not.. as long as curent limit so as to not overload the pair.. your good |
| 22:37.43 | firestrm | or better yet, use a 1:1 transformer... |
| 22:37.45 | Qwell | I'll probably stick with the easy way |
| 22:38.00 | *** join/#asterisk expressfone1 (expressfon@80.27.5.114) |
| 22:38.05 | Qwell | I'm probably gonna be lazy, and just use a dpdt switch instead |
| 22:38.07 | expressfone1 | Hi all |
| 22:38.10 | implicit | expressfone1: hi |
| 22:38.23 | firestrm | expressfone1, wish i was.. |
| 22:38.28 | Qwell | we'll see when I get to radio shack, heh |
| 22:38.33 | Qwell | bbl, thanks for the tips |
| 22:38.47 | firestrm | Qwell, radioscrap is your friend :) |
| 22:39.08 | expressfone1 | any one know where to find international dialing codes with ammount of digit per prefix?? |
| 22:39.09 | implicit | firestrm: evil friend |
| 22:39.26 | implicit | expressfone1: yes |
| 22:39.36 | implicit | expressfone1: there is a small algorithm you can use |
| 22:39.42 | firestrm | implicit, no in canada, princess auto.. is the hardware hackers evil firend |
| 22:39.44 | implicit | that will tell you how many digits of it are the country code |
| 22:40.13 | Slainte | I would like to see that alogrithm myself |
| 22:40.31 | implicit | ok |
| 22:40.33 | implicit | ill show you guys here |
| 22:40.35 | expressfone1 | implicit??? |
| 22:40.41 | implicit | if it starts with 1 or 7 its 1 digit long |
| 22:40.57 | implicit | if it starts with 20 or 27 its 3 digits long |
| 22:41.09 | implicit | 35 37 or 38 its 3 digits long |
| 22:41.14 | implicit | 42 is 3 digits long |
| 22:41.21 | RoyK | Is mark Italian? |
| 22:41.22 | implicit | 50 and 59 are 3 digits long |
| 22:41.41 | implicit | 67 68 69 3 digits long |
| 22:41.57 | implicit | 80 83 85 87 88 89 3 digits long |
| 22:41.59 | RoyK | 6768693 digits is quite a lot |
| 22:42.07 | implicit | 96 97 99 3 digits long |
| 22:42.33 | expressfone1 | i need the info for +53 8( |
| 22:43.07 | Slainte | thats fine for country code, but what about the rest of the number |
| 22:43.08 | implicit | otheri think all other ones are 2 digits long |
| 22:43.20 | implicit | so it shows you how much of it is country code and you can splice that off |
| 22:43.29 | implicit | check the algo but i think its right |
| 22:43.37 | expressfone1 | +5783XXXXX |
| 22:43.44 | expressfone1 | that one |
| 22:43.58 | expressfone1 | sorry |
| 22:44.03 | implicit | thats 2 digits just like i said |
| 22:44.08 | expressfone1 | +53783XXXXX |
| 22:44.09 | implicit | since it doesnt fall into any of the above |
| 22:44.15 | implicit | thats also 2 digits |
| 22:44.26 | implicit | first is colombia second is cuba |
| 22:44.42 | expressfone1 | i need the info for cuba |
| 22:45.12 | implicit | expressfone1: how much info do you need? |
| 22:45.20 | implicit | or what info do you need? |
| 22:45.43 | expressfone1 | for avoid to send bad dialed numbres to my outgoin route |
| 22:46.29 | expressfone1 | and wait for operator mensage if number have errors |
| 22:46.53 | expressfone1 | it take 1 channel for nothing |
| 22:46.59 | implicit | oh ok |
| 22:47.05 | implicit | you only need this for cuba? |
| 22:47.12 | expressfone1 | yes |
| 22:47.15 | expressfone1 | only cuba... |
| 22:47.17 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-9-25.w82-122.abo.wanadoo.fr) |
| 22:47.31 | implicit | you need to be careful about managing your routes then |
| 22:47.38 | implicit | cause if something opens up and you don't know about it you are screwed |
| 22:48.40 | expressfone1 | np |
| 22:49.08 | expressfone1 | that beter than make fail call |
| 22:50.15 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net) |
| 22:52.48 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 22:58.40 | *** part/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 22:59.09 | *** join/#asterisk florz (nobody@odnb-d9baa519.pool.mediaWays.net) |
| 23:00.25 | *** join/#asterisk imcdona (imcdona@66-136.175-24.bham.rr.com) |
| 23:01.08 | imcdona | I have a question |
| 23:01.32 | imcdona | When using Asterisk and Festival I get the following error in Festival |
| 23:01.57 | imcdona | SIOD ERROR: wrong type of argument to car : wholeutt |
| 23:02.00 | imcdona | Any idea's? |
| 23:03.25 | hermie | kram: I have a suggestion for Mantis |
| 23:03.32 | kram | what's that? |
| 23:03.51 | hermie | kram: put 'AddType text/plain .patch' in the httpd.conf |
| 23:04.35 | kram | hermie: that's a fantastic idea |
| 23:04.49 | WilliamK | anyone here familiar with how SMS/800 works as far as provisioning numbers/routing/costs? |
| 23:05.08 | hermie | kram: and fix the link in the bug guidelines that points to a mailto instead of the wiki |
| 23:07.46 | firestrm | anyone here tried the SI-7800H 's yet? |
| 23:08.55 | kram | hermie: what link is that? |
| 23:12.09 | hermie | kram: the link to debugging info |
| 23:12.34 | hermie | kram: Basic Qualifications #4 |
| 23:14.06 | kram | i don't see a mailto |
| 23:15.48 | kram | other than the support@digium.com one |
| 23:15.49 | kram | are you sure? |
| 23:18.07 | kram | hermie? |
| 23:18.14 | empire667 | what's the difference in using capi or Bri-stuff? |
| 23:19.58 | *** join/#asterisk sephail (sephail@phalse.2600.COM) |
| 23:20.12 | nirs | hello |
| 23:20.14 | nirs | anybody home ? |
| 23:20.22 | sephail | Anyone here do any development with iaxclient? |
| 23:20.48 | nirs | anyone had worked with extconfig and realtime sip configurations ? |
| 23:21.04 | sivana | ~seen normast |
| 23:21.06 | jbot | normast is currently on #asterisk (1d 21h 52m 59s). Has said a total of 46 messages. Is idling for 17h 36m 2s |
| 23:21.51 | nirs | kram, you there mate ? |
| 23:22.09 | nirs | ~seen kram |
| 23:22.10 | jbot | kram is currently on #asterisk (2d 16h 6m 30s). Has said a total of 10 messages. Is idling for 4m 3s |
| 23:22.31 | nirs | !@$!@$#!#! |
| 23:22.35 | nirs | anyone around tonight ? |
| 23:23.29 | Mother_ | ... |
| 23:23.56 | kFuQ | call waiting shouldn't be this hard........ |
| 23:23.58 | kFuQ | :-( |
| 23:24.08 | Sedorox | ~seen sedorox |
| 23:24.09 | jbot | sedorox is currently on #asterisk (2h 14m 25s). Has said a total of 1 messages. Is idling for 1s |
| 23:24.15 | Sedorox | :-p |
| 23:24.51 | Strom_TM | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5749434893&rd=1 |
| 23:26.01 | Sedorox | nice |
| 23:27.02 | sivana | hehe |
| 23:35.45 | hermie | hmmm |
| 23:35.49 | hermie | ~seen jbot |
| 23:35.52 | jbot | jbot is currently on #ipaq (3d 5m) #how (3d 5m) #bzleague (3d 5m) #storm (3d 5m) #orkut (3d 5m) #uphpu (3d 5m) #va (3d 5m) #asterisk (3d 5m) #nslu2-linux (3d 5m) #magnia (3d 5m) #aegis (3d 5m) #ol (3d 5m) #tacobeam (3d 5m) #byumug (3d 5m) |
| 23:38.28 | Poincare | how can I check wich codec is being used for a call/channel? |
| 23:38.58 | file[laptop] | iax2 show channels, sip show channels |
| 23:39.43 | file[laptop] | Poincare: one of those.. |
| 23:39.52 | Poincare | file[laptop]: thanks, got it |
| 23:50.52 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:51.49 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 23:55.02 | Slainte | I want to add a logo to the PolycomIP600, but cant figure out which bitmap, or animation paramater to change in the ipmid.cfg |
| 23:56.10 | *** join/#asterisk ToyMan (~konversat@user-12lcqq2.cable.mindspring.com) |
| 23:57.48 | expressfone1 | implicit |
