| 00:04.29 | *** join/#asterisk flyn_ (n=mike@nc-67-77-5-49.dyn.embarqhsd.net) |
| 00:04.44 | froy | any people in here with clue about ztdummy? |
| 00:04.50 | flyn_ | Is anyone familiar with the realtime LDAP driver? |
| 00:05.10 | ThoMe | so. good night. |
| 00:05.23 | ThoMe | successful upgrade from asterisk 1.2X to 1.4 |
| 00:05.26 | ThoMe | yeah. |
| 00:05.30 | Strom_C | froy: what about ztdummy? |
| 00:05.30 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 00:05.55 | froy | I've got the ztdummy module installed, and zttest sees it, but I can't build a trunk... |
| 00:06.26 | froy | I'm wondering if I have to stick something in zaptel.conf that makes it see the "unconfigured" ztdummy |
| 00:06.34 | Strom_C | you shouldn't have to |
| 00:06.43 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:06.50 | Strom_C | did you load asterisk /after/ loading ztdummy? |
| 00:07.02 | froy | yes. I've restarted asterisk several times. |
| 00:07.09 | echelon | unpaidbill: so i can create an extension for an external sip address? |
| 00:07.12 | Strom_C | ok -- does chan_zap load? |
| 00:07.28 | froy | ztcfg -v shows 0 channels configured.. is that right for ztdummy? |
| 00:07.42 | Strom_C | yes, but that's not what i'm asking about |
| 00:07.47 | Strom_C | does asterisk load chan_zap? |
| 00:07.56 | froy | is looking |
| 00:08.18 | echelon | is it possible to create a local extension for an external SIP address? |
| 00:08.27 | Strom_C | echelon: YES |
| 00:08.29 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 00:08.30 | echelon | how? |
| 00:08.31 | Strom_C | read the damned book |
| 00:08.35 | echelon | i can't find it |
| 00:08.40 | Strom_C | sigh |
| 00:08.44 | Strom_C | have you no patience? |
| 00:08.52 | Strom_C | READ, not skim |
| 00:09.10 | *** join/#asterisk moy (n=moyhu@189.169.71.85) |
| 00:10.39 | froy | yup, looks like it did. |
| 00:10.42 | froy | <PROTECTED> |
| 00:10.42 | froy | <PROTECTED> |
| 00:10.42 | froy | <PROTECTED> |
| 00:10.42 | froy | <PROTECTED> |
| 00:11.05 | Strom_C | ok -- so what's the error you're getting? |
| 00:11.16 | froy | so I guess the pseudo is the ztdummy? |
| 00:11.21 | froy | stella*CLI> zap show channels |
| 00:11.22 | froy | <PROTECTED> |
| 00:11.22 | froy | <PROTECTED> |
| 00:11.35 | Strom_C | yes -- and please stop pasting stuff unnecessarily... |
| 00:11.53 | froy | Jun 24 18:11:38 WARNING[16764]: chan_iax2.c:8662 build_user: Unable to support trunking on user 'lumpnet' without zaptel timing |
| 00:11.58 | froy | sorry. |
| 00:12.24 | unpaidbill | haha |
| 00:12.48 | Strom_C | hm, odd...what version of zaptel / asterisk / linux? |
| 00:12.56 | *** join/#asterisk damjan (n=damjan@217.16.95.15) |
| 00:13.24 | froy | zaptel-1.2.24, asterisk-1.2.27, gentoo. |
| 00:13.58 | Strom_C | which linux kernel version? |
| 00:14.08 | froy | 2.6.24.7 |
| 00:14.40 | froy | hrm. |
| 00:15.17 | froy | maybe my problem is that I'm trying this in a vmware host, since timing sucks on vmware, but still, it should still see the ztdummy there. |
| 00:15.24 | Strom_C | i'm curious why you're running a hemorrhaging-edge version of the kernel and an old version of asterisk |
| 00:15.40 | froy | hehehe. |
| 00:16.04 | froy | I very recently upgraded the kernel, and the latest stable asterisk in gentoo portage is still 1.2.x |
| 00:16.31 | froy | I guess I could go to 1.4, but I'd have to go to what is considered unstable for gentoo right now... |
| 00:16.49 | Strom_C | installing asterisk from a package never tends to work right |
| 00:17.23 | froy | installing from a package and installing from an ebuild are two separate things ;) |
| 00:17.29 | Strom_C | My only hunch is that perhaps there's something about the latest kernel that's not working right with the older zaptel |
| 00:17.33 | Strom_C | blah blah blah blah blah |
| 00:17.52 | Strom_C | please don't give me that typical "It's not a package !!!" gentoo crap |
| 00:18.03 | Strom_C | for all intents and purposes, it's still a package :) |
| 00:18.46 | froy | well, yeah, but you don't have library problems, and the build won't work if you do, so it's kind of different, wouldn't you say? |
| 00:19.47 | froy | all an ebuild really does is do what you would do when you manually install... it just does it in a standbox first and then keeps track of all the stuff that will be installed. |
| 00:20.05 | echelon | Strom_C: you need to set up a peer first? |
| 00:20.34 | Strom_C | echelon: look, I can answer specific technical questions, but I'm not going to sit here and walk you through the whole process. |
| 00:20.40 | Strom_C | at least not for free. |
| 00:20.54 | echelon | well i'm not asking you to |
| 00:20.57 | echelon | just a yes or no |
| 00:21.08 | Strom_C | echelon: it's a far more complicated answer than what you're asking |
| 00:21.16 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:22.56 | coreyf52 | if i build asterisk-1.6beta9 on a test box and sip peer it with my live asterisk 1.4 box, do i risk any instability on the live box? |
| 00:22.59 | Strom_C | froy: what zaptel modules do you have loaded besides ztdummy> |
| 00:23.13 | Strom_C | coreyf52: not as such, no |
| 00:23.31 | froy | can I paste? hah. |
| 00:23.35 | Strom_C | pastebin |
| 00:23.36 | Strom_C | ~pb |
| 00:23.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 00:23.48 | coreyf52 | Strom_C: great thanks |
| 00:23.58 | froy | it's just four lines, but okay. |
| 00:24.25 | Strom_C | well, list them on one line then |
| 00:24.54 | froy | zttranscode, ztdummy, zaptel, crc_ccitt |
| 00:25.11 | froy | I think zttranscode got stuffed in there when I tried adding the ztdummy to zaptel.conf |
| 00:25.44 | Strom_C | hm |
| 00:26.11 | Strom_C | the only thing I can think of, then, is some weirdness between the kernel and the module |
| 00:26.19 | echelon | why can't i just do.. Channel: SIP/<number>@sipphone.com\n ?? |
| 00:26.21 | froy | nods |
| 00:26.30 | froy | okay, I'll keep hacking at it. |
| 00:26.39 | froy | thanks for all your kind input. |
| 00:26.40 | Strom_C | froy: try 1.4, just for kicks |
| 00:26.54 | froy | okay |
| 00:27.10 | froy | does 1.4 use the same 1.2 zaptel stuff? |
| 00:27.13 | NovceGuru | hmm..I thought HP had a procurve 24 port switch with 8 poe ports for ~$500 |
| 00:27.18 | Strom_C | no, there's zaptel 1.4 too |
| 00:27.28 | froy | okay. |
| 00:28.30 | froy | will I have to upgrade the other side of the trunk? hrm. |
| 00:28.43 | froy | that side has prod stuff going on :) |
| 00:29.20 | flyn_ | I'm trying to migrate a flat file sip.conf to LDAP, get "SIP/user-08cba848 is circuit-busy." |
| 00:29.47 | Strom_C | froy: no |
| 00:30.07 | froy | okay, thanks. |
| 00:37.38 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:37.38 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.21 (2008/06/12) Asterisk 1.2.29 (2008/06/03), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.11 (2008/05/28), Libpri 1.4.4 (2008/...) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 00:52.09 | echelon | i can't find anything related in the book! |
| 00:55.36 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 00:56.00 | flyn_ | It seems my problem was related to the use of fullcontact / AstAccountFullContact. Removing this attribute from my LDAP entries fixed my problem. |
| 00:57.28 | froy | It works! |
| 00:57.32 | froy | Hah! |
| 00:58.04 | froy | Strom_C: asterisk 1.4 works. |
| 00:58.11 | Strom_C | :) |
| 00:58.20 | froy | well, there you go ;) *giggle* |
| 00:58.26 | *** join/#asterisk mikehime (i=mikehime@207-5.97-97.tampabay.res.rr.com) |
| 00:58.31 | mikehime | waves |
| 00:58.47 | froy | apparently all the worthwhile zaptel development is in 1.4 these days, eh? :) |
| 00:59.18 | Strom_C | 1.2 is very close to being completely EOL'd |
| 00:59.20 | froy | thanks Strom_C |
| 00:59.38 | froy | wow, I didn't know that. |
| 00:59.56 | Strom_C | once 1.6 is released, 1.2 goes away |
| 01:00.36 | froy | and it's working on vm(clocks suck)ware, too. |
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| 01:07.46 | froy | okay, vmware still sucks. |
| 01:07.49 | froy | sigh. |
| 01:08.09 | froy | I guess I'll have to install it on the host OS. |
| 01:09.29 | froy | the audio gets choppy and the call eventually dies. vmware's "patent pending" clock still needs lots of work IMHO |
| 01:09.40 | froy | anyway, thanks again for your help! |
| 01:09.43 | froy | goes /home |
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| 02:03.36 | NovceGuru | does anybody deal with time warners business phone service? |
| 02:03.49 | NovceGuru | wondered if they were just sip trunks or actual POTS lines |
| 02:04.04 | jeev | nope, their cable modem only |
| 02:04.20 | jeev | and unfortunately, i put myself on the technical contact and the moron woman never pays so i find myself getting collection calls |
| 02:04.44 | NovceGuru | haha nice |
| 02:05.00 | NovceGuru | "sure, sir, we'll change the number, we've NEVER heard that one before" |
| 02:06.13 | NovceGuru | they're offering 5m x 768k internets and 6 phone lines for 336/month |
| 02:06.40 | NovceGuru | so obviously they're either POTS lines, or they're subtracting bandwidth before those speed stats |
| 02:06.51 | NovceGuru | but, they don't own the copper...so... |
| 02:07.54 | NovceGuru | "Business Class Phone. Time Warner Cable Business Class Phone provides small businesses with a reliable, digital voice service that includes unlimited..." |
| 02:08.16 | NovceGuru | Intresting to see what device they have giving 6 phone lines |
| 02:10.27 | jeev | heh |
| 02:10.30 | jeev | they're expensive |
| 02:10.39 | NovceGuru | cheaper then the telco :| |
| 02:10.44 | jeev | ahh |
| 02:11.09 | NovceGuru | but if they're just voip lines then I'll just keep a line or 2 from the telco and get the voip trunks from elsewhere |
| 02:11.40 | jeev | knowing them.. |
| 02:11.44 | jeev | it has to be voip eh |
| 02:11.50 | jeev | how do cable companies pull off phone without voip ? |
| 02:12.12 | NovceGuru | Well yeah thats what I was getting at when I pasted their quote |
| 02:12.16 | NovceGuru | "Digital voice service" |
| 02:12.25 | JT | cable companies almost always use VoIP to deliver phone |
| 02:12.37 | jeev | bastards |
| 02:12.46 | JT | although in theory it is possible to run TDM over a cable network |
| 02:16.49 | *** join/#asterisk kash (i=kash@laptop.tripleback.net) |
| 02:16.57 | kash | so uhm, where do i get suppserv.h |
| 02:17.40 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e58efbbaec43fe2a) |
| 02:25.34 | *** join/#asterisk hads (n=hads@120.138.17.30) |
| 02:25.56 | echelon | If I want to setup an extension that transfers the call to a number on a remote asterisk server? I would have to set up a peer? |
| 02:26.05 | kash | no |
| 02:26.19 | Nasra | any1 in here know the Rogers box ....kind of a voip....in Toronto.. |
| 02:26.21 | kash | use SIP/extension@server.name||tr |
| 02:26.28 | echelon | ||tr? |
| 02:26.33 | Strom_C | kash: don't use 't' or 'r' unless necessary |
| 02:26.37 | Strom_C | 'r' is particularly evil |
| 02:26.43 | kash | wh? |
| 02:26.50 | Strom_C | do you even know what 'r' does? |
| 02:26.54 | kash | no |
| 02:27.22 | Strom_C | then don't use it |
| 02:27.39 | Strom_C | it forces ringing on the channel and overrides all other inband progress, i.e. disconnected or changed number recordings... |
| 02:27.42 | Strom_C | or busy signals... |
| 02:28.08 | jaytee | oh, that's handy to know! |
| 02:28.17 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 02:28.54 | echelon | so how would i set the extension? |
| 02:29.06 | Strom_C | echelon: have you read ANY of the book? |
| 02:29.12 | echelon | yeah |
| 02:30.34 | echelon | exten => 411,1,Dial(SIP/18004664411@proxy01.sipphone.com) |
| 02:31.23 | Strom_C | assuming that proxy takes unauthenticated calls like that... |
| 02:31.29 | echelon | yes |
| 02:31.33 | Strom_C | well, good |
| 02:31.42 | echelon | i was able to reach it directly |
| 02:31.45 | Strom_C | then you didn't need to go kvetching about in here while you tried that now, did you |
| 02:33.02 | kash | Strom_C, where can i get the misdn package |
| 02:35.16 | echelon | ok, i reloaded |
| 02:36.08 | echelon | <PROTECTED> |
| 02:36.19 | echelon | that's what it said after i made the call |
| 02:36.44 | echelon | is that enough to transfer? |
| 02:36.56 | *** part/#asterisk shido6 (n=shido6@74-130-124-83.dhcp.insightbb.com) |
| 02:37.10 | Strom_C | echelon: what do you mean "transfer"? |
| 02:37.55 | echelon | after i call extension 411 on the local asterisk server, it should redirect to the specified number |
| 02:38.10 | echelon | let me paste the entire log |
| 02:38.39 | *** part/#asterisk hads (n=hads@120.138.17.30) |
| 02:38.39 | echelon | http://rafb.net/p/fZLotS15.txt |
| 02:39.08 | echelon | when i called directly from the softphone, it went through |
| 02:39.29 | echelon | so i'm guessing the redirect isn't taking place? |
| 02:40.47 | Strom_C | echelon: that's not a transfer |
| 02:41.04 | echelon | what's happening then? |
| 02:41.11 | Strom_C | you're bridging multiple call legs together |
| 02:42.00 | echelon | exten => 411,1,Dial(SIP/18004664411@proxy01.sipphone.com||tr) ..i should do this? |
| 02:42.24 | Strom_C | no no no no and no. |
| 02:42.34 | Strom_C | look at the error messages |
| 02:42.36 | Strom_C | the call is timing out |
| 02:42.44 | echelon | right |
| 02:43.10 | Strom_C | so why do you expect the options for "enable inband #-activated transfer" and "force ringing tone" to fix that problem? |
| 02:43.55 | echelon | when i call sip:18004664411@proxy01.sipphone.com it works fine |
| 02:43.59 | echelon | directly |
| 02:44.00 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8fe4d90ffd73dc44) |
| 02:45.41 | Strom_C | yes, but that's not what I asked you |
| 02:47.14 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:47.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:48.11 | echelon | i thought a transfer was what i was trying to do? |
| 02:48.37 | Strom_C | what did I say earlier |
| 02:48.40 | Strom_C | THIS IS NOT A TRANSFER |
| 02:48.55 | Strom_C | you should go read the book THOROUGHLY |
| 02:48.59 | Strom_C | and then give it another go |
| 02:49.02 | Strom_C | perhaps also read this: |
| 02:49.04 | Strom_C | ~101 |
| 02:49.05 | jbot | hmm... 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 02:49.22 | *** join/#asterisk Hadi- (n=Hadi@CPE001310492769-CM001225e00576.cpe.net.cable.rogers.com) |
| 02:49.23 | Hadi- | hellp |
| 02:49.24 | Hadi- | hello |
| 02:49.37 | Strom_C | echelon: i know you want it to work right now, but you need to sit down and learn what the hell you're doing before you hurt yourself. |
| 02:50.12 | Hadi- | is there a howto online on how to install asterisk with the res_config_pgsql module? |
| 02:52.08 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 02:52.12 | Alpha_AI | Hello |
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| 02:56.25 | klathzazt | in my voicemail.conf file- I want to use externnotify to execute a script. Can I use ${VM_CALLERID} as a parameter here? |
| 03:01.46 | *** join/#asterisk suma (n=suma@c-71-196-147-116.hsd1.co.comcast.net) |
| 03:02.19 | suma | hi |
| 03:02.32 | suma | i need help in maintaining high volume asterisk servers |
| 03:02.38 | suma | what are the things i need to take care ? |
| 03:02.47 | Strom_C | do you have a specific question, or are you looking for a consultant? |
| 03:03.10 | suma | My employer is going to deploy High Volume Asterisk Servers |
| 03:03.29 | suma | He asked me to prepare the checklist, I'm just blank |
| 03:03.44 | Strom_C | so you need a consultant, then. |
| 03:04.05 | suma | may be |
| 03:04.13 | suma | How would I get such information ? |
| 03:04.27 | Strom_C | well, any high-volume job has to be tailored to its exact application |
| 03:04.31 | Strom_C | there is no single "how-to" |
| 03:04.59 | Strom_C | what kind of call volume are you going to handle? |
| 03:05.42 | suma | around 50,000 calls in one hour |
| 03:05.49 | suma | It is for an IVR solution |
| 03:06.01 | suma | and call origination |
| 03:08.13 | Strom_C | ok -- what's that in terms of erlangs? |
| 03:08.31 | Strom_C | (I'm assuming you've run your erlang formulae already...) |
| 03:09.27 | suma | I should have this formula for each asterisk server ? |
| 03:09.55 | Strom_C | are you familiar with the erlang formulae? |
| 03:10.18 | suma | yes |
| 03:10.43 | suma | Calls per hour |
| 03:10.59 | suma | 10K erlangs |
| 03:11.04 | suma | sorry |
| 03:11.37 | suma | yes it is 10K erlangs |
| 03:12.45 | Strom_C | so each call lasts an average of 12 minutes? |
| 03:12.58 | echelon | it worked! |
| 03:13.00 | echelon | gawd |
| 03:13.13 | unpaidbill | and you didnt even have to learn anything in the process! |
| 03:13.37 | echelon | you were watching this entire time? |
| 03:13.38 | suma | 60 minutes each call |
| 03:13.49 | unpaidbill | i am always watching. |
| 03:14.00 | suma | 10,000 x 60 /60 |
| 03:14.01 | Strom_C | suma: that's not 10,000 erlangs for 50,000 60-minute calls in an hour |
| 03:14.01 | echelon | beyond sadistic |
| 03:14.18 | Strom_C | 50,000 one-hour calls in an hour is 50,000 erlangs |
| 03:14.18 | suma | oops |
| 03:14.27 | suma | yes |
| 03:14.43 | *** join/#asterisk stupidnic (n=foo@cpe-70-94-229-122.sw.res.rr.com) |
| 03:14.49 | Strom_C | so essentially what you're telling me is that you're just taking stabs in the dark here and you really don't know what you're doing |
| 03:14.56 | Strom_C | I would seriously recommend you hire a consultant |
| 03:15.08 | Strom_C | you are going to drown if you try to take this project on alone |
| 03:15.39 | stupidnic | is there some place I can read up about what the different syntax options mean for something like ${EXTEN:0, :1} etc mean |
| 03:15.41 | unpaidbill | i'm a fan of the furlong myself. |
| 03:15.43 | suma | no there are group, but I need to present to my Project Manager regarding technical needs |
| 03:15.48 | [TK]D-Fender | Strom_C: Arrrgh matey! |
| 03:16.20 | unpaidbill | man, i wish openoffice had a command line print utility |
| 03:16.28 | Strom_C | suma: you need to run some serious traffic analysis formulae before you even begin to think about technical needs |
| 03:16.37 | unpaidbill | you should see this .doc email->fax shit i am doing |
| 03:16.39 | unpaidbill | it's retarded. |
| 03:17.00 | stupidnic | unpaidbill: interfax? |
| 03:17.04 | suma | Anywhere I can look for the same ? |
| 03:17.11 | unpaidbill | interfax? |
| 03:17.15 | unpaidbill | hylafax! |
| 03:17.20 | stupidnic | xml to fax gateway provider |
| 03:17.22 | Strom_C | suma: that's what the erlang formulae are |
| 03:17.25 | klathzazt | suma: where are you located? I know some consultants in NYC area |
| 03:17.44 | Strom_C | I do consulting in the los angeles area (and remotely everywhere) |
| 03:17.46 | suma | I'm in longmont, CO |
| 03:18.51 | stupidnic | how can I strip the IAX2/2001-3 passed to VoiceMailMain to be just the extension? |
| 03:19.01 | Strom_C | stupidnic: use the CUT function |
| 03:19.20 | suma | Strom_C: what will be the outcome of the traffic analysis ? |
| 03:19.33 | suma | If I were asked to broadcast a file to 10,000 users, what I need to consider ? |
| 03:19.40 | klathzazt | hey stupidnic, you might be able to help me here: in my voicemail.conf file- I want to use externnotify to execute a script. Can I use ${VM_CALLERID} as a parameter here? |
| 03:19.42 | [TK]D-Fender | stupidnic: And that is not an EXTENSION. That is a channel based on the IAX2 PEER |
| 03:20.24 | stupidnic | [TK]D-Fender: well when I call VoiceMailMain() with ${EXTEN} that is what gets passwed to it |
| 03:20.25 | Strom_C | suma: it will give you a close enough approximation of real-world numbers to begin figuring out what kind of hardware you need to build out |
| 03:20.36 | echelon | but now that i have the extension set up, how do i associate it with a Channel? |
| 03:20.51 | stupidnic | klathzazt: I haven't a clue man. Sorry |
| 03:21.01 | [TK]D-Fender | stupidnic: And you could use CUT, or if the length and position of the portion you want to isolate is assured and consistent you can just strip it from the variable directly. |
| 03:21.03 | suma | Strom_C: that is awesome, thanks for the heads up |
| 03:21.28 | stupidnic | [TK]D-Fender: any specific examples for that? does it regex/preg? |
| 03:22.00 | [TK]D-Fender | stupidnic: http://www.voip-info.org/wiki-Asterisk+variables |
| 03:22.32 | Strom_C | suma: may I PM you? |
| 03:22.37 | stupidnic | [TK]D-Fender: thanks... that is what I was looking for... I couldn't find a reference to that syntax using my search terms |
| 03:22.59 | klathzazt | [TK]D-Fender: can I use asterisk variables in conf files? |
| 03:23.34 | suma | Strom_C: sure |
| 03:23.39 | [TK]D-Fender | klathzazt: USE them? Not really. |
| 03:25.15 | [TK]D-Fender | klathzazt: Well... you can use them in extensions.conf (and related) anyways ;) |
| 03:25.19 | klathzazt | [TK]D-Fender: I wanted to use externnotify string in teh conf file so that it calls a script when a voicemail is left- but I don't see a way to send parameters to the script- so it is essentially useless. Is there another way I can provision for voicemail calling for external application? |
| 03:25.21 | echelon | ok.. i tried to enter that into the APM, and got the error below.. http://rafb.net/p/uJxttI89.txt |
| 03:25.59 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.03 | echelon | asterisk manager* |
| 03:26.13 | klathzazt | [TK]D-Fender: so that would work then- I can use VM_CALLERID there- so that when the string is evaluated (execute the script) it would execute with the proper params. |
| 03:31.48 | stupidnic | [TK]D-Fender: Thanks for your guidance... that got me where I needed to be. I eventually went with ${CHANNEL:5:4} to get the extension to pass to VoiceMailMain |
| 03:32.16 | [TK]D-Fender | stupidnic: Now that will work with IAX2, but not SIP clearly... |
| 03:32.32 | stupidnic | yeah all clients are IAXy boxes |
| 03:32.32 | [TK]D-Fender | stupidnic: What is the callerid on your device? |
| 03:32.37 | [TK]D-Fender | EW |
| 03:32.47 | stupidnic | we set it global to the default CID for the company |
| 03:32.54 | *** join/#asterisk hfb (n=hfb@cpe-76-87-173-149.socal.res.rr.com) |
| 03:32.58 | stupidnic | less heartache dealing with NAT |
| 03:33.04 | [TK]D-Fender | stupidnic: I mean in your IAX2 peeer itself? |
| 03:33.43 | stupidnic | I did try CALLERID(num) but it defaulted to the global one for some reason |
| 03:34.00 | stupidnic | I assume its a global thing in asterisknow |
| 03:34.12 | [TK]D-Fender | stupidnic: "NO COMMENT" |
| 03:34.34 | [TK]D-Fender | stupidnic: You could also set a variable inside your IAX2 peer itself, but thats outside the scope of the GUI as well. |
| 03:40.47 | stupidnic | Yeah I have found many of the things I am trying to do "outside the scope of the gui" |
| 03:41.08 | stupidnic | I went with teh gui because I didn't want to be the one maintaining the installation |
| 03:41.18 | stupidnic | but at the end of the day, it turns out I won't get that lucky |
| 03:41.42 | *** join/#asterisk Gwayne (n=Gwayne@bb116-14-95-72.singnet.com.sg) |
| 03:42.53 | unpaidbill | i dont suppose any of you have use asterisk to integrate with a security alarm modem |
| 03:43.31 | unpaidbill | haha holy fuck AlarmReceiver |
| 03:43.36 | unpaidbill | god i love asterisk |
| 03:45.30 | stupidnic | I only wish asterisk came out a couple of years earlier... then I wouldn't have had to suffer through the NEC NX2000 unit I had to maintain at my employer for 7 years |
| 03:45.59 | stupidnic | using the command line to edit config files is like using Windows XP compared to that things interface |
| 03:47.17 | *** join/#asterisk mike8901 (n=mike@ool-45781de2.dyn.optonline.net) |
| 03:47.37 | echelon | if an extension 411 exists, would the Channel be SIP/411? |
| 03:47.54 | *** join/#asterisk rickbradley (n=rickbrad@www.eastcore.com) |
| 03:48.05 | stupidnic | only if it is registered under that name |
| 03:48.12 | stupidnic | in sip.conf as [411] |
| 03:48.17 | echelon | oh |
| 03:48.30 | mike8901 | is anyone here familiar with trixbox(since no one seems to be active in that channel)? |
| 03:48.36 | echelon | stupidnic: what if the account is registered on a remote server? |
| 03:48.58 | stupidnic | then you would call it via the registered peer identifier |
| 03:49.01 | rickbradley | should I still see chan_zap.so in /usr/lib/asterisk/modules/ under asterisk 1.6.1? |
| 03:49.07 | stupidnic | SIP/peername/extension |
| 03:49.11 | echelon | peer identifier? |
| 03:49.38 | stupidnic | channel Id a better term? |
| 03:49.43 | unpaidbill | ok this is so goddamn awesome. Wait("Zap/1-1", "5") in new stack .... AlarmReceiver("Zap/1-1", "") in new stack... == AlarmReceiver: Received Event 2305181132010198 |
| 03:49.44 | unpaidbill | !!!!! |
| 03:49.47 | unpaidbill | can i believe it. no |
| 03:49.58 | stupidnic | what ever the correct term is |
| 03:50.25 | unpaidbill | ok im going home |
| 03:50.27 | unpaidbill | good day to you sirs. |
| 03:51.04 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 03:51.11 | stupidnic | beer + TF2 == awesome |
| 03:51.17 | stupidnic | and with that... I am out |
| 03:52.14 | [TK]D-Fender | mike8901: trixbox is NOT supported here. |
| 03:52.22 | TrentCreek | it is now |
| 03:52.27 | TrentCreek | ;-) |
| 03:52.31 | echelon | so how does asterisk differentiate between a register user and a peer? |
| 03:52.44 | rickbradley | I'd had this tdm400p working under 1.4, then upgraded the machine's OS completely and put on 1.6.1 and now the lights on the wildcard are on on the 2 configured ports, and I "hear" the local phone port is connected but no dialtone and chan_zap isn't in the modules list |
| 03:52.53 | TrentCreek | ~peer |
| 03:52.54 | jbot | from memory, peer is the most elusive script kiddie this side of Jupiter |
| 03:53.08 | echelon | ~peer ? |
| 03:53.08 | jbot | i guess peer is the most elusive script kiddie this side of Jupiter |
| 03:53.30 | echelon | yeah, that's helpful |
| 03:54.45 | TrentCreek | Well, who knows why when I call a foreign exchange and the dial tone changes, asterisk considers that answered? |
| 03:55.00 | [TK]D-Fender | rickbradley: ..... there is no Asterisk 1.6.1.... |
| 03:55.29 | [TK]D-Fender | TrentCreek: What are you placing your call on? |
| 03:55.34 | echelon | where do i make a server definition? |
| 03:55.38 | rickbradley | hm. maybe I ended up on trunk rather than a tagged branch! |
| 03:55.42 | TrentCreek | An ATA |
| 03:55.47 | [TK]D-Fender | echelon: What kind of "server"? |
| 03:55.50 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:55.50 | rickbradley | goes back to svn and makes sure he didn't screw up |
| 03:56.00 | echelon | [TK]D-Fender: so i can set up a "peer identifier" |
| 03:56.00 | [TK]D-Fender | TrentCreek: No. What is your means of reaching the PSTN <- |
| 03:56.13 | [TK]D-Fender | echelon: Depends on the protocol. |
| 03:56.16 | TrentCreek | VOIP termination service |
| 03:56.17 | echelon | SIP |
| 03:56.29 | [TK]D-Fender | echelon: easy. sip.conf |
| 03:56.44 | rickbradley | yep. I'm on trunk. |
| 03:56.52 | [TK]D-Fender | TrentCreek: Could be they consider the call answered as soon as they accept it from you regardless of any extra call progress. |
| 03:57.48 | TrentCreek | I am just looking at the call progress ,,it was already accepted, rang a few american tones, then changes to foreign tone |
| 03:58.22 | [TK]D-Fender | TrentCreek: Yes, but through SIP, * doesn't care about tones. SIP is*is* your signalling. |
| 03:58.37 | [TK]D-Fender | TrentCreek: So you're out of luck. |
| 03:59.34 | echelon | do i need to set a password for a peer? |
| 04:00.48 | echelon | since they're registered on a remote server? |
| 04:01.26 | [TK]D-Fender | echelon: You SHOULD. And since when does your * know or care, or have anything to do with a "registration" on any OTHER server? |
| 04:01.55 | TrentCreek | [TK]D-Fender: I am using IAX2 for termination, if that helps |
| 04:02.05 | [TK]D-Fender | TrentCreek: Nope, same boat. |
| 04:02.12 | [TK]D-Fender | TrentCreek: Titanic ][ |
| 04:02.56 | echelon | [TK]D-Fender: all I'm trying to do is set up a Channel Id so I can invite them to the conference using a call file |
| 04:03.02 | TrentCreek | so NOTHING can be done about it? I would sure hate to get a calling card biz started and get 10,000 complaints about being charged for calls they attemped to make |
| 04:03.11 | echelon | channel id for a user that's not on the local server |
| 04:03.37 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 04:03.47 | [TK]D-Fender | TrentCreek: Blame your PROVIDER. They are the ones not passing progress to you. |
| 04:04.05 | [TK]D-Fender | echelon: Look at any phone setup. Thats all any peer is. |
| 04:04.32 | echelon | how do you mean? |
| 04:04.51 | TrentCreek | [TK]D-Fender: I just checked my CDR..seems showing some same calls that were not actually answered, but others do not appear |
| 04:05.02 | [TK]D-Fender | echelon: You make a peer so a phone you have can pace calls through *. Same for "coming from a server". Every call is "just another call" |
| 04:05.21 | [TK]D-Fender | echelon: "server" is not some magical word. It has no meaning. A call is a call is a call. |
| 04:05.52 | echelon | hmm |
| 04:05.56 | echelon | i see |
| 04:10.04 | echelon | so this is enough? [411]\n type=peer\n secret=blahblahblah\n username=18004664411\n host=proxy01.sipphone.com ? |
| 04:10.36 | [TK]D-Fender | echelon: it could be... if thats what the OTHER side is expecting. |
| 04:11.07 | echelon | but the password wouldn't be necessary at all, since i won't be logging in locally |
| 04:11.31 | [TK]D-Fender | echelon: Now that looks like your typical ITSP. Some ITSP's send you calls where they don't expect to gt challenged for calls. That is another matter. |
| 04:11.39 | [TK]D-Fender | echelon: .. HUH!?" |
| 04:11.51 | [TK]D-Fender | echelon: "logging in locally"... wtf? |
| 04:12.01 | echelon | logging into the local asterisk box |
| 04:12.20 | [TK]D-Fender | echelon: your eggs are well scrambled... |
| 04:12.49 | [TK]D-Fender | echelon: what is this "you" you are referring to? "Logged in"? Whats that supposed to mean? |
| 04:13.09 | [TK]D-Fender | echelon: Perhaps you shuold be more thorough in your description of the pieces in play. |
| 04:13.28 | echelon | i just need that peer id setup so I can invite the remote user to the conference |
| 04:13.50 | [TK]D-Fender | *b00m* |
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| 04:14.32 | echelon | [TK]D-Fender: to automate a call I need to create a call file right? |
| 04:15.15 | [TK]D-Fender | echelon: define "automate". And there is AMI Originate which is equivalent to "call files" as well, as well as "originate" from * CLI. |
| 04:16.42 | echelon | to have asterisk call the remote user and connect it to the Conference room |
| 04:17.24 | [TK]D-Fender | echelon: yes, fo * to place the call, and not be initiated from an existing channel for the purpose of bridging, yes. |
| 04:17.33 | Alpha_AI | hello |
| 04:17.36 | Alpha_AI | im getting this error |
| 04:17.41 | Alpha_AI | can someone help me? |
| 04:17.55 | echelon | Alpha_AI: why don't you just ask your quiestion |
| 04:17.57 | Alpha_AI | [Jun 25 14:13:09] NOTICE[2544] chan_iax2.c: Rejected connect attempt from 150.101.183.34, request '61734397063@incoming' does not exist |
| 04:18.32 | pputman | Alpha_AI, do you have that extension in the incoming context? |
| 04:18.33 | Alpha_AI | I tried to make an incoming call to 61734397063 |
| 04:18.33 | echelon | [TK]D-Fender: so how would i invoke a call with a remote user? |
| 04:18.44 | [TK]D-Fender | echelon: VOODOO. |
| 04:18.55 | echelon | using originate in CLI? |
| 04:19.17 | [TK]D-Fender | Alpha_AI: Its telling you exactly what extension it is looking for and in which context. It clearly isn't finding a match in there. Go fix it. |
| 04:19.21 | Alpha_AI | yes I do have that extension in there. |
| 04:19.44 | [TK]D-Fender | echelon: In CLI you would type the command right on the line like any other command you'd issue there. |
| 04:19.55 | [TK]D-Fender | Alpha_AI: * begs to differ. |
| 04:20.02 | [TK]D-Fender | Alpha_AI: PASTEBIN your dilaplan. |
| 04:20.04 | [TK]D-Fender | ~pb |
| 04:20.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:20.07 | [TK]D-Fender | ^^^^ |
| 04:20.09 | [TK]D-Fender | dialplan* |
| 04:21.06 | echelon | what's tech/data? |
| 04:21.52 | [TK]D-Fender | echelon: "core show application dial" |
| 04:21.56 | echelon | originate <conference exten #> <remote user exten #> ? |
| 04:22.27 | [TK]D-Fender | echelon: READ |
| 04:23.06 | Strom_C | [TK]D-Fender: he's been steadfastly refusing to read the book or the documentation all day |
| 04:23.09 | Strom_C | i wouldn't bother |
| 04:23.20 | echelon | i _am_ reading |
| 04:23.50 | Strom_C | echelon: you need to step away from the computer, read through the book, and then come back |
| 04:24.11 | echelon | how am i supposed to call a remote user if the only parameter it asks for is Channel Id? |
| 04:24.29 | echelon | SIP/user@remotehost ? |
| 04:24.29 | Alpha_AI | I think i fix it |
| 04:24.35 | Alpha_AI | but i got another error now |
| 04:24.38 | Alpha_AI | chan_sip.c: No such host: 100 |
| 04:25.14 | pputman | Alpha_AI, that doesn't help at all without pastebin'ing more information, such as the console logs... |
| 04:25.49 | Alpha_AI | oh ok |
| 04:26.01 | Alpha_AI | im using asterisknow how do i get access to a copy of the dialplan |
| 04:26.38 | [TK]D-Fender | Alpha_AI: if you don't even know what the dialplan is, stop right now, downlaod the book and get reading. You're trying to run without even knowing what LEGS are. |
| 04:26.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:26.52 | echelon | i know what a dialplan is |
| 04:27.12 | echelon | Alpha_AI: extensions.conf |
| 04:27.13 | pputman | or trying to call without knowing what call LEGS are :/ |
| 04:27.14 | [TK]D-Fender | Alpha_AI: And most things in AsteriskNOW are controlled through the GUI which is not supported here anyways. Please read the channel topic for links to their support channel |
| 04:27.15 | Alpha_AI | i got it to work |
| 04:27.23 | Alpha_AI | i received an incoming call |
| 04:27.29 | Alpha_AI | the instructions from the provider were wrong |
| 04:27.37 | Alpha_AI | i hate that part the most |
| 04:33.04 | Alpha_AI | now i can make outbound calls |
| 04:41.29 | echelon | this is the address i'm trying to reach.. sip:18004664411@proxy01.sipphone.com ..what format would it's channel format be? |
| 04:41.37 | echelon | if i wanted to use it in a Dial() parameter |
| 04:41.49 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-153-243.dsl.teksavvy.com) |
| 04:43.09 | [TK]D-Fender | echelon: http://www.voip-info.org/wiki/view/SIPphone |
| 04:45.30 | *** join/#asterisk macros73 (n=cs@c-67-165-49-184.hsd1.pa.comcast.net) |
| 04:48.16 | echelon | oi.. that's a lot better |
| 05:01.02 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) |
| 05:03.24 | macros73 | Getting smacked around on a new PiAF / Asterisk 1.6b9. I have two test extensions defined and registered. Calls from one to the other return 404 errors. Any advice? |
| 05:04.15 | TrentCreek | why not try new asterisk with a new product? |
| 05:05.24 | gitguy | asterisk with new product? |
| 05:06.07 | [TK]D-Fender | macros73: I advise you to see WHAT is 404-ing and fix it. |
| 05:06.51 | TrentCreek | you did say NEW PiaF, but using an old version of asterisk |
| 05:07.33 | macros73 | Bloody everything is 404'ing. Okay, Wireshark shows a 401 first. |
| 05:07.51 | macros73 | Invite, 401, Ack, Invite, 404, ACK, Options, OK. |
| 05:07.54 | [TK]D-Fender | macros73: Stop looking at wireshark. You should be looking at SIP debug from * CLI. |
| 05:08.16 | [TK]D-Fender | macros73: And pastebin is your friend.... |
| 05:10.46 | macros73 | okay, enagled sip debug for peer 102 |
| 05:19.34 | macros73 | I think it's trying to NAT on the local LAN |
| 05:20.01 | macros73 | sending to 192.168.2.235:5071 (NAT), but there's no need to NAT on the LAN. |
| 05:20.33 | [TK]D-Fender | macros73: almost a complete viewpoint... |
| 05:20.42 | [TK]D-Fender | macros73: jsut missing critical backup |
| 05:21.04 | macros73 | [TK]D-Fender: Which would be...? |
| 05:21.22 | [TK]D-Fender | macros73: You showed who it was contacting, not where it expects an answer to be sent to. |
| 05:22.00 | [TK]D-Fender | macros73: if its on the local LAN, it is still being sent to the right IP. |
| 05:22.07 | [TK]D-Fender | macros73: so that is not itself a problem. |
| 05:26.28 | *** join/#asterisk fnordus (n=dnall@70.71.224.2) |
| 05:26.34 | macros73 | [TK]D-Fender: Peer audio RTP is at port 192.168.2.235:5012, Looking for 102 in from-internal (domain 192.168.2.183), reliably transmitting (NAT) to 192.168.2.235:5072, SIP/2.0 404 Not Found |
| 05:27.02 | [TK]D-Fender | macros73: Congratulations, that officially pins it as a DIALPLAN error. |
| 05:27.21 | [TK]D-Fender | macros73: "Looking for 102 in from-internal" |
| 05:27.40 | [TK]D-Fender | macros73: Pretty blatant message there... |
| 05:28.40 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 05:29.24 | macros73 | [TK]D-Fender: Okay. Aside from "Fix it", what should I check next? |
| 05:29.56 | [TK]D-Fender | macros73: Its telling you exactly what extensio the call is targeting and in which context. You clearly don't have a match there.... |
| 05:32.00 | kmshanah | is there a simpler way to get the number of the extension initiating a call than pulling a substring out of the ${CHANNEL} variable? |
| 05:32.55 | unpaidbill | caller id? |
| 05:33.16 | kmshanah | this is to set the caller id in the outgoing context, based on the caller's extension |
| 05:33.51 | [TK]D-Fender | kmshanah: "core show function IAXPEER" , "core show function SIPCHANINFO" |
| 05:35.24 | kmshanah | [TK]D-Fender: ah, that's what I need - thanks :) |
| 05:35.57 | *** part/#asterisk rickbradley (n=rickbrad@www.eastcore.com) |
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| 05:38.52 | Alpha_AI | hello |
| 05:39.40 | unpaidbill | wtf, the ademco contact id protocol isnt free? those bastards. |
| 05:42.11 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
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| 05:46.57 | macros73 | [TK]D-Fender: Thanks for your pointers earlier. My PC died, now I'm on the laptop trying to fix the PC, so...the phone system will have to wait. |
| 05:48.13 | [TK]D-Fender | macros73: Well that last message tels you exaclty whats wrong and where to go to fix it. Not a guessing game and this shouldn't take more than a moment to correct. |
| 05:48.44 | unpaidbill | when it rains it pours, macros |
| 05:49.11 | unpaidbill | and apparently the spec for SIA contact id isnt pay-for.. damn voip-info.org misinformationing me |
| 05:51.01 | macros73 | [TK]D-Fender: If I knew my way around a dialplan, that would be true. :) from-internal points to from-internal-transfer, which includes several other items.. |
| 05:51.38 | [TK]D-Fender | macros73: No matter, it ain't Raw-Cat Science. |
| 05:59.50 | [TK]D-Fender | Alright, bed time. Later all. |
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| 06:14.51 | echelon | ok, i've had it.. let me put it in simple terms |
| 06:15.14 | echelon | what's the best way to dial-out from within a conference using the app_conference module? |
| 06:20.27 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 06:22.57 | obnauticus | http://digg.com/odd_stuff/What_the_hell_google <-- LMAO!!! |
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| 06:37.41 | J4zen | Does anyone know a good text-to-speech application which supports the dutch language? |
| 06:38.42 | *** join/#asterisk Oy90 (n=ivan@213.187.111.94) |
| 06:38.53 | TrentCreek | Arabic? |
| 06:43.27 | J4zen | Dutch |
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| 07:00.48 | yangvnc | hio tzafrir_laptop |
| 07:01.03 | tzafrir_laptop | hi |
| 07:01.45 | unpaidbill | so who loves to partay |
| 07:03.18 | echelon | asterisk is a lost cause.. freeswitch is ahead of the game |
| 07:03.46 | unpaidbill | k |
| 07:04.17 | Dovid | i like freeswitch but i think brian is too full of hot air |
| 07:05.29 | Strom_C | echelon: did you read the book like I've been telling you to? |
| 07:05.56 | echelon | it's too late now, i'm not coming back |
| 07:06.11 | unpaidbill | reading... it's.....tohard |
| 07:06.32 | echelon | they have dialing-out directly from the conference |
| 07:06.55 | unpaidbill | fancy |
| 07:07.05 | Strom_C | ~wglwat |
| 07:07.06 | jbot | it has been said that wglwat is well, good luck with all that |
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| 07:33.01 | TrentCreek | ~unpaidbill |
| 07:33.16 | TrentCreek | we need to fix that |
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| 07:50.40 | casix | hello |
| 07:51.25 | casix | when I transfer a call the callerid of the new call is the callerid of the original call. Can I know (in wich variable) can I see who is making the transfer?? |
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| 07:53.56 | mvanbaak | J4zen: try ceptral |
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| 07:59.07 | J4zen | mvanbaak: thanks |
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| 08:03.12 | J4zen | mvanbaak: I see no Dutch voice support in that application? https://www.cepstral.com/cgi-bin/store/home |
| 08:03.23 | J4zen | Are you sure they support it? |
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| 08:48.18 | Great_Anta_Baka | has anyone used sip ver 3 with a polycom 650 |
| 08:48.37 | Great_Anta_Baka | mine keeps crashing since i have updated it |
| 08:48.44 | LuisTorres | Morning |
| 08:49.29 | Great_Anta_Baka | morning |
| 08:52.19 | MooingLemur | I wonder if these support SIP: http://www.pizdaus.com/pics/MnyQvDqtiqBs.jpg |
| 08:53.41 | Great_Anta_Baka | lol |
| 08:59.26 | J4zen | MooingLemur: I want them |
| 08:59.33 | J4zen | Should look great on my desk |
| 09:00.51 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 09:06.23 | J4zen | Does anyone know a dutch voice-set for asterisk, commercially available |
| 09:06.48 | J4zen | Superbartt, mvanbaak: happen to know any? |
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| 09:20.18 | J4zen | Does anyone know a good european VoIP hardware(telephones and such) supplier? |
| 09:22.30 | gr0mit | J4zen, what are you looking for? |
| 09:22.31 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 09:22.39 | gr0mit | what qty and what products? |
| 09:26.22 | J4zen | Well per order it'd probably be between 10~30 phones |
| 09:26.39 | J4zen | PolyCom 550's , 650's, 670's and their operator expansion sets |
| 09:27.07 | J4zen | possibly some PRI/BRI interfaces, but only every now and then |
| 09:28.59 | J4zen | happen to know any gr0mit? |
| 09:30.27 | gr0mit | well voipon.co.uk |
| 09:30.33 | gr0mit | seem to be ok |
| 09:30.38 | gr0mit | i can get you snoms |
| 09:32.06 | gr0mit | but if you want polycom then voipon are good |
| 09:32.24 | J4zen | ouch |
| 09:32.27 | J4zen | that's GBP |
| 09:32.36 | J4zen | really hurts to bulk-order there :P |
| 09:32.47 | gr0mit | why? |
| 09:33.13 | J4zen | exchange rate |
| 09:33.21 | J4zen | we operate in EUR |
| 09:35.08 | gr0mit | click the EU flag |
| 09:35.14 | gr0mit | they will price in EUR too |
| 09:35.20 | J4zen | ah :) thanks |
| 09:35.25 | J4zen | *doh* moment |
| 09:36.49 | gr0mit | but i dunno if that is converted from GBP back to EUR |
| 09:37.22 | gr0mit | if you want snommys let me know though |
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| 09:45.02 | l0verb0y | hey |
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| 10:00.29 | dieno | can any one tell what is the solution to speed up festival because right now its speaking like robot |
| 10:00.33 | dieno | with low batteries :D |
| 10:01.46 | Strom_C | maybe you've got some sampling rate mismatch |
| 10:01.58 | dieno | hmm i think |
| 10:02.18 | dieno | because in last file i didnt compiled sample files and it was working perfect |
| 10:02.28 | dieno | but this time its really ... |
| 10:02.48 | dieno | and how do i change the festival voice |
| 10:03.22 | Strom_C | no |
| 10:03.24 | Strom_C | not sample files |
| 10:03.26 | Strom_C | sampling rate |
| 10:03.34 | Strom_C | two completely different things |
| 10:03.44 | dieno | ohh sorry then |
| 10:03.51 | dieno | how do i get this done |
| 10:04.50 | Great_Anta_Baka | has anyone used sip version 3 on a polycom 650, mine keeps crashing since i updated it |
| 10:04.51 | dieno | can you tell me how do i sort this out |
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| 10:16.56 | tapic | hi all, |
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| 10:33.33 | wonderworld | how would i configure asterisk to recieve sip-calls from not registered clients? is that possible? |
| 10:34.47 | wonderworld | to make it possible for people to call me with any softphone without having to enter login information.... |
| 10:42.49 | Great_Anta_Baka | well you cant just phone from a sip client |
| 10:43.01 | Great_Anta_Baka | you have to be a registered user on some sip server |
| 10:43.15 | Great_Anta_Baka | otherwise the call just wont know where to go |
| 10:43.43 | Great_Anta_Baka | but i am just a n00b wonderworld :P |
| 10:44.01 | wonderworld | wouldn't it be possible to do somthing like call:ip-address ? |
| 10:44.04 | Great_Anta_Baka | also i dont know if asterisk accepts anonymous sip calls by default :/ |
| 10:44.10 | wonderworld | i am a noob too, that's why i am asking ;) |
| 10:44.30 | Great_Anta_Baka | yes thats how you make a sip call but the calling party has to be registered on a sip server as well |
| 10:45.06 | Great_Anta_Baka | i havent heard of a stand alone sip client |
| 10:46.16 | wonderworld | i think you can make calls with ekiga without having registered first. but i can be wrong here too. |
| 10:47.15 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 10:50.28 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 10:50.31 | tapic | I have a AGI DEBUG output where I receive "510 Invalid or unknown command" for Answer command, would someone like to examine it? |
| 10:50.42 | tapic | thank you very much... |
| 10:51.36 | wonderworld | tapic: php? |
| 10:53.43 | Mike8861 | hello everybody |
| 10:53.53 | Mike8861 | would any like to help me with IM ?? |
| 10:54.01 | Mike8861 | I am running asterisk 1.4 |
| 10:55.55 | tapic | wonderworld : I use fastagi |
| 10:56.14 | tapic | wonderworld : I actually use asterisk.net agi |
| 10:56.42 | wonderworld | sorry, no idea then. i scripted some php-agis... |
| 10:57.00 | tapic | I have typed agi debug in asterisk console and have some output about the issue |
| 10:57.26 | wonderworld | i can have a look... |
| 10:57.37 | tapic | thanks.. |
| 10:58.14 | Mike8861 | can any one help with IM ? |
| 10:58.19 | wonderworld | do other commands work fine in your agi? is it just about Answer ? |
| 10:58.50 | *** join/#asterisk tapic (n=tap@88.255.77.200) |
| 11:00.17 | dieno | anyone with festival experience |
| 11:02.37 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 11:03.12 | cristina_crow | Hello |
| 11:03.37 | dieno | helo, do you know something abt festival |
| 11:04.07 | cristina_crow | I am using asterisk 1.6 |
| 11:04.12 | cristina_crow | I want to use SIP over TLS |
| 11:04.20 | cristina_crow | could you please indicate some tutorials for me? |
| 11:04.42 | cristina_crow | I have been doing some research on google, but I must have searched in the wrong place |
| 11:04.59 | cristina_crow | I've tried the digium forum TLS configs |
| 11:05.13 | cristina_crow | but asterisk doesn't listen on TLS on port 5061, as expected |
| 11:05.19 | cristina_crow | thanks in advance |
| 11:05.59 | mvanbaak | wonderworld, Great_Anta_Baka: you can indeed make such calls |
| 11:06.09 | mvanbaak | in sip.conf is a setting for anonymous calls |
| 11:06.53 | mvanbaak | make sure you also set the context there |
| 11:07.19 | mvanbaak | I have: context=anon-sip |
| 11:07.26 | mvanbaak | and in extensions.conf I have |
| 11:07.30 | mvanbaak | [anon-sip] |
| 11:07.38 | mvanbaak | exten => mvanbaak,1,..... |
| 11:08.13 | wonderworld | mvanbaak: cool thanks i'll try that out |
| 11:12.07 | wonderworld | mvanbaak: i can't find that option for sip.conf (http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf) |
| 11:14.44 | *** join/#asterisk RoyK (n=roy@ip-183-25-149-91.dialup.ice.no) |
| 11:14.52 | *** part/#asterisk redax (i=redax@r6.hu) |
| 11:17.46 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:19.56 | mvanbaak | gimme a minute |
| 11:21.24 | mvanbaak | wonderworld: |
| 11:21.25 | mvanbaak | ;allowguest=no ; Allow or reject guest calls (default is yes) |
| 11:22.07 | Great_Anta_Baka | thanks mvanbaak |
| 11:22.16 | Great_Anta_Baka | has anyone used sip version 3 on a polycom 650, mine keeps crashing since i updated it |
| 11:26.12 | wonderworld | mvanbaak: tnx |
| 11:31.06 | Mike8861 | can any one help with IM ? |
| 11:31.30 | Mike8861 | i cannot implement SIP IM with asterisk 1.4 |
| 11:31.53 | wonderworld | mvanbaak: ok i changed the sip config. but how would i call my asterisk now from a softphone? just sip:IP-ADDRESS ? |
| 11:32.45 | mvanbaak | sip:exten_in_the_context_you_specified@ip_address |
| 11:33.53 | mvanbaak | like in my setup: sip:mvanbaak@lunteren.vanbaak.info |
| 11:34.42 | yangvnc | I am getting a strange error with GotoIfTime syntax - the call cannot be routed...http://www.pastebin.sk/en/7161/ |
| 11:35.39 | Mike8861 | can any one help with IM ? |
| 11:36.42 | mvanbaak | yangvnc: I dont see any errors with the GotoIfTime syntax there |
| 11:36.49 | mvanbaak | all I see is other errors |
| 11:36.56 | mvanbaak | # |
| 11:36.57 | mvanbaak | [Jun 25 13:31:55] WARNING[24591]: pbx.c:1539 func_args: Can't find trailing parenthesis? |
| 11:37.08 | yangvnc | mvanbaak: Its a SIP error like - only 1 sip call ata atime or? |
| 11:37.09 | mvanbaak | [Jun 25 13:31:55] WARNING[24593]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 11:37.25 | mvanbaak | fix those |
| 11:37.36 | yangvnc | well how, i wonder why it doesnt call |
| 11:37.50 | mvanbaak | what is it you dont understand ? |
| 11:38.03 | mvanbaak | the first is because you miss a ) |
| 11:38.10 | yangvnc | oh, where |
| 11:38.11 | mvanbaak | the second is a 'no route to destination' |
| 11:38.23 | yangvnc | where do i miss a = |
| 11:38.25 | yangvnc | ) |
| 11:38.26 | mvanbaak | yangvnc: in the context buster |
| 11:38.34 | mvanbaak | you did not pastebin that |
| 11:38.57 | yangvnc | oh geez i will paste |
| 11:40.39 | Mike8861 | helllo, would anyone be able to help ? |
| 11:41.01 | Mike8861 | I am running asterisk 1.4, but cannot get SIP message to work. |
| 11:41.05 | yangvnc | http://www.pastebin.sk/en/7162/ |
| 11:42.40 | tapic | Any idea on what encoding types are supported for socket communication in Asterisk 1.4? |
| 11:46.01 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 11:46.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 11:49.28 | Mike8861 | uhmm.... |
| 11:49.52 | Mike8861 | hey tapic |
| 11:51.09 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 11:51.48 | yangvnc | mvanbaak: i wanted to route all calls on saturdays to the contxt in frondesk, however i don't suceed in that |
| 11:52.52 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 11:53.06 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:53.21 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 11:53.25 | Uatec | o/ |
| 11:54.14 | *** join/#asterisk bartpbx (n=bartpbx@217.24.210.202) |
| 11:54.20 | bartpbx | hello everybody |
| 11:54.29 | Uatec | hi there |
| 11:54.44 | bartpbx | im stuck in the convertion of our asterisk system from 1.2 to 1.4 |
| 11:55.04 | bartpbx | there seams to be a difference in the voicemail unixodbc system i dont unterstnad |
| 11:55.19 | bartpbx | anyone has woring 1.4 with odbc voicemailstorage running? |
| 11:55.46 | hi365 | tried to use gmail for vm imap storage |
| 11:56.20 | Mike8861 | hello |
| 11:56.42 | Mike8861 | can anyone help with IM ? I know its not easy, but would like to give it a try |
| 11:57.04 | hi365 | IM what? |
| 11:58.07 | RoyK | thinks that might mean instant messanging with sip |
| 11:58.45 | Mike8861 | using asterisk 1.4 to send and recieve IM with xlite client |
| 11:58.56 | Mike8861 | I have get presence working, but not the IM part.... |
| 11:59.05 | hi365 | never did anything of the sort |
| 12:00.38 | Mike8861 | uhm.....thanks |
| 12:00.56 | Mike8861 | some article says that Patching asterisk will make IM work |
| 12:01.06 | *** join/#asterisk Porpes (n=chatzill@189-19-2-118.dsl.telesp.net.br) |
| 12:01.13 | Mike8861 | is there any guid on patching asterisk ? |
| 12:03.10 | tapic | hi Mike8861 |
| 12:03.36 | Mike8861 | hello tapic ^_^ |
| 12:03.47 | Mike8861 | can I help u ? |
| 12:04.25 | tapic | thanks! do you know if asterisk 1.4 AGI does not support UTF8 encoding |
| 12:06.08 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:06.48 | *** join/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 12:07.11 | Mike8861 | not really sure about this. |
| 12:07.52 | Mike8861 | I never try agi, but can u change the encoding type on the scripting file ? |
| 12:08.17 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:08.22 | cristina_crow | hello, is there still anyone that could help me on TLS, pls?:-s |
| 12:09.15 | Maliuta | TLS is a very generic support request |
| 12:09.36 | cristina_crow | humm... I only need to make a register from eyebeam |
| 12:09.43 | yangvnc | helo [TK]D-Fender |
| 12:09.49 | cristina_crow | the problem seems to be with the SSL cert |
| 12:10.04 | cristina_crow | I have created the keys (public and private) using asteriskgenkey |
| 12:10.14 | cristina_crow | then created a .csr request with openssl |
| 12:10.23 | cristina_crow | and signed that csr using my local CA |
| 12:10.33 | cristina_crow | then put the asterisk.pem in /etc/asterisk/ |
| 12:10.38 | cristina_crow | in the asterisk CLI |
| 12:10.41 | cristina_crow | I type |
| 12:10.43 | cristina_crow | cvintila-ipsec01*CLI> keys init |
| 12:10.43 | cristina_crow | cvintila-ipsec01*CLI> sip reload |
| 12:10.43 | cristina_crow | <PROTECTED> |
| 12:10.43 | cristina_crow | <PROTECTED> |
| 12:10.43 | cristina_crow | <PROTECTED> |
| 12:10.44 | cristina_crow | SSL cert error </etc/ssl/certs/asterisk.pem> |
| 12:10.48 | cristina_crow | <PROTECTED> |
| 12:11.02 | Maliuta | did you put a passphrase on the key file? |
| 12:11.07 | cristina_crow | same if asterisk.pem is in /etc/asterisk/, and in /etc/ssl/certs/asterisk.pem |
| 12:11.13 | cristina_crow | yes, I have put a password |
| 12:11.16 | Maliuta | because that could cause an "issue" |
| 12:11.21 | yangvnc | [TK]D-Fender: I am getting a strange error with GotoIfTime syntax - the call cannot be routed by SIP ...http://www.pastebin.sk/en/7161/ my extensions are in http://www.pastebin.sk/en/7162/ |
| 12:11.21 | cristina_crow | ugh :-s |
| 12:11.30 | cristina_crow | ok..I'll remove the password |
| 12:11.31 | cristina_crow | thanks |
| 12:11.34 | Maliuta | as a general rule you don't put passphrases on server certs |
| 12:11.55 | Maliuta | notes cristina_crow has "the genius gene" ;) |
| 12:12.21 | cristina_crow | sry.. |
| 12:12.22 | cristina_crow | but |
| 12:12.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:12.33 | cristina_crow | it won't accept the key without a passphrase:D |
| 12:12.36 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:12.38 | [TK]D-Fender | yangvnc: There's obviously a probelm with SIP/80. Go look at it. |
| 12:12.40 | cristina_crow | Enter pass phrase for asterisk.key: |
| 12:12.41 | cristina_crow | 13444:error:28069065:lib(40):UI_set_result:result too small:ui_lib.c:849:You must type in 4 to 8191 characters |
| 12:12.41 | cristina_crow | Enter pass phrase for asterisk.key: |
| 12:12.49 | cristina_crow | any other ideas:-/ |
| 12:12.50 | Mike8861 | tapic, what does AGI do ? |
| 12:13.30 | Maliuta | I'd generate a new one, without the passphrase. |
| 12:13.34 | Maliuta | I also use plain old openssl to generate my keys |
| 12:13.56 | cristina_crow | mhm...fine..I'll do the keys without using asteriskgenkey :d |
| 12:14.04 | cristina_crow | let me see if I can get it working |
| 12:14.09 | [TK]D-Fender | Mike8861: AGI is an interface so you can use a "normal" programming language interface with a caller for more complete processing. |
| 12:15.03 | Mike8861 | thanks [TD]D-Fender, how can i learn AGI programming language ? is there any docs recoomand to start |
| 12:15.08 | yangvnc | [TK]D-Fender: yes SIP/80 phone is beng unreachable becouse its out of electricity, i would just like the extensions to ring a SIP/detel/041710598 on saturday and not the other extensions |
| 12:15.59 | cristina_crow | I can't generate from openssl a key without a passphrase |
| 12:16.01 | cristina_crow | it won't let me |
| 12:16.02 | cristina_crow | :-? |
| 12:16.08 | *** join/#asterisk Porpes (n=sergio@189-19-2-118.dsl.telesp.net.br) |
| 12:16.09 | cristina_crow | what should I do about it? |
| 12:16.14 | J4zen | Does anyone have a SoundPoint® IP 550 with an expansion module, i found the Backlit expansion modules from the SoundPoint IP series but it only states support for the 650 |
| 12:16.29 | J4zen | yet they use the 550 in combination with that same expansion module for their callcenter solution |
| 12:16.43 | [TK]D-Fender | J4zen: only the 6XX series supports expansion modules |
| 12:16.52 | J4zen | [TK]D-Fender: Awesome, thanks |
| 12:17.25 | tapic | my AGI just answers the call but if the encoding is UTF8 asterisk 1.4 can not receive correct messages, I changed the socket encoding (I use fastAGI) and the problem solved |
| 12:17.40 | yangvnc | [TK]D-Fender: however i don't receive a call to that mobile number |
| 12:17.53 | tapic | now I can not connect to AMI |
| 12:20.19 | yangvnc | [TK]D-Fender: i have put SIP/80 out now there are no errors anymore in CLI, but SIP/detel-001c4d80 is ringing , however its not ringing |
| 12:21.30 | [TK]D-Fender | yangvnc: detel? HUH? |
| 12:21.58 | [TK]D-Fender | yangvnc: make a new pastebin. |
| 12:22.57 | yangvnc | [TK]D-Fender: i am limited to two channels by my vpip provider |
| 12:23.25 | [TK]D-Fender | yangvnc: make a completely new pastebin of your current extensions & CLI output, and describe the new "problem". |
| 12:23.34 | bartpbx | did anyone ever test voicemail with odbc storage in 1.4ß |
| 12:23.34 | tapic | I got "timeout waiting for response to challenge" message while connecting to AMI.. |
| 12:23.52 | bartpbx | it looks like there is somesting strange |
| 12:24.13 | yangvnc | http://www.pastebin.sk/en/7163/ |
| 12:25.57 | bartpbx | i always get an "SQL Get Data error!" |
| 12:26.09 | bartpbx | and i dont understand the code, when this can happen |
| 12:26.27 | bartpbx | on out 1.2 maschines the same storage works fine |
| 12:26.43 | yangvnc | [TK]D-Fender: http://www.pastebin.sk/en/7162/ here are all extensions, i wonder why 041710598 does not ring |
| 12:26.43 | *** join/#asterisk tryzor (n=tryzor@c-68-82-126-151.hsd1.pa.comcast.net) |
| 12:26.58 | bartpbx | and i can't find anythin in the doc that there is an change inthe config |
| 12:27.02 | tapic | <PROTECTED> |
| 12:27.37 | tapic | is the message I get from asterisk CLI when I try to connect with external computer |
| 12:28.23 | yangvnc | [TK]D-Fender: and i think that i am missing some syntax in extensions, i just can't notice where |
| 12:29.43 | [TK]D-Fender | yangvnc: CLI please... |
| 12:35.08 | yangvnc | http://www.pastebin.sk/en/7163/ |
| 12:35.17 | cristina_crow | quick question..pls: is there any limitation in the size of the RSA key for asterisk with TLS? |
| 12:44.06 | cristina_crow | humm |
| 12:44.31 | cristina_crow | I have created an RSA key, no passphrase, 1024 bit long (2048 bit long was not accepted) |
| 12:44.44 | cristina_crow | and put it in /var/lib/asterisk/keys |
| 12:44.45 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 12:44.48 | cristina_crow | now it accepts the public key |
| 12:45.02 | cristina_crow | (I have extracted the key from teh certificate) |
| 12:45.07 | cristina_crow | but, when I reload SIP |
| 12:45.09 | cristina_crow | I still get |
| 12:45.20 | cristina_crow | SSL cert error </etc/ssl/certs/asterisk.pem> |
| 12:45.23 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 12:45.24 | cristina_crow | :( any ideas? |
| 12:46.13 | Corydon76-dig | Is Asterisk running as root or as a regular user? |
| 12:46.40 | Corydon76-dig | If as a regular user, you might want to check permissions |
| 12:47.21 | Mike8861 | has anyone been able to implement wideband codec on asterisk ? |
| 12:47.47 | Corydon76-dig | Mike8861: G722 is available in 1.6 |
| 12:48.18 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:48.40 | Mike8861 | thanks for the prompt @Corydon76-dig |
| 12:48.57 | Mike8861 | i cannot get G722, coz xlite or eyebeam doesnt support it ? |
| 12:49.11 | Mike8861 | any client recommand? |
| 12:49.25 | Corydon76-dig | Polycom phone? |
| 12:49.51 | cristina_crow | asterisk runs as root |
| 12:49.53 | Corydon76-dig | The 550 and 650 both support wideband codecs |
| 12:50.18 | Mike8861 | uhmm, i will try it out! thanks |
| 12:51.44 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
| 12:51.52 | cristina_crow | permissions are ok |
| 12:54.31 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 12:56.59 | [TK]D-Fender | yangvnc: Ok... whats the error in here? |
| 12:57.34 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:00.03 | yangvnc | [TK]D-Fender: the error is that i dont receive the call to detel/041710598 |
| 13:00.20 | [TK]D-Fender | yangvnc: I don't even see WHY its being called. |
| 13:00.44 | yangvnc | [TK]D-Fender: doesnt it show out of frontdesk? |
| 13:01.01 | [TK]D-Fender | yangvnc: Wait, ok, I see 2 calls mixed in there |
| 13:01.58 | *** part/#asterisk cristina_crow (n=cvintila@212.146.94.66) |
| 13:02.21 | [TK]D-Fender | yangvnc: Ok, looks like you are calling yourself through your ITSP. Does an outside call work ok? |
| 13:03.09 | yangvnc | yes, i can call 041710598 over detel frommy softphone |
| 13:04.47 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:05.35 | jaytee | anyone know how many watts a Polycom IP330 uses for POE? |
| 13:05.46 | Great_Anta_Baka | has anyone used sip version 3 on a polycom 650, mine keeps crashing since i updated it |
| 13:06.21 | *** join/#asterisk Inez (n=faceoff@devel4.net) |
| 13:06.24 | Inez | Hello! |
| 13:07.17 | bartpbx | helllo |
| 13:07.24 | jaytee | hello |
| 13:07.31 | Mike8861 | hi there! |
| 13:07.37 | Mike8861 | welcome to #asterisk |
| 13:08.23 | *** join/#asterisk tompaw (n=tompaw@gw.krakow.tompaw.net) |
| 13:08.23 | *** join/#asterisk Arsenick- (n=auid@modemcable026.33-70-69.static.videotron.ca) |
| 13:08.27 | tompaw | hello |
| 13:08.36 | tompaw | how can I access "advanced options" menu? I can't find it anywhere in my GUI. |
| 13:08.41 | [TK]D-Fender | yangvnc: Then you have a SIP issue, not a dialplan one. |
| 13:08.48 | tompaw | I've checked the forums and everything, but no answer found there :< |
| 13:09.02 | [TK]D-Fender | tompaw: GUIs are not supported in this channel. please see the channel topic for relevent links |
| 13:09.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.33 | tompaw | [TK]D-Fender: I'm sorry, it was too long and I missed that part. |
| 13:11.07 | Great_Anta_Baka | #asteriskgui |
| 13:11.15 | Great_Anta_Baka | #asterisknow |
| 13:11.16 | Great_Anta_Baka | soz |
| 13:11.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.03 | Arsenick- | hi guy's, I have a problem with and old version of asterisk and I can't upgrade so I would like to knows if someone of you can give me a clue on what I need to do or where's my problem.. When someone call the call enter in the system as usual, but the caller still ringing after few ring asterisk write: "WARNING[21691]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 10" ... I've search on google |
| 13:12.03 | Arsenick- | with this error and I didn't find an interesting solution.. |
| 13:12.19 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 13:13.22 | [TK]D-Fender | Arsenick-: how "old"? |
| 13:14.16 | Arsenick- | 1.2.12.1 |
| 13:14.34 | Arsenick- | we will update this server in few weeks but for now he make me crazy.. |
| 13:15.38 | [TK]D-Fender | Arsenick-: Ok... ancient. Don't bet on much help without upgrading. |
| 13:16.16 | Arsenick- | yup I know.. maybe it's a bug but I can't find out why this started last week.. everything was fine before lol.. |
| 13:17.11 | Arsenick- | is it possible that it's an hardware issue ? |
| 13:17.32 | Corydon76-dig | Arsenick-: No |
| 13:18.16 | Arsenick- | ok |
| 13:18.37 | Corydon76-dig | 1.2.12.1 == completely unsupported |
| 13:19.27 | brad_mssw | Arsenick-: any reason you can't at least upgrade to the latest 1.2 series, even if you can't go 1.4? |
| 13:19.31 | Arsenick- | yeah.. I think we're gonna update sooner :p |
| 13:20.12 | *** join/#asterisk xenonex (n=xenonex@89.218.234.49) |
| 13:21.00 | Arsenick- | the hole server is a trash.. I don't want to touch anything on this, there's a lot of compiled stuff etc.. few ppl have worked on this server and no one had asterisk knowledge... guess the number of weird problem I got in the last week :P |
| 13:21.44 | Arsenick- | the new server is ready so I think I will switch this week end.. |
| 13:22.24 | *** join/#asterisk bjartis (n=test@195.1.73.1) |
| 13:22.25 | Arsenick- | thx for your help guy's |
| 13:22.27 | *** part/#asterisk bjartis (n=test@195.1.73.1) |
| 13:23.49 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:26.08 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:27.30 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 13:28.50 | x86 | morning |
| 13:30.35 | *** part/#asterisk tryzor (n=tryzor@c-68-82-126-151.hsd1.pa.comcast.net) |
| 13:42.15 | *** join/#asterisk oilinki (n=oil@ppp-124-122-182-115.revip2.asianet.co.th) |
| 13:47.15 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:48.04 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:48.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:50.06 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:50.10 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 13:53.29 | sumasuma | morning |
| 13:53.54 | lmadsen | morning |
| 13:55.26 | sumasuma | Is there is any information in Setting up, Configuring and Maintaining high volume asterisk servers ? |
| 13:55.46 | sumasuma | I have setup asterisk servers on individual machines |
| 13:56.10 | sumasuma | is the setup different for high volume asterisk servers or any tweaks need to be done ? |
| 13:56.36 | *** join/#asterisk [Akemi] (n=akemi@69-196-132-208.dsl.teksavvy.com) |
| 13:56.46 | lmadsen | sumasuma: there is a presentation I did at www.leifmadsen.com, but other than that, no, because it's a pretty complex thing to setup, and not many people have really done it |
| 13:56.55 | *** join/#asterisk neurosys0 (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 13:57.01 | lmadsen | but really the trick is to spread the load out among multiple box |
| 13:57.25 | sumasuma | What is the best way to spread the load ? |
| 13:57.43 | Great_Anta_Baka | lief can you tell me what you would use a high volume asterisk server for? |
| 13:58.00 | lmadsen | sumasuma: I use func_odbc + mysql + DUNDi |
| 13:58.05 | gr0mit | erm, high volumes of calls?! |
| 13:58.07 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:58.07 | lmadsen | Great_Anta_Baka: I don't understand the question |
| 13:58.15 | Great_Anta_Baka | i mean what kind of setup |
| 13:58.25 | Great_Anta_Baka | hotels? |
| 13:58.26 | Great_Anta_Baka | call centres? |
| 13:58.28 | gr0mit | big pbx |
| 13:58.35 | lmadsen | Great_Anta_Baka: call centres and ITSPs |
| 13:58.38 | Great_Anta_Baka | how big are we talking? |
| 13:58.42 | gr0mit | hosted voip server |
| 13:58.42 | Great_Anta_Baka | i see |
| 13:58.43 | lmadsen | that's what I've built -- I usually do clustering |
| 13:58.48 | sumasuma | Great_Anta_Baka: it is for broadcasting songs and informations to the users |
| 13:58.52 | lmadsen | and vPBXs |
| 13:58.57 | sumasuma | and receive calls and record them |
| 13:59.00 | Great_Anta_Baka | so not really the amount of users but throughput? |
| 13:59.03 | sumasuma | and play around with that |
| 13:59.16 | lmadsen | Great_Anta_Baka: amount of users means nothing -- it's the number of simultaneous calls that matters |
| 13:59.35 | yangvnc | hello lmadsen |
| 13:59.35 | Mike8861 | hello all. |
| 13:59.35 | gr0mit | well, Great_Anta_Baka it is more limited by call throughput and if you transcode, by CPU power |
| 13:59.36 | Great_Anta_Baka | k |
| 13:59.47 | sumasuma | simultaneous calls I presume it is limited to 240 calls per box |
| 14:00.01 | sumasuma | no transcoding |
| 14:00.02 | lmadsen | sumasuma: there is no limit other than what the box can handle and what your testing shows |
| 14:00.03 | Great_Anta_Baka | that sounds like a beast of a machine |
| 14:00.14 | lmadsen | Asterisk does not have a built in limit |
| 14:00.20 | sumasuma | yes |
| 14:00.28 | sumasuma | but the safe limit in my experience is 240 |
| 14:00.35 | lmadsen | depends on the box, and what you're using |
| 14:00.39 | sumasuma | that is true |
| 14:00.40 | lmadsen | I've gotten 500 calls before |
| 14:00.41 | Mike8861 | i have trouble implement IM with asterisk 1.4, can any one help ?? |
| 14:00.58 | lmadsen | does use instant messaging with Asterisk |
| 14:01.04 | Great_Anta_Baka | if i have a four port pri interface installed wat kind of cpu am i looking at.. i know there are other requirements,,eg hdd, ram, etc |
| 14:01.29 | lmadsen | Mike8861: ask a question of a specific problem and you'll get more answers, not just, "please setup my <xyz_problem> for me" |
| 14:01.46 | sumasuma | Is there is any software to maintain group of asterisk servers ? |
| 14:01.46 | Great_Anta_Baka | 500 calls simultaneous????? wtf??? |
| 14:01.59 | lmadsen | Great_Anta_Baka: I'd just use a Dell 2950 for that, or some other equivalent |
| 14:02.08 | lmadsen | sumasuma: doubt it |
| 14:02.15 | lmadsen | I've never heard of any |
| 14:02.52 | Mike8861 | uhmm....actually, I need some guide for this. |
| 14:02.55 | Great_Anta_Baka | i have had problems with my dell nic conflicting with my pri card.. but i guess thats cos i am using asterisk 1.2 |
| 14:03.12 | sumasuma | What other factors i need to consider in maintaining high volume asterisk servers ? |
| 14:03.17 | lmadsen | Mike8861: I don't think you'll find one |
| 14:03.20 | Mike8861 | i got the Patch for IM, but not sure how to patch it to asterisk to make IM work |
| 14:04.02 | lmadsen | Mike8861: 1) you need the source code 2) cd /usr/src/asterisk-src ; patch -p0 < /path/to/patch 3) recompile / reinstall 4) test |
| 14:04.06 | mvanbaak | Mike8861: patch -p0 < path_to_file |
| 14:04.11 | mvanbaak | lmadsen: lol |
| 14:04.12 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 14:04.21 | lmadsen | mvanbaak: I'm better than you ! :) |
| 14:04.26 | Uatec | hello there |
| 14:04.33 | Mike8861 | thanks for helping |
| 14:04.37 | mvanbaak | lmadsen: suuuuuuuuuuuuuuuuuuuuuuure |
| 14:04.45 | lmadsen | mvanbaak: that's what the girls say anyways |
| 14:04.52 | Great_Anta_Baka | lol |
| 14:04.52 | Mike8861 | my production server is install from ISO image, so there is no source code involed |
| 14:05.06 | lmadsen | Mike8861: use "Real Asterisk(tm)" |
| 14:05.12 | Great_Anta_Baka | well you gonna have to yum it then |
| 14:05.16 | mvanbaak | lmadsen: who's the one living alone without wife/gf etc ? |
| 14:05.18 | mvanbaak | it's not me |
| 14:05.26 | lmadsen | mvanbaak: I didn't say I didn't have girls over :) |
| 14:05.32 | Mike8861 | whats Real Asterisk(TM).... |
| 14:05.33 | mvanbaak | true |
| 14:05.36 | Uatec | lmadsen, but you don't, do you ... :P |
| 14:05.49 | lmadsen | Mike8861: a fake product I just made up -- you have to download the source then -- stop using Trixbox |
| 14:05.54 | sumasuma | Mike8861: the code from the digium svn |
| 14:06.00 | Mike8861 | oh. |
| 14:06.01 | Great_Anta_Baka | trixbox FTLLLL |
| 14:06.08 | Great_Anta_Baka | its the bane of my existence |
| 14:06.10 | lmadsen | Uatec: but I do :) |
| 14:06.19 | Uatec | :P |
| 14:06.26 | mvanbaak | ~trixbox |
| 14:06.27 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 14:06.42 | sumasuma | If I need to send 10,000 calls to 10 servers playing a file, what is the efficient way to do ? |
| 14:06.50 | Mike8861 | well, can I compile patch on Real Asterisk(TM) and copy the copiled file to trixbox ?? |
| 14:06.53 | Great_Anta_Baka | asteriskgui just has to implement pin sets and fail over trunks and then i can freepbx off this box |
| 14:06.57 | Mike8861 | i am not sure, since I am new to this |
| 14:07.01 | Great_Anta_Baka | take * |
| 14:07.03 | lmadsen | sumasuma: OpenSER in front with the distribution module to multiple asterisk servers |
| 14:07.30 | Uatec | see, i have a problem, i have sip phones connected to my asterisk box with a B410P ISDN 2 card in it, but when I ring another system, the other system can very rarely recognise my DTMF tones |
| 14:07.31 | mvanbaak | Mike8861: I have no idea. I think not because it's not stock asterisk to start with |
| 14:07.37 | Uatec | i've even tried to dial back on myself |
| 14:07.43 | mvanbaak | Mike8861: ask that question in #trixbox |
| 14:07.49 | lmadsen | Mike8861: like I said "Real Asterisk(tm)" is not a real product... I was just trying to emphasize that you should build Asterisk from source, and build your own system, as we don't support products like Trixbox in here |
| 14:07.52 | Uatec | (bizarrely, we dial out via a sip provider and in via isdn" |
| 14:07.58 | *** join/#asterisk spokra (n=spokra@gumby.sea0.speakeasy.net) |
| 14:08.16 | lmadsen | Mike8861: basically, with trixbox, you don't modify it -- you get what you get with it, and if you need something beyond that, you have to build a real Asterisk system from source and configure it |
| 14:08.38 | sumasuma | lmadsen: Why one need OpenSER? With Asterisk Manager API is it not easy to distribute ? |
| 14:08.52 | lmadsen | sumasuma: are you placing the calls, or sending the calls? |
| 14:08.59 | Mike8861 | thanks, @lmadsen. |
| 14:09.11 | lmadsen | sumasuma: errr... s/sending/receiving/ |
| 14:09.13 | sumasuma | I'm originating calls from the box to the users |
| 14:09.19 | lmadsen | sumasuma: then use the AMI, yes |
| 14:09.31 | lmadsen | just triggers calls on multiple systems... nothing too hard about that |
| 14:09.47 | sumasuma | what is the best way to transfer the file to each system ? |
| 14:09.52 | lmadsen | sumasuma: scp |
| 14:09.53 | sumasuma | for the 10,000 calls |
| 14:10.07 | lmadsen | goes off to do breakfast, shower, then paid work |
| 14:10.09 | sumasuma | ooh one cannot do that manually all the time |
| 14:10.19 | sumasuma | any best automated way ? |
| 14:10.35 | x86 | lmadsen: mmmmm paid work ;) |
| 14:10.47 | lmadsen | that sounds like a problem to solve outside of asterisk using the tools available in your distro |
| 14:10.52 | lmadsen | ~bestquestions |
| 14:10.54 | lmadsen | ~best |
| 14:10.55 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 14:11.03 | x86 | gah... wtf.... why is AT&T trying to charge me $0.46 per minute, when I have an ABN contract that locks us in at $0.0293/min |
| 14:11.04 | lmadsen | lol |
| 14:11.08 | Uatec | urethra franklin |
| 14:11.17 | lmadsen | Men Without Hats! |
| 14:11.25 | lmadsen | actually... George Thorogood |
| 14:11.28 | jaytee | Safety Dance |
| 14:11.36 | x86 | lmadsen: who do you love? |
| 14:11.43 | lmadsen | safely leaves the room |
| 14:11.59 | jaytee | he doesn't love anyone cuz he's "bad to the bone" (ducks) |
| 14:12.00 | sumasuma | i mean to transfer the file from application to individual asterisk machine |
| 14:12.12 | x86 | was only quoting george thorogood ;) |
| 14:12.17 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:12.23 | Great_Anta_Baka | haha |
| 14:12.24 | *** join/#asterisk riddlebox (n=james@adsl-66-143-58-253.dsl.ksc2mo.swbell.net) |
| 14:13.55 | jaytee | I'm looking at various solutions for POE to Polycom phones. Any recommendations for midspan hubs/injectors or POE equipped switches? any brands I should avoid like the plague? |
| 14:14.59 | Uatec | netgear have a good range of PoE switches |
| 14:16.26 | [TK]D-Fender | jaytee: D-Link's have worked great for me and everyone who've bought them on my reocmmendation. |
| 14:16.40 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 14:16.40 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:17.14 | x86 | [TK]D-Fender: I like HP's stuff... considerably cheaper than Cisco, but still same level of manageability |
| 14:17.27 | x86 | web management is just ass, imho |
| 14:17.27 | jaytee | [TK]D-Fender, hmmm they also seem to offer the lowest price per port I've found so far. |
| 14:17.39 | [TK]D-Fender | x86: Yes, HP is super rock-solid stuff, but still pricey |
| 14:18.23 | x86 | [TK]D-Fender: if you're looking for something with cisco-level functionality and manageability, but you don't have a cisco budget, HP is the best |
| 14:18.36 | [TK]D-Fender | x86: Fair enough. |
| 14:18.48 | mvanbaak | sumasuma: why use a call file ? best to use the AMI |
| 14:22.44 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 14:23.16 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 14:23.45 | [T]ank | once again, I have another T1 card that has come available. This one is a few months older than the last one so I will let it go for less. I have a sangoma a104D. I will let this one go for $900. Anyone interested? |
| 14:24.21 | rwaite | hello, all. i am having an issue with an iax connection and i believe i may know what is causing it. if i reboot the asterisk machine or run /etc/init.d/asterisk restart the iax provider will not register. i can run iax2 reload and it will then register, but after every "restart" i have to do that |
| 14:24.49 | rwaite | i also get a message that there is no iax provisioning configured, iax provisioning disabled - could this be the cause? |
| 14:25.33 | Uatec | it could be |
| 14:25.40 | Uatec | do something about it and see if it fixes it |
| 14:25.58 | rwaite | my problem is i dont know how to setup iax provisioning or even what that is |
| 14:26.10 | rwaite | voip-info doesnt have anything that really explains it (that i could find) |
| 14:26.25 | *** join/#asterisk Great_Anta_Baka (i=c4219f6b@gateway/web/ajax/mibbit.com/x-33fa0f99bd8d0cd4) |
| 14:26.44 | [TK]D-Fender | rwaite: thats only for the IAXY |
| 14:27.05 | Uatec | dies at the mention of IAXY |
| 14:27.22 | rwaite | hmm. okay thanks. any thoughts on why it will not "automatically" register on start up> |
| 14:28.00 | Uatec | how is the device supposed to know that the asterisk server has just restarted? |
| 14:28.23 | Uatec | hmm, actually, is this a device registering with asterisk via iax |
| 14:28.29 | rwaite | um what? the iax config has the auth details |
| 14:28.34 | Uatec | or asterisk registering with something else via iax? |
| 14:28.38 | rwaite | no, this is my asterisk machine registering with my iax provider |
| 14:28.41 | rwaite | sorry if i wasnt more clear |
| 14:29.14 | rwaite | if i do a iax2 show peers, it lists being offline but iax2 reload will consistently correctly register and from then (until the next restart) will work great |
| 14:29.50 | [TK]D-Fender | rwaite: You could try changing your register interval |
| 14:30.47 | rwaite | would that be register=\d* instead of yes? |
| 14:33.10 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 14:34.04 | hsv-al | . |
| 14:36.31 | Mike8861 | exit |
| 14:36.42 | [TK]D-Fender | rwaite: .... huh? |
| 14:37.26 | rwaite | i dont see anything for setting a reg interval. i see one for qualify/ok/notok, but the only other thing is maxregexpire |
| 14:37.32 | rwaite | is that what i'm lookin for? |
| 14:37.54 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:37.54 | *** join/#asterisk ManxPower (n=manxpowe@85.sub-70-222-247.myvzw.com) |
| 14:39.18 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 14:40.03 | [TK]D-Fender | rwaite: Sounds relevent, no? |
| 14:40.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:41.16 | rwaite | well, sort of, but wouldnt that handle expiration of an already registered channel |
| 14:42.11 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 14:45.32 | *** join/#asterisk JenniferAkemi (n=akemi@69-196-132-208.dsl.teksavvy.com) |
| 14:46.49 | *** join/#asterisk dieno (i=771e600b@gateway/web/ajax/mibbit.com/x-eeba0847b61715a5) |
| 14:47.07 | dieno | does any one have experience in Festival |
| 14:47.57 | J4zen | Does anyone remember an calculator that would tell you how many concurrent calls and such you can run on a specified system? |
| 14:48.55 | dieno | [TK]D-Fender u there ? |
| 14:49.07 | puzzled | J4zen: have never seen one. only calculator that tells you how many calls you can run over a certain link with a certain codec |
| 14:49.42 | J4zen | Do you remember that one puzzled? |
| 14:49.58 | puzzled | J4zen: not sure but check asteriskguru |
| 14:50.09 | J4zen | i know there is a calculator as i described, it's been a while tho = |
| 14:50.11 | J4zen | =\ |
| 14:50.17 | J4zen | will do, thanks. |
| 14:50.40 | [TK]D-Fender | dieno: ... I love lobster! |
| 14:50.53 | tzafrir_laptop | J4zen, look for: voip bandwidth calculator |
| 14:51.03 | dieno | :( this time i am really stucked :'( |
| 14:51.11 | J4zen | thanks tzafrir_laptop, long time no see |
| 14:51.26 | dieno | [TK]D-Fender festival is speaking like robot and breakin up |
| 14:51.33 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:51.40 | tzafrir_laptop | jbot, tell dieno about ask |
| 14:51.55 | [TK]D-Fender | puzzled: c:\Windows\System32\calc.exe |
| 14:52.17 | [TK]D-Fender | dieno: I've never used it, and its rude to target people randomly with wuestions like that. |
| 14:52.22 | tzafrir_laptop | bash: c:WindowsSystem32calc.exe: command not found |
| 14:52.35 | dieno | oops :D |
| 14:52.39 | [TK]D-Fender | tzafrir_laptop: kill-joy |
| 14:52.48 | puzzled | [TK]D-Fender: I would not know. Haven't used windows since win98 :) |
| 14:53.27 | Rico29 | hi |
| 14:53.38 | Rico29 | I know it's doesn't concern directly asterisk |
| 14:53.53 | Rico29 | but I have trouble with a cisco registering on my * |
| 14:54.19 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 14:54.22 | Rico29 | the cisco registers with the line1_name |
| 14:54.48 | Rico29 | and not with line1_authname and line1_password |
| 14:55.15 | Rico29 | does anybody knows why ? |
| 14:55.30 | Rico29 | ( with autocreatepeer=yes in sip.conf ) |
| 14:55.40 | rwaite | hmm. removing host=dynamic from iax.conf seemed to fix it |
| 14:56.31 | Rico29 | huh ? |
| 14:56.49 | Rico29 | there's not IAX in my conf ! |
| 14:56.52 | Rico29 | no |
| 14:56.57 | rwaite | disregard |
| 14:57.10 | [TK]D-Fender | Rico29: And don't assume everybody is talking to YOU :p |
| 14:57.23 | Rico29 | mmh sorry |
| 14:57.41 | rwaite | cool. [TK] thanks for your help |
| 14:57.41 | Rico29 | it was a really strange answer |
| 14:58.23 | *** part/#asterisk Porpes (n=sergio@189-19-2-118.dsl.telesp.net.br) |
| 15:01.10 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 15:01.40 | Rico29 | nobody can helpme ? |
| 15:01.48 | Rico29 | nobody can help me ? |
| 15:01.59 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 15:04.23 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 15:04.50 | tzafrir_laptop | Rico29, what "cisco" exactly? |
| 15:04.59 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 15:05.53 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 15:06.00 | gsiener | Hi all. I'm looking for a consultant to provide advice and guidance on setting up a system. I plan to build/configure, but would like an expert to weigh in. Thanks. |
| 15:06.22 | tzafrir_laptop | Rico29, generally a useful stanza for asking about a problem: "I did X, expected Y to happen but Z actually happened" |
| 15:07.50 | Rico29 | tzafrir_laptop> 7960 |
| 15:07.53 | Rico29 | with SIP firm |
| 15:08.40 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 15:08.41 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 15:09.08 | *** join/#asterisk ludan (n=daniele@192.167.215.122) |
| 15:09.36 | tzafrir_laptop | Does it successfully regisster? Or just tries to register? Why do you expect it to register with that specific name? |
| 15:09.45 | ludan | hi |
| 15:09.59 | tzafrir_laptop | a pastebin of your sip.conf may help to clarify things |
| 15:10.14 | *** join/#asterisk guilherme-jorge (n=guilherm@200.213.210.2) |
| 15:11.14 | ludan | i'd like to link my asterisk server with a toll free number so that user can call a local number and the call can be forwarded towards my server. is it possible? |
| 15:11.36 | Rico29 | tzafrir_laptop> it registers |
| 15:11.58 | Qwell | ludan: toll free or local? completely different things |
| 15:12.06 | Rico29 | but in autocreatepeer mode, imagine if two cisco registers whith the same id !?N |
| 15:12.25 | Rico29 | tzafrir_laptop> ok |
| 15:12.28 | ludan | Qwell: i'd like that people can callo cheaply from their landline, give me a hint please |
| 15:12.59 | Qwell | anybody in the US? anybody within a 12 mile radius? same state? |
| 15:13.14 | ludan | some people are in the US, some other in europe |
| 15:13.25 | Qwell | so none of the above |
| 15:13.31 | *** join/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 15:13.35 | Rico29 | tzafrir_laptop> http://debian.pastebin.com/m726b958a |
| 15:14.17 | ludan | Qwell: ? some of them are in US |
| 15:15.18 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:15.44 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 15:17.41 | tzafrir_laptop | Rico29, that is just the [general] section. Have you actually defined any sip peer entry? |
| 15:17.54 | tzafrir_laptop | [entry] |
| 15:18.07 | tzafrir_laptop | type=peer; or type=friend |
| 15:18.13 | tzafrir_laptop | host=dynamic |
| 15:18.25 | jjshoe | [07:55] <Rico29> ( with autocreatepeer=yes in sip.conf ) |
| 15:18.34 | Rico29 | no, the first time phones want to register, they use autocreatepeer |
| 15:18.39 | Rico29 | :) |
| 15:18.50 | Rico29 | and next, everything is in realtime |
| 15:18.53 | tzafrir_laptop | and you should use 'language=fr" rather than "lang=fr", but it really doesn't matter here |
| 15:19.01 | Rico29 | ok |
| 15:21.07 | Rico29 | I find something : by using Line1_shortname="Login" and Line1_name="345676543", the phone displays "Login" as line name and registers with 345676543/345676543 |
| 15:21.37 | Rico29 | it's not a good solve, but it seems to work... |
| 15:23.19 | Rico29 | now, the second part of my problem : exactly the same thing with a thomson ST2030S huhuhu |
| 15:26.01 | guilherme-jorge | Hello, I' ve a asterisk server and I want to install some Asterisk GUI (free), to create extensions, record calls... What is the most popular free asterisk GUI to do this? |
| 15:26.44 | ManxPower | guilherme-jorge: then go ask on #asterisk-gui |
| 15:27.03 | ManxPower | guilherme-jorge: we don't use GUIs here |
| 15:27.03 | guilherme-jorge | ManxPower: tks |
| 15:27.38 | ludan | any clue about toll free number to link with asterisk? |
| 15:27.42 | ludan | or landline... |
| 15:27.45 | ludan | please help me |
| 15:27.59 | ManxPower | ludan: Order one from the telco |
| 15:28.06 | ludan | :| |
| 15:28.18 | ManxPower | ludan: your question has no useful information |
| 15:28.30 | ManxPower | ~ask |
| 15:28.30 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:29.39 | ludan | ManxPower: i've asterisk 1.4.21 up and running and i can create a conference room to make people talking to each other using a sip client (ekiga). i'd like that the conference could be joined also by people calling from a landline |
| 15:30.56 | [TK]D-Fender | ludan: Go get a card to connect a physical line to *, or sign up with an ITSP |
| 15:30.59 | [TK]D-Fender | ~itsp |
| 15:31.00 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 15:32.24 | ludan | oh guys, i know it's funny to treat a person like an idiot, but I did remember that there was such a service available |
| 15:33.29 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:35.18 | [TK]D-Fender | ludan: Then why are you asking about "how to get a number"? |
| 15:35.28 | [TK]D-Fender | ludan: Since you now say you already know all of this? |
| 15:35.51 | ludan | [TK]D-Fender: i know it's possible, i do not know how to make it working! |
| 15:36.04 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 15:36.24 | [TK]D-Fender | ludan: Who have you signed up with? |
| 15:36.27 | Rico29 | OK, i fixed the problem for Cisco phones |
| 15:36.45 | Rico29 | but i still have the same pb with thomson ST2030S phones |
| 15:36.58 | ludan | [TK]D-Fender: it was a free service |
| 15:37.06 | ludan | but i've no clue |
| 15:37.11 | [TK]D-Fender | ludan: "it"? |
| 15:37.17 | ludan | i got this system by another guy |
| 15:37.32 | ludan | that no longer maintains the system itself |
| 15:38.22 | [TK]D-Fender | ludan: Ok, so what doesn't your ITSP do for you right now? |
| 15:39.03 | ludan | [TK]D-Fender: nothing it doesn't work anymore |
| 15:39.33 | [TK]D-Fender | ludan: And why is that? |
| 15:44.23 | *** join/#asterisk moy (n=moyhu@189.169.71.85) |
| 15:48.05 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:48.15 | x86 | does Asterisk 1.4 and 1.6 support IPv6? |
| 15:48.31 | x86 | I'd like to setup an IAX2 peer over IPv6 |
| 15:51.55 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:53.12 | *** join/#asterisk r0land (n=hellp@194.146.154.54) |
| 15:53.16 | r0land | hello all |
| 15:53.22 | [TK]D-Fender | x86: Planned for 1.6, but not final. |
| 15:53.29 | r0land | hey [TK]D-Fender |
| 15:53.34 | x86 | so it's not in 1.6 yet? |
| 15:53.41 | x86 | I'm running beta9 on one end |
| 15:54.07 | BBHoss | x86: http://www.asteriskv6.org/ |
| 15:54.21 | ajohnson | I was hoping I could get a sanity check. On an Asterisk system that does about 25k calls a day, after 10 days Asterisk is using 2.3gb of memory |
| 15:54.50 | x86 | BBHoss: hey thanks |
| 15:54.57 | *** join/#asterisk gitguy (n=diego@adsl-128-132.click.com.py) |
| 15:55.06 | r0land | i have 6 lines, with 6 sip.conf extensions directed to them, is there a way i could edit my extensions.conf so if a user dials a number, asterisk tries one of these lines after another till it finds a none occupied line to dial from ?! |
| 15:55.22 | gitguy | i just changed the codec to one of my users to gsm, and then he suddendly can't dial anymore |
| 15:55.25 | [TK]D-Fender | BBHoss: ML referred to that site as a complete re-write, not even a fork. |
| 15:55.29 | gitguy | he gets "person unavailable" on his phone |
| 15:55.32 | gitguy | wtf? |
| 15:55.42 | x86 | BBHoss: doesn't do me a lot of good though... |
| 15:55.43 | gitguy | then i put ulaw on his user and he can |
| 15:55.46 | x86 | BBHoss: "only SIP (and the relevant auxiliary libs: rtp, netsock,...) is ported to IPv6. IAX, h323, etc... are not yet ported to ipv6." |
| 15:55.47 | [TK]D-Fender | r0land: Yes. Just dial them back to back. |
| 15:56.08 | [TK]D-Fender | gitguy: Fix your peer |
| 15:56.18 | gitguy | [TK]D-Fender: what do you mean with "fix" |
| 15:56.30 | BBHoss | x86: its funny you mention ipv6, i've been trying all day to get ipv6 working with my talisman firmware router (WRT160N) |
| 15:56.41 | [TK]D-Fender | gitguy: if only the codec changes, then you aren't permitting the right ones. |
| 15:56.54 | r0land | [TK]D-Fender could u explain a bit more if u may |
| 15:57.10 | BBHoss | x86: i don't know alot about the asterisk code, but IPV6 shouldnt be |that| big of a deal to add |
| 15:57.14 | [TK]D-Fender | r0land: Go set your codes in your sip peer. |
| 15:57.29 | gitguy | [TK]D-Fender: i have different peers/users, i put disallow=all in all of them, some use different codecs (ulaw/alaw, gsm, speex, etc) |
| 15:57.33 | BBHoss | x86: you might try asterisk-dev and see if they have any comments |
| 15:57.43 | [TK]D-Fender | r0land: Sorry, bad aim |
| 15:57.57 | r0land | hmm |
| 15:57.59 | [TK]D-Fender | r0land: Dial one. then on your next priority, dial the NEXT one, etc |
| 15:58.04 | r0land | ah ok |
| 15:58.06 | r0land | i got it |
| 15:58.07 | r0land | thakn u :) |
| 15:58.19 | [TK]D-Fender | gitguy: PASTEBIN <- |
| 16:02.00 | gitguy | [TK]D-Fender: http://pastebin.com/m4159bd1e ... i had to change allow=gsm for allow=ulaw, otherwise he wouldn't be able to dial a conference (meetme) extension |
| 16:02.21 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:02.58 | [TK]D-Fender | gitguy: Show me CLI with sip debug and your config updated to gsm and non-functional. |
| 16:03.19 | x86 | BBHoss: heh, I just use DD-WRT on my WRT's and ipv6 works fine ;) |
| 16:03.29 | x86 | BBHoss: but then again, DD-WRT > Talisman ;) |
| 16:03.35 | BBHoss | x86: what tunnel do you have |
| 16:04.12 | *** part/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 16:05.07 | BBHoss | x86: and my router hasn't been tested very well from what i've seen. says something about "install rc7 mini first then std" what does that mean |
| 16:05.21 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:05.29 | ManxPower | BBHoss: I think that means "go to the channel with people that can help" |
| 16:05.49 | BBHoss | yeah, that might be a good idea |
| 16:06.04 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:06.13 | *** join/#asterisk gitguy (n=diego@adsl-128-132.click.com.py) |
| 16:08.08 | fogo | I've got some T1s bouncing on a REC alarm - anyone know what REC is? |
| 16:09.40 | Qwell | a typo of RED? |
| 16:10.36 | [TK]D-Fender | fogo: You've reached the end and have to flip the cassette to "Side B" |
| 16:10.37 | fogo | Qwell: I don't think so; they decided to go RED now, but they were bouncing between REC and OK |
| 16:10.48 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:10.58 | ManxPower | fogo: We've never seen a REC alarm |
| 16:11.37 | Qwell | didn't want help, I guess |
| 16:12.05 | Corydon76-dig | ManxPower: "REC" means Recovering |
| 16:12.32 | Qwell | so then there must've been some other type of alarm |
| 16:12.38 | Corydon76-dig | i.e. kind of OK, but hasn't had a full 2 seconds of OK yet |
| 16:12.45 | ManxPower | Cool |
| 16:12.59 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:13.01 | ManxPower | too bad he doesn't know the actual type of alarm |
| 16:13.18 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 16:13.35 | fogo | ... what a perfect time for ksirc to crash |
| 16:13.38 | Corydon76-dig | Well, he's unlikely to have had a BLUE alarm... |
| 16:13.52 | fogo | so, what did I miss? any ideas on the REC? |
| 16:13.53 | Qwell | well, if it's RED now.. |
| 16:13.57 | *** part/#asterisk bartpbx (n=bartpbx@217.24.210.202) |
| 16:14.06 | Corydon76-dig | fogo: REC means Recovering |
| 16:14.15 | fogo | ah, ok.. that makes sense |
| 16:14.35 | Corydon76-dig | fogo: No alarms detected, but there hasn't been a full 2 seconds of no alarms at that point in time |
| 16:14.45 | Qwell | Alarm: No alarm. |
| 16:14.49 | fogo | Qwell: yeah, I'm waiting on my provider |
| 16:15.09 | Corydon76-dig | fogo: was the alarm yellow or red beforehand? |
| 16:15.30 | fogo | Corydon76-dig: it was OK |
| 16:15.40 | fogo | then it kept bouncing between OK and REC |
| 16:15.47 | fogo | now they're both (yes 2) RED |
| 16:16.13 | Corydon76-dig | fogo: something is probably not agreeing on the signalling |
| 16:16.56 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 16:16.58 | fogo | Corydon76-dig: well, the funny thing is that one of the 3 (all in the same group, from the same provider) is still OK |
| 16:17.09 | *** join/#asterisk MrNaz (n=naz@ppp59-167-94-76.lns2.mel6.internode.on.net) |
| 16:17.09 | tzanger | f |
| 16:17.13 | jblack | I know this has very little to do with asterisk, but please call your congressman and ask them to support Dodd and Feingold's fillibuster to prevent FISA (that's the warrentless wiretap). It'll take 1 minute to google your representatives, and 1 minute on the phone. |
| 16:17.24 | jblack | s/man/men |
| 16:17.57 | fogo | jblack: didn't that already pass the house though? |
| 16:18.22 | jblack | Dodd and Feingold are fillibustering _right now_ it to prevent it from passing in the house. |
| 16:18.35 | Corydon76-dig | jblack: you mean Senate |
| 16:18.52 | Corydon76-dig | There's no such thing as a filibuster in the House |
| 16:19.23 | jblack | uh, shit. You're right. Dodd and Feingold are senators |
| 16:20.06 | Corydon76-dig | And a filibuster requires 40. 2 aren't adequate |
| 16:20.17 | jblack | 40 to block closure |
| 16:20.28 | Corydon76-dig | "cloture" |
| 16:20.30 | ManxPower | hands jblack a bottle with a picture of a flower on it and says "drink this", jblack drinks it, then I say "Oh shit, I gave you the bottle with the poison in it!" |
| 16:20.39 | ManxPower | moral of the story, being inaccurate is bad. |
| 16:22.35 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 16:22.50 | jblack | Moral of the story: Some people are so detail focused, that they'll screw themselves over in the big picture because of hurried pronouncement and typos. |
| 16:23.24 | *** join/#asterisk peterpen (n=fantasti@212.57.232.254) |
| 16:23.35 | RoyK | ManxPower: moral of story - someone always keeps bottled poison in times of great need |
| 16:24.22 | peterpen | Hey I've successfully made a bash AGI script and I can access $agi_callerid inside, but my dialplan has its own variable (e.g. $month) can I do something like $agi_month? or am I stuck passing the variables and using $1, $2, $3, etc... ? |
| 16:25.09 | jblack | Ok. Feingold is on CSPAN-2, starting the fillibuster now. |
| 16:27.00 | Corydon76-dig | There's actually still a chance for the lawsuit to proceed, even if the bill passes. All the court has to find is that the compromise unconstitutionally delegates a power reserved to the Congress to the executive branch. |
| 16:27.08 | peterpen | If anyone has any thoughts on that - I'll be right back, need to reboot |
| 16:27.58 | Corydon76-dig | and if the next president is Democratic, he may choose not to appeal that decision, thus letting the lawsuit stand |
| 16:28.49 | jblack | Dude, unless you like warrentless wiretaps, or don't live in the US, make the two calls. Tell me what state you live in, and i'll look up the phone numbers to your two senators. |
| 16:29.15 | Corydon76-dig | My two senators are a lost cause |
| 16:29.23 | Corydon76-dig | Alexander and Corker |
| 16:29.52 | jblack | Call anyways. Then get everyone you know to call too. |
| 16:30.03 | jblack | They get a few dozen calls, they'll get the point that their job is at risk. |
| 16:30.15 | Corydon76-dig | Their jobs aren't at risk |
| 16:30.49 | fogo | I'm afraid my senator's a lost cause even if his job isn't up - Hatch :P |
| 16:30.53 | jblack | Sigh. Ok. Anyone else that wants help with it? |
| 16:30.58 | jblack | Call him anyways! |
| 16:31.15 | Qwell | jblack: mine won't listen to me, I'm sure |
| 16:31.26 | Qwell | "Are you a registered voter?" "No sir." |
| 16:31.28 | jblack | They can't ignore you if you don't call? |
| 16:31.33 | jblack | Nope, Mine didn't ask that. |
| 16:31.41 | Qwell | read: I don't care. |
| 16:31.47 | jblack | You call the number, it takes 30 seconds, and they put another checkmark next to a line item. |
| 16:33.24 | jblack | fogo: 202-224-5251. All you need to say is "I'm a constituent of Orin Hatch. I do not support fisa and I do not want the fillibuster squashed". They might ask your address, they will say "I'll pass that along" |
| 16:33.41 | Qwell | say |
| 16:34.01 | fogo | jblack: wow.. you even looked up the number for me! I was planning on calling anyway; now I really will :) |
| 16:34.28 | *** join/#asterisk makkksimal (n=makkksim@e177213029.adsl.alicedsl.de) |
| 16:34.31 | Qwell | jblack: go ahead and call for me |
| 16:34.49 | [TK]D-Fender | I love how they keep trying to go back in time an legalize past crimes. |
| 16:35.41 | [TK]D-Fender | McCain is breaking campain finince laws and nobody started the lynch mob to drag his ass to court. |
| 16:36.22 | *** join/#asterisk nauticalthinker (n=mratliff@205.149.185.2) |
| 16:36.27 | [TK]D-Fender | Abuse of law is flaunted so much its scary. That kind of coverage should get people jailed instantly. |
| 16:38.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:38.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:39.18 | mog | Corydon76-dig, do you really think obama would revert said decision? |
| 16:39.19 | fogo | hmmm.. looks like my T1 is now Alarm status LB.. any idea what that is? |
| 16:39.42 | fogo | hmmm.. looks like my T1 is now Alarm status LB.. any idea what that is? |
| 16:39.50 | fogo | oops.. sorry :) |
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| 16:42.20 | Corydon76-dig | mog: not revert, just fail to appeal it |
| 16:42.22 | fogo | just got a call from my provider - apparently the two leased lines have problems on the line (their end :( ).. could be 4 hours :P |
| 16:44.56 | *** part/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 16:46.15 | x86 | LB generally means loopback |
| 16:47.53 | coreyf52 | does anyone know is it possible to build zaptel/dahdi into the kernel (not a module)? I want to disable kernel module loading. Also I thought the name was changed to dahdi, i can't find any dahdi release-is that svn only so far? |
| 16:50.50 | Corydon76-dig | coreyf52: It's theoretically possible, but it's not supported |
| 16:51.24 | Corydon76-dig | Why in god's name do you want to disable kernel module loading? |
| 16:52.24 | Qwell | Corydon76-dig: false sense of security |
| 16:52.52 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 16:53.10 | Corydon76-dig | Qwell: that was my thought, too. Or he's a Gentoo ricer who wants to squeeze all the performance he can get out of the boot process |
| 16:53.15 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 16:53.22 | coreyf52 | not for performance |
| 16:53.24 | jjshoe | linuxbios for the boot process++; |
| 16:53.26 | coreyf52 | for security |
| 16:53.39 | Qwell | mIRC. Not a ricer. |
| 16:53.55 | Corydon76-dig | coreyf52: whoever told you it was more secure was blowing smoke up your ass |
| 16:54.21 | Qwell | You need to be root to load a module. |
| 16:54.33 | Qwell | If you are root already, you can do *MUCH MORE* than just loading a malicious module. |
| 16:54.53 | coreyf52 | ok understood... |
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| 16:58.49 | sumasuma | If I need to send 10,000 calls to 10 asterisk servers playing a file, what is the efficient way to do ? |
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| 17:02.49 | Corydon76-dig | Ya think? |
| 17:03.31 | jaytee | it was either that or they reset something in the matrix. |
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| 17:08.41 | [TK]D-Fender | Corydon76-dig: On that FISA idea, correct me if I'm wrong, but once president Obama could not simply overturn the law, no? That'd require congress to change the law or for the judiciary branch to discard it as being non-constitutional or illegal for some other reason, right? |
| 17:08.41 | russellb | wanders around |
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| 17:12.53 | Rico29 | mmh ? |
| 17:12.53 | outtolunc | swears.. i didn't do that <G> |
| 17:12.53 | nauticalthinker | have any of you connected an asterisk server to an old fujitsu 9600? |
| 17:12.54 | nauticalthinker | what type of connection is recommended for this? |
| 17:12.54 | Corydon76-dig | outtolunc: tripped over the power cord again, eh? |
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| 17:12.54 | outtolunc | yeah .. too much coffee i flipped *all* the switches <G> |
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| 17:12.54 | outtolunc | evil grin |
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| 17:16.30 | [TK]D-Fender | Corydon76-dig: So am I ont he right track? |
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| 17:17.46 | *** join/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 17:17.46 | _MrSeb_ | Hi to all |
| 17:18.04 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:20.11 | _MrSeb_ | [TK]D-Fender: are you present? |
| 17:20.57 | [TK]D-Fender | _MrSeb_: Yes. |
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| 17:21.19 | nauticalthinker | maybe you guys missed my last post... |
| 17:21.40 | [TK]D-Fender | nauticalthinker: Doubtful anyone has. |
| 17:21.55 | nauticalthinker | ... |
| 17:21.57 | nauticalthinker | ok |
| 17:22.12 | [TK]D-Fender | nauticalthinker: Best way would be in order : PRI, SIP, Analog. |
| 17:22.13 | nauticalthinker | to you have an idea then? |
| 17:22.38 | [TK]D-Fender | nauticalthinker: NO "idea" to look for. Connect it via the best quality tech it supports. |
| 17:22.39 | nauticalthinker | so use t1 card to connect to old pbx system? |
| 17:22.59 | [TK]D-Fender | nauticalthinker: T1 would be a solid standard (usually) link |
| 17:23.24 | nauticalthinker | I'll just have to research this box to find the best approach I suppose |
| 17:23.42 | _MrSeb_ | [TK]D-Fender: I've done some trials, but without success... a thing probably wrong in my configuration is that every sip provider must be with nat=yes, so in debug when it try to connect to external appear ad NAT |
| 17:24.12 | Maliuta | _MrSeb_: is your * box behind a NAT? |
| 17:24.22 | _MrSeb_ | Maliuta: yes |
| 17:24.22 | [TK]D-Fender | _MrSeb_: I've told you what to do several times now. |
| 17:24.56 | _MrSeb_ | [TK]D-Fender: yes, but yesterday you've told to put nat on no for sip provider |
| 17:25.10 | Maliuta | _MrSeb_: I am guessing [TK]D-Fender has pointed you at the SIP howto, read it |
| 17:25.11 | [TK]D-Fender | _MrSeb_: and indeed you never should |
| 17:25.21 | Maliuta | SIP+nat is not that hard |
| 17:25.36 | gitguy | [TK]D-Fender: why do i have to enable sip debug for the codec change... i know for "debugging it" and see whats wrong with it, but isn't this something asterisk should handle "just fine" ? |
| 17:25.45 | _MrSeb_ | Maliuta: I think so before put a router |
| 17:25.45 | gitguy | [TK]D-Fender: i'm talking about the codec change from ulaw to gsm |
| 17:25.59 | [TK]D-Fender | gitguy: ...HUH? |
| 17:26.12 | Maliuta | is with [TK]D-Fender |
| 17:26.13 | [TK]D-Fender | gitguy: pastebin the damned failed call & its associated config |
| 17:26.24 | [TK]D-Fender | gitguy: I asked for this hours ago. |
| 17:28.23 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 17:28.25 | gitguy | it wasn't from my endpoint that failed... it works here with ulaw... but it doesn't appear to work on windows with xlite, i'll try later |
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| 17:29.16 | _MrSeb_ | Maliuta: I've and asterisk server and client behind nat, general with externalhost, nat=yes and conreinvite=no, sip provider with nat=yes and canreinvite=no... registration is good, but when I call my number nothing, in debug no packet are displayed |
| 17:31.21 | _MrSeb_ | Maliuta: the asterisk server is setted as dmz, so all port are redirected |
| 17:31.26 | *** join/#asterisk DSpair (n=D-Spare@163.muaa.syrc.chcgil24.dsl.att.net) |
| 17:31.32 | DSpair | Hello all... |
| 17:31.52 | outtolunc | howdy |
| 17:31.56 | DSpair | I have a problem that my vendor cannot solve, and I'm hoping that perhaps someone here might be able to point me in a good direction. |
| 17:32.12 | Corydon76-dig | North is a good direction |
| 17:32.22 | outtolunc | was gonna say 'go west young man' |
| 17:32.32 | [TK]D-Fender | _MrSeb_: Go west! life is peaceful there! |
| 17:32.36 | DSpair | I have an IBM 2U server (Dual Intel Xeon Dul Core proc, 2GB RAM, Mirrored SCSI Disks). |
| 17:33.03 | DSpair | I am running Asterisk 1.4.19. |
| 17:33.33 | DSpair | From time to time, when one of our users calls into the voicemail app, the SIP channel gets hung with lastmsg of BYE. |
| 17:33.55 | Strom | [TK]D-Fender: in the open air? |
| 17:34.00 | [TK]D-Fender | DSpair: does the call show up under a straight "core show channels"? |
| 17:34.04 | DSpair | When this happens, the PBX Load starts to creep up until it steadies at well over 2.0 |
| 17:34.33 | DSpair | [TK]D-Fender, I don't know for sure. I restarted Asterisk last night and it hasn't happened again yet. |
| 17:34.43 | DSpair | It does show up under "sip show channels" |
| 17:34.47 | [TK]D-Fender | Strom : indeed! |
| 17:35.06 | Corydon76-dig | DSpair: recompile with DEBUG_THREADS and DONT_OPTIMIZE. If you still get the problem, obtain the output of 'core show locks' |
| 17:35.08 | [TK]D-Fender | DSpair: then forget about it. jsut a liginering hint of a call.. |
| 17:35.10 | _MrSeb_ | [TK]D-Fender: if you explain to me what to try I do it... this night I've tryed a lot of thing, but without success, I0ve corrected only the last thing you have told to me with debug info, now debug info seems correct, but now the client don't ring... |
| 17:35.38 | [TK]D-Fender | _MrSeb_: and I showed you that you were passing your INTERNAL address, therefor you've messed seomthing up |
| 17:36.01 | DSpair | Corydon76-dig, AHA!!! I knew there had to be a way to turn on debugging!! |
| 17:36.14 | DSpair | Thanks so much!!! |
| 17:36.26 | [TK]D-Fender | _MrSeb_: So again, pastebin your config and a failed call. Then remove the registry setting from aAstDB, reload SIP and cath the register attempt |
| 17:36.28 | Corydon76-dig | DSpair: 'make menuselect' |
| 17:37.07 | DSpair | Corydon76-dig, I can't right now as this is a production PBX which is in use by over 70 users. |
| 17:37.46 | Corydon76-dig | DSpair: unlike on Windows, you're free to reinstall while the process is running. |
| 17:38.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:38.22 | DSpair | Corydon76-dig, I know that I can compile and install the binaries without a restart, but I cannot restart to put the new application in place until tonight, so I will just wait and do it all at once. |
| 17:38.24 | [TK]D-Fender | DSpair: and then issue a "restart when convenient" |
| 17:38.37 | Maliuta | _MrSeb_: pastebin your config and I'll have a look |
| 17:38.44 | [TK]D-Fender | DSpair: It'll automaticall restart when all channels have cleared |
| 17:38.49 | DSpair | [TK]D-Fender, That will take hours. Our call volume is MASSIVE! |
| 17:38.57 | Corydon76-dig | DSpair: and if you have trouble after we've all gone to bed.... |
| 17:39.17 | DSpair | There's ~4800 calls per day, averaging 12 minutes. |
| 17:39.27 | [TK]D-Fender | DSpair: Well its your circumstances. If you want to cut them off earlier, thats your choice. I jsut gave you a "sit back and watch" option. |
| 17:39.48 | DSpair | Corydon76-dig, Well, I appreciate the concern, but I have to do what I feel it right for my network. |
| 17:40.02 | DSpair | One more quick question. Is anyone here running Asterisk on Ubuntu Gutsy? |
| 17:40.20 | DSpair | Or have any reason to think that would be sub-optimal? |
| 17:40.28 | [TK]D-Fender | DSpair: probably a bunch of people. |
| 17:40.40 | [TK]D-Fender | DSpair: Its linux. It'll run more or less like any other. |
| 17:40.52 | [TK]D-Fender | DSpair: Once everythings compiled its all the same. |
| 17:40.58 | [TK]D-Fender | (largely) |
| 17:41.06 | DSpair | [TK]D-Fender, I figured. My vendor keeps complaining about wanting to switch us to CentOS, but our entire network is Ubuntu and I would prefer to standardize on Ubuntu. |
| 17:41.23 | outtolunc | find a new vendor |
| 17:41.37 | DSpair | outtolunc, They're the only Digium provider for 200 miles. |
| 17:41.52 | [TK]D-Fender | DSpair: CentOS being a RHEL clone has a much larger user base, is less "bleeding edge", and by default comes with dev tools, etc, and uses more tried & true inits, etc. |
| 17:42.18 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 17:42.23 | [TK]D-Fender | DSpair: this does say something about the kind of environment you'll be working in, but if you're very comfortable managing Ubuntu, then by all means. |
| 17:42.28 | outtolunc | you shittin me, only 1 in chicago? |
| 17:42.33 | DSpair | Would it be better is I ran on Debian Sarge? or Woody? |
| 17:43.01 | Corydon76-dig | outtolunc: probably only one authorized reseller |
| 17:43.08 | outtolunc | ah |
| 17:43.13 | DSpair | I have long since lost my love for RedHat based distros and their crappy package management. |
| 17:43.35 | [TK]D-Fender | DSpair: Works great for most... |
| 17:43.45 | DSpair | outtolunc, My IP address shows Chicago, but I am not in Chicago. That's my DSL provider. |
| 17:43.56 | Corydon76-dig | outtolunc: Digium has been trying to make that title actually mean something about the quality of the provider, instead of just any fool who wants to buy parts at reseller cost |
| 17:44.09 | outtolunc | understood |
| 17:44.25 | outtolunc | <- is just a fool <G> |
| 17:44.33 | Strom | DSpair: "we demand that you switch distros" is usually a cover for "duh, we're clueless and don't want to learn" |
| 17:44.54 | DSpair | I went out on a limb to convince my boss that Asterisk was the way to go, and our vendor has lead us down a road where my reputation has been damaged because the vendor cannot perform as expected. |
| 17:45.05 | DSpair | Strom, I felt the same way. |
| 17:45.19 | Strom | DSpair: what do you need your vendor to do? |
| 17:45.48 | outtolunc | maybe when i hit the 10 year mark for asterisk usage i'll finally get my dcap, as 5 years has come and gone already <G> |
| 17:45.51 | DSpair | I've been developing on Linux for almost 16 years now, so I m by no means the 'tard' that my vendor appears to think I am. |
| 17:46.00 | *** join/#asterisk CRASH69 (n=crash@201.200.94.66) |
| 17:46.15 | Strom | outtolunc: perhaps you'll also learn to emote properly |
| 17:46.22 | [TK]D-Fender | DSpair: then spend some time & learn * yourself, or there's always remote consultants |
| 17:46.45 | DSpair | Strom, We wanted to outsource the management of the system because we have a VERY small shop to manage 19 branches in 14 states. We already have our hands full. |
| 17:46.50 | outtolunc | here we go |
| 17:47.02 | Corydon76-dig | DSpair: have you dCAP certification yet? |
| 17:47.14 | DSpair | Corydon76-dig, No, I haven't put any time into it. |
| 17:47.14 | outtolunc | why is using a <G> from the old bbs days NOW invalid strom? |
| 17:47.25 | Strom | no no no |
| 17:47.27 | Strom | using /me |
| 17:47.31 | Strom | not the <G> |
| 17:47.37 | DSpair | outtolunc, Because he's too young to remember Bulletin Boards. |
| 17:47.47 | Strom | I know perfectly well what <G> means :) |
| 17:48.04 | outtolunc | i use /me when it is a 'general comment i am making' |
| 17:48.22 | outtolunc | falls down |
| 17:48.24 | Strom | notes that emotes like this one work better in the third person |
| 17:48.49 | DSpair | agrees with Strom. |
| 17:49.26 | outtolunc | 1 down |
| 17:49.40 | DSpair | heads over to Safari to get a book on Asterisk. |
| 17:49.52 | Qwell | ~buybook |
| 17:49.52 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 17:50.12 | DSpair | Why buy it? I have an unlimited Safari subscription. |
| 17:50.24 | Qwell | because lmadsen needs beer money. |
| 17:50.52 | Corydon76-dig | and buying it ensures that the authors are likely to come out with a new edition |
| 17:51.08 | DSpair | Qwell, but Jim and Jared don't? |
| 17:51.24 | Qwell | they do too, but they aren't here. :p |
| 17:51.30 | Qwell | and neither is Leif, apparently |
| 17:51.48 | Strom | let's all say nasty things about them |
| 17:51.58 | DSpair | ROFL.... |
| 17:52.20 | DSpair | You guys are almost as much fun as the #linuxhelp gang on Undernet. |
| 17:52.26 | Corydon76-dig | Not too nasty. I wrote a fair amount of the appendices... |
| 17:52.35 | Qwell | DSpair: #linuxhelp folks on EFnet are far better |
| 17:52.39 | Qwell | and I'm not biased or anything. |
| 17:52.55 | _MrSeb_ | [TK]D-Fender: I've some problem on server, I prepare all things for the next time (config and sip debug info)... very thanks, and excuse me, but you seem to have clear idea on nat |
| 17:52.55 | Strom | that Corydon76-dig, what a fop |
| 17:52.58 | DSpair | Qwell, Well, we all have fond memories of where we grew up... |
| 17:53.11 | Qwell | Does Undernet even...exist anymore? |
| 17:53.27 | DSpair | Qwell, Yes, alive and well.... No thanks to you I might say. |
| 17:53.35 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:53.43 | Qwell | I don't buy it. |
| 17:53.44 | Corydon76-dig | Strom: "two British fops" is one of my all-time favorite SNL skits |
| 17:54.03 | Strom | i dont think ive seen it |
| 17:54.56 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 17:55.08 | DSpair | Corydon76-dig, I like the funny commercials they always did... Like "Big Red" the bloody viking. |
| 17:58.14 | Strom | i think my favorite fake commercial was "chess for girls" |
| 17:58.18 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:58.32 | scampbell | More cowbell. |
| 17:58.48 | x86 | hahaha |
| 17:58.50 | Qwell | scampbell: Best skit ever. Period. |
| 17:58.52 | Strom | scampbell: no, see, that one's been quoted to death six times over |
| 17:59.00 | *** join/#asterisk LeBowlingAlley (n=derek@71.16.158.170) |
| 17:59.13 | DSpair | Anyone familiar with the cause of this message? " chan_zap.c:8249 zt_pri_error: !! Got reject for frame 4, retransmitting frame 4 now, updating n_r!" |
| 17:59.39 | Strom | DSpair: probably a transmission error on the DS1 |
| 17:59.41 | LeBowlingAlley | I'm trying to do a direct dial to an extension's voicemail by dialing *+extension and the call just keeps dropping. Here's the pastebin: http://pastebin.ca/1055867 |
| 17:59.57 | LeBowlingAlley | this works on other extensions |
| 18:00.11 | DSpair | Strom, Not to sound condescending, but I figured that much... |
| 18:00.43 | Strom | LeBowlingAlley: i'm still waiting for pastebin to load |
| 18:00.50 | LeBowlingAlley | yeah...it's slow |
| 18:00.55 | Strom | LeBowlingAlley: use pastebin.com |
| 18:01.08 | Strom | or a different pastebin |
| 18:01.27 | LeBowlingAlley | http://pastebin.com/m64fd84d8 |
| 18:01.34 | [TK]D-Fender | LeBowlingAlley: ... |
| 18:01.36 | [TK]D-Fender | ~freepbx |
| 18:01.37 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:01.37 | Strom | LeBowlingAlley: oh christ, it's freepbx, isnt it |
| 18:01.38 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 18:01.53 | LeBowlingAlley | haha |
| 18:02.08 | LeBowlingAlley | yeah...i'm there. patiently waiting |
| 18:02.19 | [TK]D-Fender | LeBowlingAlley: You keep on waiting then |
| 18:02.27 | ManxPower | This channel is NOT 2nd level #Freepbx support. |
| 18:02.35 | Qwell | hell, it's not any level |
| 18:02.42 | seanbright | 7th circle |
| 18:02.47 | LeBowlingAlley | keep it coming guys |
| 18:02.48 | LeBowlingAlley | i can take it |
| 18:02.51 | [TK]D-Fender | burns |
| 18:02.54 | Strom | what kind of tard decides "*" is the voicemail prefix? that'll just completely blow away every single one of your VSC assignments |
| 18:02.57 | DSpair | Nanu Nanu? |
| 18:03.18 | [TK]D-Fender | DSpair: The mothership is coming! |
| 18:03.22 | ManxPower | Strom: one that does not know telecom? |
| 18:03.29 | [TK]D-Fender | Strom : VSC -whore :p |
| 18:03.29 | Strom | ManxPower: I guess so |
| 18:03.33 | DSpair | Mork calling orsen, come in orsen... |
| 18:03.52 | [TK]D-Fender | DSpair: Careful.. you're (carbon)dating yourself ;) |
| 18:04.17 | ManxPower | LeBowlingAlley: Why exactly are you on this channel? |
| 18:04.26 | DSpair | [TK]D-Fender, You should have seen it the first time I made a "Hogan's Hero's" reference around my gf. |
| 18:04.30 | LeBowlingAlley | so i can feel your wrath |
| 18:04.42 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 18:04.51 | ManxPower | someone /kickban LeBowlingAlley so he can really feel it. |
| 18:05.01 | *** join/#asterisk Ghost1 (n=Ghost1@pool-71-180-28-17.tampfl.fios.verizon.net) |
| 18:05.08 | [TK]D-Fender | LeBowlingAlley: Haven't earned our wrath yet, but ridicule is complimentary ;) |
| 18:05.08 | LeBowlingAlley | did i do something to you? |
| 18:05.21 | Strom | LeBowlingAlley: read this |
| 18:05.23 | Strom | ~vsc |
| 18:05.24 | jbot | extra, extra, read all about it, vsc is Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 18:05.28 | [TK]D-Fender | hands LeBowlingAlley some more peanuts |
| 18:05.46 | LeBowlingAlley | Thanks for the hospitality. didn't mean to disturb anyone's sleep. |
| 18:05.54 | [TK]D-Fender | Strom : He's on FreePBS anyways... its not like he has a lot of choices anyways. |
| 18:06.28 | Strom | *shrug* I have no idea how easy or difficult it is to tell freepbx how to handle voicemail and VSCs |
| 18:07.09 | [TK]D-Fender | Strom : think of it like "LA LA LA LA CAN'T HEAR YOU!!!!" |
| 18:07.16 | DSpair | Strom: Is there anything bad about using hash (#) marks to start a feature code? |
| 18:07.18 | [TK]D-Fender | plugs his ears |
| 18:07.24 | Qwell | DSpair: everything |
| 18:07.27 | [TK]D-Fender | DSpair: OMG! |
| 18:07.28 | Strom | DSpair: yes |
| 18:07.34 | [TK]D-Fender | DSpair: Now you've done it! |
| 18:07.35 | DSpair | Please explain? |
| 18:07.39 | LeBowlingAlley | kickban him |
| 18:07.40 | Strom | DSpair: # means "I'm finished dialing now" |
| 18:07.41 | Qwell | the fact that many phones can't even dial a # |
| 18:07.43 | DSpair | cowers in fear. |
| 18:07.43 | [TK]D-Fender | DSpair: Strom is going to burst a brain cell now! |
| 18:07.52 | Strom | [TK]D-Fender: give it a rest |
| 18:07.56 | Ghost1 | I wrote a bash script but I cant get the System command or Agi command to run it. I already did chmod 777 on the script but asterisk still wont run it, Is there something else I need to do? |
| 18:08.14 | edoceo | [TK]D-Fender: From our discussion yesterday - please check email |
| 18:08.54 | [TK]D-Fender | Ghost1: pastebin the complete attempt at verbose 10 and include the "ls" showing where it is, etc. Also run it manually from CLI to prove its at least somewhat functional. |
| 18:08.55 | DSpair | Strom, Hmmm . . . I wonder why our old Avaya system used them for quick dials... Weird. |
| 18:09.11 | Strom | perhaps whoever set it up wasn't clued in properly |
| 18:09.32 | Qwell | chances are, the avaya phones had a "feature" key |
| 18:09.35 | Qwell | and it was mapped to ## |
| 18:09.42 | Ghost1 | It runs fine on its own. |
| 18:10.01 | jeev | i had a dream that i was getting a 22 Floor estate |
| 18:10.12 | [TK]D-Fender | Ghost1: please provide the pastebin requested. |
| 18:10.23 | Qwell | DSpair: right? |
| 18:10.35 | [TK]D-Fender | DSpair: feel free to use whatever codes you like. Its your system. |
| 18:11.57 | [TK]D-Fender | DSpair: Nobody said you have to follow telco standard VSC's or dialing patterns. |
| 18:12.25 | [TK]D-Fender | DSpair: though somewhat recommended for telco-nazis, it doesn't matter much in the big picture. |
| 18:13.29 | Ghost1 | -- Executing System("SIP/205-b7534f40", "/var/lib/asterisk/agi-bin/gpstest2.sh") in new stack is the output I get. |
| 18:14.30 | keith4 | so... VoicePulse recommends using SIP instead of IAX2, apparently |
| 18:14.40 | Ghost1 | exten => s,8,System(/var/lib/asterisk/agi-bin/gpstest2.sh) |
| 18:16.22 | [TK]D-Fender | Ghost1: pastebin please... |
| 18:16.31 | [TK]D-Fender | keith4: Yup |
| 18:16.37 | Ghost1 | what is pastebin ? |
| 18:16.53 | ManxPower | Ghost1: what happens if you do the following at the OS CLI: /var/lib/asterisk/agi-bin/gpstest2.sh |
| 18:16.55 | [TK]D-Fender | ~pb |
| 18:16.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:16.57 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 18:18.14 | CRASH69 | :p |
| 18:18.29 | Ghost1 | It does not produce an output because its not supposed to. |
| 18:18.36 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 18:18.46 | ManxPower | Ghost1: but does it generate errors or not run for some other reason? |
| 18:18.54 | ManxPower | or just start and then wait |
| 18:18.57 | Ghost1 | Nope |
| 18:19.13 | Ghost1 | The script works. But asterisk wont run it. |
| 18:19.37 | ManxPower | Ghost1: a script that works with System() won't with with AGI(), the reverse is also true. |
| 18:19.45 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:20.04 | [TK]D-Fender | Ghost1: Please pastebin the *nix CLI output of your execution attempt as well as the "ls" backup I requested |
| 18:20.47 | Ghost1 | In new to these terms I don't understand what you want. |
| 18:20.48 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:21.09 | ManxPower | Ghost1: jbot just TOLD you what pastebin is |
| 18:21.35 | Ghost1 | #!/bin/bash |
| 18:21.36 | DarKnesS_WolF | ~pastebin |
| 18:21.36 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:21.37 | Ghost1 | lat1=$(cat /gpsraw | cut -c1-7) |
| 18:21.38 | Ghost1 | lng1=$(cat /Gpsraw2 | cut -c1-9) |
| 18:21.40 | Ghost1 | lynx -dump "http://ws.geonames.org/findNearestAddress?lat=$lat1&lng=$lng1&style=full" > gpslivedata |
| 18:21.41 | Ghost1 | That's the script |
| 18:22.03 | *** join/#asterisk egypcio (n=korn@unaffiliated/egypcio) |
| 18:22.03 | ManxPower | Ghost1: you are either an asshole or a moron. We can't fix the script when you FLOOD the channel with it. |
| 18:22.13 | [TK]D-Fender | ManxPower: its 4 lines.... |
| 18:22.23 | ManxPower | Ghost1: you can either follow the instructions here or nobody will want to heklp you. |
| 18:22.41 | [TK]D-Fender | ManxPower: if he hit 10, he'd ahve bounced :) |
| 18:22.43 | ManxPower | [TK]D-Fender: shows he's not listening. |
| 18:23.05 | [TK]D-Fender | Ghost1: now please provide the rest of what I requested in a pastebin. |
| 18:23.36 | ManxPower | I need to get back to paying work. Good luck, [TK]D-Fender. |
| 18:24.04 | DarKnesS_WolF | [TK]D-Fender: be nice to the ppl don't be mean :P |
| 18:24.04 | DarKnesS_WolF | ducks |
| 18:24.04 | [TK]D-Fender | DarKnesS_WolF: I am being nice.... |
| 18:24.24 | DarKnesS_WolF | [TK]D-Fender: evil :P |
| 18:24.39 | DarKnesS_WolF | [TK]D-Fender: u and x86 are always evils :P |
| 18:24.50 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 18:25.18 | ManxPower | DarKnesS_WolF: that's because both of them, instead of telling the user to go screw themselves, they keep trying to help. |
| 18:25.38 | *** join/#asterisk Rico29 (n=Rico@nor75-29-88-188-75-101.fbx.proxad.net) |
| 18:25.39 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 18:25.47 | ManxPower | I figure if a user can't concentrate on the issue and follow instructions then they don't deserve my free, unpaid, volunteer help. |
| 18:25.52 | Ghost1 | http://pastebin.com/d6aaaabc |
| 18:26.08 | [TK]D-Fender | Ghost1: and the rest? |
| 18:26.32 | ManxPower | You don't tell some mechanic that helps you when you are broken down at the side of the road that they are wrong and the problem is not a flat tire. |
| 18:26.49 | Ghost1 | What else did you want? the ls of the directory? |
| 18:27.09 | Maliuta | well for starters how are you trying to get * to execute it |
| 18:27.29 | ManxPower | Maliuta: a System command |
| 18:27.45 | Maliuta | error output, you know the usual fault finding stuff |
| 18:27.59 | ManxPower | which, based on the script he vomited into the channel, is not an agi |
| 18:28.00 | rwaite | hi all, i am having a pretty elusive problem with my iax provider. i am supposed to have 4 'lines' from them, so at any time i should be able to have 4 total calls going on. i cant do that, the 3rd (sometimes it works) or 4th 'line' i try to open by calling my did, it gives busy signal. if i do iax2 set debug, i can see a diff between the incoming calls that work |
| 18:28.16 | rwaite | and those that dont. after iax subclass "authreq" is sent |
| 18:28.19 | DarKnesS_WolF | ManxPower: i was just kidding wiht [TK]D-Fender :) |
| 18:28.24 | rwaite | the provider sents a subclass "vnak" |
| 18:28.38 | ManxPower | rwaite: the provider is rejecting the call |
| 18:28.40 | rwaite | is this is problem i should take up with my provider? |
| 18:29.04 | rwaite | ahh |
| 18:29.32 | rwaite | i know my provider is running asterisk too - any reason why they would reject it? (i'm trying to cover my bases before i call to yell at them) |
| 18:29.42 | [TK]D-Fender | Ghost1: I asked you for about 4 things. Repeatedly. |
| 18:29.51 | keith4 | so, VoicePulse has two main SIP gateways, it looks like... NYC and SFO. They recommend using NYC, but the server I'm coming from is in california... thoughts? |
| 18:29.55 | [TK]D-Fender | Ghost1: If you want assistance, be thorough with your backup. |
| 18:30.09 | [TK]D-Fender | keith4: Try & see. |
| 18:30.30 | keith4 | what is "too much" latency for SIP? |
| 18:30.37 | Ghost1 | I don't understand what you want. |
| 18:30.53 | Ghost1 | Im a noob. |
| 18:30.59 | [TK]D-Fender | Ghost1: Please pastebin the *nix CLI output of your execution attempt as well as the "ls" backup I requested |
| 18:31.01 | Maliuta | keith4: 2000ms |
| 18:31.14 | Ghost1 | what is an ls backup? |
| 18:31.22 | [TK]D-Fender | Ghost1: "man ls" |
| 18:32.27 | Ghost1 | what is *.nix |
| 18:32.48 | keith4 | facepalms |
| 18:32.51 | [TK]D-Fender | Ghost1: .... seriously. Get a clue. |
| 18:32.56 | Maliuta | amen |
| 18:33.01 | [TK]D-Fender | Ghost1: Linux UXIN, BSD, WHATEVER! |
| 18:33.06 | [TK]D-Fender | UNIX |
| 18:33.20 | [TK]D-Fender | Ghost1: I wasn't going to assume which exact OS you were running. |
| 18:33.34 | Ghost1 | Centos. |
| 18:33.43 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 18:33.44 | [TK]D-Fender | Ghost1: FINE. Now get moving. |
| 18:33.45 | rwaite | hmm then my asterisk sends an inval |
| 18:33.52 | keith4 | can I 'include' other files into extensions.conf ? |
| 18:33.57 | rwaite | shakes head |
| 18:34.08 | Maliuta | Ghost1: no, the OS is GNU/Linux |
| 18:35.42 | mog | Maliuta, dont be a jerk |
| 18:38.06 | rwaite | is sip better than iax for connecting to a svc provider? |
| 18:40.42 | [TK]D-Fender | rwaite: Certainly more stable. |
| 18:40.57 | Ghost1 | I know asterisk is having the problem executing it because I have never been able to run a shell script from any context. I think asterisk may not be able to use bash or something. I know you requested the cli output but there is no output to the script. and the ls of the directory wont help because all the relevant files are in their correct places. |
| 18:41.37 | DarKnesS_WolF | mog: there is no way soon that this issue about if the internet is down.. the local SIP phones can't register ? if i have a SIP provider ? |
| 18:41.40 | [TK]D-Fender | Ghost1: prove the file is in the right place with the right permissions, as the right user, and the rest of what I asked for. |
| 18:41.41 | DarKnesS_WolF | it is hell of a bug :( |
| 18:41.43 | rwaite | TK: becuase it seems that i've been having tons of issues with this provider, and i dont know if its my fault, if they are just a shoddy place or what |
| 18:41.48 | [TK]D-Fender | ghorAnything else is wasting our time. |
| 18:42.01 | [TK]D-Fender | Ghost1: Anything else is wasting our time. |
| 18:42.06 | keith4 | rwaite: "VoicePulse recommends using SIP for your incoming calls ; instead of IAX2. SIP allows for various failover ; mechanisms to ensure your calls are handled as quickly ; and reliably as possible." |
| 18:42.16 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:42.20 | rwaite | hmm |
| 18:42.42 | rwaite | would you recommend voice pulse in a business environment? |
| 18:42.58 | keith4 | no idea |
| 18:43.03 | keith4 | I just started using them |
| 18:43.11 | rwaite | seem pretty solid? |
| 18:43.15 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 18:43.17 | keith4 | haven't even set it up yet, still reading their documentation ;-) |
| 18:43.21 | keith4 | but |
| 18:43.25 | keith4 | ~itsp |
| 18:43.26 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 18:43.35 | keith4 | er |
| 18:43.38 | keith4 | ~itsplist-us |
| 18:43.39 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:44.16 | keith4 | it's 2nd on the list, in order of "respect" |
| 18:44.43 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:45.35 | [TK]D-Fender | keith4: not the order in the list.... just BEING ont he list. |
| 18:45.57 | DSpair | Would LVM/LVM2 add any undue overhead to an Asterisk box? |
| 18:48.35 | [TK]D-Fender | DSpair: Shouldn't |
| 18:51.28 | DarKnesS_WolF | dose asterisk works fine in VLANs? |
| 18:53.25 | Ghost1 | http://pastebin.com/d59ce4470 |
| 18:53.49 | spokra | rwaite: I personaly hate voicepulse and broadvoice both have terrible tech support. |
| 18:54.05 | DSpair | [TK]D-Fender, Thanks for the info. |
| 18:54.14 | rwaite | this place we're with now sucks. vipvoipsystems.com, tc systems |
| 18:54.23 | rwaite | so unreliable |
| 18:54.25 | DarKnesS_WolF | Ghost1: don'tthink it did run from the right place ! |
| 18:54.29 | DarKnesS_WolF | u have to try |
| 18:54.42 | DarKnesS_WolF | /var/lib/asterisk/agi-bin/gpstest2.sh |
| 18:54.47 | DarKnesS_WolF | as a full path |
| 18:54.54 | Ghost1 | ok |
| 18:54.58 | DarKnesS_WolF | did u include agi-bin to ur $PATH? |
| 18:55.22 | gr0mit | ~itsp-uk |
| 18:55.30 | gr0mit | ~itsplist-uk |
| 18:55.31 | jbot | from memory, itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 18:55.47 | spokra | broadvoice is just a reseller they have no idea what they are doing other then selling you an account |
| 18:56.11 | DarKnesS_WolF | gr0mit: try voicetrading they are good but u have to buy bulk and also euroIAX is okay .. fonity.de |
| 18:56.14 | DarKnesS_WolF | also |
| 18:56.25 | jeev | Fender, the site you told me about doesn't finance less than 5k! where else can i purchase ~20 polycom 330's with financing |
| 18:56.33 | DarKnesS_WolF | voipjet in USA is ok also i did use this one |
| 18:56.43 | Ghost1 | http://pastebin.com/d3a490da9 |
| 18:57.07 | DarKnesS_WolF | Ghost1: ok great it did crete the file ? |
| 18:57.11 | keith4 | [TK]D-Fender: am I misinterpreting "starting with the more respected ones" ? |
| 18:57.15 | Ghost1 | Yep. |
| 18:57.25 | Ghost1 | With the proper output. |
| 18:57.35 | DarKnesS_WolF | Ghost1: ok now pastbin ur extensions.conf " the part where u dial that AGI script |
| 18:58.26 | *** join/#asterisk l0verb0y (n=l0verb0y@124.107.127.238) |
| 18:59.48 | Ghost1 | http://pastebin.com/m192f8d95 |
| 18:59.54 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:01.34 | Ghost1 | All of it works except it wont execute the gpstest2.sh. |
| 19:02.08 | Ghost1 | Im calling it from an IVR. |
| 19:03.02 | [TK]D-Fender | Ghost1: And * CLI output of the failed usage attempt? |
| 19:03.30 | [TK]D-Fender | jeev: Get a bank. |
| 19:03.55 | DarKnesS_WolF | Ghost1: change the script to lynx -dump "http://ws.geonames.org/findNearestAddress?lat=$lat1&lng=$lng1&style=full" > /var/lib/asterisk/agi-bin/gpslivedata |
| 19:04.07 | Ghost1 | ok. |
| 19:04.08 | DarKnesS_WolF | and then execute and post ur CLI |
| 19:04.11 | DarKnesS_WolF | output |
| 19:05.29 | *** join/#asterisk Prez00 (n=fmedina@dynamic21-170.MAN-B2-1.cablenet.com.ni) |
| 19:05.31 | Prez00 | hello |
| 19:05.50 | Prez00 | a little off topic, but I figured someone here may know, are there any cell phone channels, I am looking for the name of some special codes carriers use... |
| 19:06.40 | Prez00 | I just need to know what they are called, not any codes per se, codes sent using * before dialing.. |
| 19:06.44 | jeev | Fender, yea, he has equipment finance services from bank of america~ |
| 19:06.44 | jeev | ! |
| 19:08.13 | Ghost1 | http://pastebin.com/d19ce7ffa |
| 19:08.21 | unpaidbill | good morning sirs |
| 19:10.15 | *** join/#asterisk jicksta (n=jicksta@adsl-69-109-235-72.dsl.pltn13.pacbell.net) |
| 19:10.23 | DarKnesS_WolF | Ghost1: that is all ? |
| 19:10.36 | DarKnesS_WolF | can't be ! there is should be another file played and then hangup |
| 19:11.14 | DarKnesS_WolF | check if u have /var/lib/asterisk/agi-bin/gpslivedata |
| 19:11.16 | Ghost1 | it works!. |
| 19:11.16 | Prez00 | so, no cell phone people arounf.. |
| 19:11.33 | Strom_C | Prez00: |
| 19:11.36 | Strom_C | ~vsc |
| 19:11.36 | jbot | well, vsc is Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 19:11.46 | DarKnesS_WolF | Ghost1: so the problem was in ur script |
| 19:11.52 | rwaite | i wonder if compiling asterisk can cause the problems im having |
| 19:12.03 | rwaite | maybe i should use a debian pkg |
| 19:12.05 | Ghost1 | It was the permissions on gpslivedata. |
| 19:12.17 | DarKnesS_WolF | i think the file were create somewhere else ? |
| 19:12.24 | DarKnesS_WolF | ah ic |
| 19:12.35 | DarKnesS_WolF | Ghost1: what u were trying to do anyway ? |
| 19:12.48 | Prez00 | thanks, I was under the impression there were some codes called something like SSMB codes or something like that? |
| 19:13.03 | Strom_C | *shrug* check the numbering plan administrator |
| 19:14.07 | Ghost1 | Reverse Geocoding to my gps position's so I could find out what street im on over the phone. I use accutracking to keep track of where I am and I wanted to know the zip code im in and the nearest street over the phone. |
| 19:14.55 | DarKnesS_WolF | nice |
| 19:15.03 | Qwell | Prez00: supplementary service codes? |
| 19:15.14 | DarKnesS_WolF | Ghost1: i'm sure there is much clearner way to do it :-) but anyway enjoy ;-) |
| 19:15.51 | Ghost1 | I will try a rewrite in php. but for the time being it works. |
| 19:16.02 | Ghost1 | :-) |
| 19:16.06 | DarKnesS_WolF | Ghost1: ok congratulations |
| 19:16.22 | Prez00 | Qwell: might be... I know i use them on my carrier's "POS" phone to reload othere phones with minutes from the POS phone, using something liek *811*1*phone to reload*amount to reload*PIN# |
| 19:16.33 | Prez00 | Qwell: then a menu pops on my phone to confirm reload |
| 19:16.40 | Qwell | it's provider specific |
| 19:16.52 | Prez00 | Qwell: are those SSCs? |
| 19:16.54 | Qwell | like t-mobile has #min# |
| 19:17.56 | Prez00 | Qwell: Ok, so those are called SSCs? |
| 19:18.17 | Qwell | no name is given |
| 19:18.21 | Qwell | just "codes for supplementary services" |
| 19:18.28 | Prez00 | Qwell: gotcha |
| 19:18.32 | Prez00 | Qwell: thanks |
| 19:19.20 | ManxPower | good grief! here is a list of the NANPA SVCs: http://nanpa.com/number_resource_info/vertical_service.html |
| 19:19.35 | Prez00 | what about USSD? |
| 19:19.56 | Prez00 | After looking up SSC bumped into that, think those might be it.. |
| 19:20.04 | rwaite | hmm. would it ever be right to have two host= entries under one [] in iax.conf? one had an ip and the other has dynamic |
| 19:20.10 | Qwell | still provider specific |
| 19:20.15 | Qwell | (hence the "unstructured") |
| 19:20.17 | [TK]D-Fender | rwaite: No. |
| 19:20.22 | ManxPower | rwaite: never |
| 19:20.26 | Prez00 | Qwell: perfect... thanks again.. |
| 19:20.32 | [TK]D-Fender | rwaite: there is a "defaulthost" for that though |
| 19:20.35 | rwaite | i see. |
| 19:20.46 | [TK]D-Fender | rwaite: which I think does what you're looking for. |
| 19:21.14 | DarKnesS_WolF | gtg guys |
| 19:21.15 | DarKnesS_WolF | good night |
| 19:21.45 | *** join/#asterisk implicit (n=implicit@dhcp-x218-164.mobile.uci.edu) |
| 19:22.58 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 19:23.00 | IPkaf | hi |
| 19:23.19 | IPkaf | when i try to do this i got this error message |
| 19:23.39 | IPkaf | http://paste.ubuntu.com/22934/ |
| 19:24.37 | IPkaf | hello |
| 19:24.41 | x86 | try to do _what_ exactly? |
| 19:24.44 | spokra | your missing autoconf |
| 19:24.55 | unpaidbill | this, x86! |
| 19:25.12 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:25.23 | IPkaf | i try to follow this guide |
| 19:25.23 | unpaidbill | haha autoconf missing |
| 19:25.24 | Qwell | why would you need to run autogen? |
| 19:25.48 | Qwell | wow, they don't ship one. That's silly |
| 19:25.53 | unpaidbill | apt-get install autoconf, yum install autoconf |
| 19:26.05 | spokra | guess you would have to look at autogen.sh ... probably calls autoconf... :> |
| 19:26.13 | IPkaf | thx unpaidbill |
| 19:31.51 | keith4 | anyone ever used a UTStarcom F3000? I figure they can't be great if voipsupply is having a "buy one, get one free" sale on them |
| 19:31.54 | keith4 | but it's tempting |
| 19:32.45 | rwaite | how safe is it to 'make install' over top of an already installed asterisk? should i go thru and rm all the previous binaries/configs first? |
| 19:33.07 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 19:33.34 | unpaidbill | rwaite i have never had a problem installing over the top, just make sure if you change versions that you compile all the same modules, or delete the modules directory before the make install |
| 19:33.49 | rwaite | thanks :) |
| 19:34.14 | rwaite | its been so long since slackware, i forgot what its like to not use packages |
| 19:34.21 | unpaidbill | haha |
| 19:35.10 | Kobaz | ghetto |
| 19:35.12 | rwaite | i looked in the bug tracker and it seems this is a major issue with .21, so im not the only one |
| 19:35.57 | Kobaz | rwaite: i always build asterisk (or any other apps from source) with ./configure --prefix=/apps/asterisk-1.4.14 |
| 19:36.01 | Kobaz | or whatever |
| 19:36.07 | Kobaz | so each app is completely self contained |
| 19:36.22 | rwaite | and then have the start scripts reference it in that subdir> |
| 19:36.25 | rwaite | thats a good idea |
| 19:46.43 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 19:47.01 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 19:52.06 | Superbartt | hmmfg, what would be the term for "Incoming calls to all phones and the first to pick up gets the call?"^^ |
| 19:52.56 | Qwell | not so much a term, but it can be accomplished with something like Dial(SIP/101&SIP/102) |
| 19:53.14 | [TK]D-Fender | Superbartt: "First come, first served" :p |
| 19:53.37 | Superbartt | hmmfg, but that isn't really "managable" if new extensions come and go ^^ |
| 19:54.50 | [TK]D-Fender | Superbartt: As manageable as any other means. |
| 19:55.26 | [TK]D-Fender | Superbartt: For which that basically leaves app_queue.... |
| 19:55.37 | Superbartt | so that's the best/preffered method for letting all phones ring on an incoming call? |
| 19:56.32 | [TK]D-Fender | Superbartt: Exactly what Qwell told you. |
| 19:57.18 | Superbartt | I found some people referring to ring groups, but i think that's an asterisk home feature? |
| 19:57.30 | Superbartt | it's not quite clear, excepted that something like a ring group is mentione |
| 19:57.34 | Superbartt | mentioned* |
| 19:58.14 | [TK]D-Fender | Superbartt: An inappropriate term. |
| 19:58.28 | [TK]D-Fender | Superbartt: What you've got is what we've mentioned. |
| 19:58.55 | Superbartt | ok than i know enough :) thanks |
| 20:05.34 | Superbartt | Jun 25 22:05:18 ERROR[3588]: rtp.c:1014 ast_rtp_new_with_bindaddr: No RTP ports remaining. Can't setup media stream for this call. |
| 20:05.41 | Superbartt | doesn't seem right :x |
| 20:07.42 | ManxPower | "ring groups" are not designed to work with devices randmonly coming and going |
| 20:07.50 | ManxPower | We call that "anarchy" |
| 20:09.14 | Superbartt | yeah I kind of get that it doesn't make much sense ^^ |
| 20:09.15 | [TK]D-Fender | 'cause iiiiiiiii wanna bring.... ANARCHY! |
| 20:09.26 | rwaite | ill tell you what, setting up voip systems is enough to drive a man to suicide |
| 20:09.51 | ManxPower | Superbartt: how many rtp ports are you using? |
| 20:10.02 | [TK]D-Fender | Superbartt: And lack of rtp ports says you've got a ton of lingering calls, or you've closed your range too small. |
| 20:10.23 | ManxPower | rwaite: you need to understand telecom, networking, NAT, RTP, SIP, phones, Asterisk, Linux. This is not a walk in the park. |
| 20:10.33 | jeev | fender!!!!!! |
| 20:10.40 | Superbartt | a college tought it was fun to see how many simultanious connections his desktop could make to the server -_- |
| 20:10.42 | Superbartt | moron oO |
| 20:10.57 | Superbartt | anyone has a patch for stupdity yet? |
| 20:10.57 | rwaite | linux is the only one i feel confident with |
| 20:10.58 | rwaite | lol |
| 20:11.32 | [TK]D-Fender | Superbartt: I have one. its large, highly adhesive and is meant to cover up all repiratory orifices simultaneously. |
| 20:12.25 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 20:12.36 | Superbartt | lol |
| 20:14.03 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 20:14.27 | IPkaf | hi |
| 20:14.33 | IPkaf | i got a little problem |
| 20:14.50 | IPkaf | i try to configure Gtalk on my asterisk |
| 20:14.53 | IPkaf | serveer |
| 20:15.16 | IPkaf | i don't know how to make call from gtalk client |
| 20:15.29 | IPkaf | what i have to do ? |
| 20:15.48 | [TK]D-Fender | IPkaf: Go read the WIKI page on it. |
| 20:15.50 | [TK]D-Fender | ~wikis |
| 20:15.50 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 20:15.59 | [TK]D-Fender | NEXT!@@! (c) BKW |
| 20:16.03 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-230-77.unitz.ca) |
| 20:16.19 | IPkaf | yes i get info there |
| 20:16.56 | eric2 | If I do something like this: Dial(SIP/500&SIP/602) is it possible to change the callerID for SIP/602 so they see something slightly different? |
| 20:17.09 | [TK]D-Fender | eric2: No. |
| 20:17.13 | IPkaf | and this is the link that i followed |
| 20:17.17 | IPkaf | to configure asterisk |
| 20:17.18 | [TK]D-Fender | eric2: Both will see the same thing |
| 20:17.19 | IPkaf | http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 20:17.32 | eric2 | is there a way? what if I split up the Dial commands |
| 20:17.38 | eric2 | like fork them or som'n |
| 20:17.42 | IPkaf | where there's no explaination on usage of gtalk |
| 20:17.56 | [TK]D-Fender | eric2: If you split them then its 2 calls and you do whatever you want before each call out |
| 20:18.03 | IPkaf | once configured all information that u can see there |
| 20:18.12 | IPkaf | how to make call ? |
| 20:18.23 | [TK]D-Fender | eric2: Or reconsider how you're dialing the other one (think "other tech"). Think long and hard on that one. |
| 20:18.33 | eric2 | really just trying to call an IP phone and a cell phone at the same time but have an extra character appear on the cell phone caller id so you'd know that someone's calling the office and not me directly |
| 20:18.33 | IPkaf | have i have to add all my sip extension on the gtalk clien t? |
| 20:18.35 | IPkaf | ??? |
| 20:18.36 | [TK]D-Fender | IPkaf: what have you tried? |
| 20:18.54 | IPkaf | following that guide http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 20:19.05 | IPkaf | i configure gtalk |
| 20:19.58 | IPkaf | now i just want to make call from gtalk client to my asterisk server's sip extension |
| 20:20.00 | IPkaf | ???? |
| 20:20.05 | IPkaf | hwo to ood ? |
| 20:20.09 | IPkaf | do ? |
| 20:20.45 | IPkaf | or is it possible ? |
| 20:21.01 | IPkaf | or is it for that purpose that link ? |
| 20:21.03 | IPkaf | ??? |
| 20:21.05 | [TK]D-Fender | IPkaf: add the user that * logs in as as a friend and CALL IT. |
| 20:21.29 | IPkaf | okay |
| 20:21.37 | IPkaf | if i understand it correctly |
| 20:22.06 | IPkaf | i just have to add my server's sip extension |
| 20:22.16 | [TK]D-Fender | IPkaf: NOT A SIP EXTENSION. |
| 20:22.22 | IPkaf | on my gtalk client is it ? |
| 20:22.27 | IPkaf | okay |
| 20:22.28 | [TK]D-Fender | IPkaf: * logs in to google talk just like a NORMAL CLIENT. |
| 20:22.42 | [TK]D-Fender | IPkaf: You call it like you would call any other client. |
| 20:23.37 | IPkaf | what u mean saying asterisk log ? |
| 20:23.51 | IPkaf | sorry i m beginner |
| 20:24.12 | [TK]D-Fender | IPkaf: * connects as a USER on gtalk. that is just like anybody else connected to gtalk. you call it the same way |
| 20:31.02 | IPkaf | is it freeswitch better than asterisk |
| 20:31.04 | IPkaf | is it ??? |
| 20:32.11 | [TK]D-Fender | IPkaf: You should really stop running like a headless chicken and get a clue. |
| 20:32.33 | [TK]D-Fender | IPkaf: "better" is a pretty useless comparison. Go try it yourself and see what they're about. |
| 20:32.56 | [TK]D-Fender | IPkaf: They are meant for differnt things. |
| 20:33.06 | [TK]D-Fender | ok, off home, bbiab. |
| 20:33.32 | IPkaf | ok |
| 20:33.37 | gitguy | yes, it is |
| 20:33.43 | IPkaf | it is not solving my problem |
| 20:34.24 | *** join/#asterisk spokra (n=spokra@gumby.sea0.speakeasy.net) |
| 20:34.55 | IPkaf | logically |
| 20:34.59 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 20:34.59 | IPkaf | gtalk client |
| 20:35.10 | IPkaf | is isoled to my asterisk |
| 20:35.12 | IPkaf | server |
| 20:35.21 | IPkaf | i can make call from gtalk |
| 20:35.23 | IPkaf | server |
| 20:35.23 | ManxPower | isoled? |
| 20:35.25 | IPkaf | and |
| 20:35.42 | IPkaf | also there is no dialpad on my gtalk client |
| 20:35.51 | IPkaf | yes isoled |
| 20:35.56 | IPkaf | speraer |
| 20:36.00 | IPkaf | separed |
| 20:36.11 | ManxPower | isoled is not a word. Perhaps you are looking for "isolated" or "seperated" |
| 20:36.32 | IPkaf | yes i know it's difficult to remain all my ten language vocabulary |
| 20:36.49 | ManxPower | IPkaf: then perhaps you need to find someone that speaks your language to help you? |
| 20:37.35 | ManxPower | Or you could ask on the mailing list, that will give you an opportunity to clearly and carefully think about the problem as well as list the items you have tried. |
| 20:38.20 | jeev | where is lmadsen |
| 20:38.20 | jeev | i have a proposal to change the name from asterisk to jeev, it suits it better. |
| 20:38.20 | IPkaf | no |
| 20:38.21 | IPkaf | i want to improve my english language |
| 20:38.26 | IPkaf | i knwo it's bad |
| 20:38.31 | IPkaf | okay |
| 20:38.33 | IPkaf | for my problem |
| 20:38.36 | IPkaf | what to do ? |
| 20:39.20 | IPkaf | whonoz drmessano and adam1 on freepbx ??? |
| 20:39.51 | IPkaf | and also errr from freepbx cahnel |
| 20:39.55 | IPkaf | chanel |
| 20:39.57 | IPkaf | ?? |
| 20:40.11 | errr | go away IPkaf |
| 20:40.27 | TJNII | Or at least start putting one complete thought per line. |
| 20:40.35 | IPkaf | ban him errr |
| 20:40.43 | errr | TJNII: he is mad cause we banned him from #freepbx |
| 20:40.46 | IPkaf | he got a big holl on his mouth |
| 20:40.50 | IPkaf | hole |
| 20:40.54 | TJNII | errr: I see. |
| 20:41.11 | errr | he is a no good trolling help vampire |
| 20:41.58 | IPkaf | shu t ur mouth |
| 20:42.00 | IPkaf | errr: |
| 20:42.03 | IPkaf | bye to all |
| 20:42.12 | IPkaf | sorry there is a mistaken here |
| 20:42.13 | IPkaf | bye |
| 20:42.16 | IPkaf | thx to all |
| 20:42.19 | IPkaf | by |
| 20:42.33 | IPkaf | tetedecafar errr |
| 20:43.42 | errr | what a troll. |
| 20:44.03 | Strom_C | IIRC he's been problematic in here too |
| 20:44.13 | errr | yes |
| 20:44.25 | errr | he should be banned from the internet as a whole |
| 20:44.58 | Strom_C | perhaps be forced to make a public statement apologizing for wasting the internet |
| 20:46.39 | jeev | Strom, what do you think bout the aformentioned change of the name ASTERISK to JEEV? |
| 20:46.52 | Qwell | jeev: Sure, send the suggestion to marketing@digium.com |
| 20:46.55 | Qwell | we'll get right on that |
| 20:47.07 | jeev | ;) |
| 20:47.07 | ManxPower | I think it should be changed to the Asterisk PBX Tookit, not the Asterisk PBX |
| 20:47.22 | Strom_C | I think we should rename it "Phyllis" |
| 20:47.29 | ManxPower | Because it's not really a PBX anywat. |
| 20:47.45 | ManxPower | Strom_C: That's a Diller of a suggestion |
| 20:52.40 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:53.18 | *** join/#asterisk RoyK (n=roy@ip-183-25-149-91.dialup.ice.no) |
| 20:56.18 | jaytee | Phyllis? why not Rhoda? |
| 20:56.40 | jeev | Phyllis would be a turn off, i'd use the damn microsoft VOIP system thing if this changed to Phyllis |
| 20:56.46 | jaytee | lol |
| 20:57.04 | jaytee | how about stfuVOX |
| 20:57.07 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:59.41 | jaytee | or we could just name if Marvin after the depressed robot in HHGTTG |
| 20:59.57 | *** join/#asterisk kensukeido (i=c829e4f4@gateway/web/ajax/mibbit.com/x-fd234107e67c5f44) |
| 21:00.03 | jeev | heh |
| 21:01.02 | jaytee | still, it doesn't matter whether we call it shit or shinola we'll still get all the dweebes coming in here asking trixbox/freepbx questions. |
| 21:01.18 | gitguy | haha |
| 21:01.51 | gitguy | just use /ban /kick |
| 21:02.02 | gitguy | best commands ever |
| 21:02.28 | gitguy | or /ignore |
| 21:03.22 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 21:06.53 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 21:07.42 | [TK]D-Fender | gitguy: kick-ban spares OTHERS. be kind to your neighbours ;) |
| 21:08.10 | jblack | Just out of curiosity about pris..., if one set switchtype to dms100, and the provider was set to 5ess, is it possible that things "mostly" work? |
| 21:08.55 | gitguy | yeah i guess |
| 21:10.13 | ManxPower | jblack: correct. |
| 21:10.38 | jblack | That's very good to know. |
| 21:10.39 | Strom_C | jblack: why guess? just call them and ask |
| 21:11.31 | jblack | There's been calls made, and questions asked. I'm not entirely convinced they're 100% on the game themselves. |
| 21:12.14 | Strom_C | are you sure they're not just using NI2? |
| 21:12.17 | jblack | Any good suggestions on a pile-o-stuff I could RTFM to get up to snuff on this stuff? |
| 21:12.22 | Strom_C | ~101 |
| 21:12.23 | jbot | hmm... 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 21:12.29 | Qwell | what he said |
| 21:12.34 | Strom_C | what I said |
| 21:13.01 | jblack | This is perfect |
| 21:13.20 | ManxPower | jblack: most people use NI2 (national) |
| 21:14.24 | jblack | Ok. Following most people is generally a good thing. I think I can put in a request to change switch type. |
| 21:14.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:14.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:16.02 | *** part/#asterisk RoyK (n=roy@ip-183-25-149-91.dialup.ice.no) |
| 21:19.47 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 21:21.20 | talntid | Hey ManxPower, if the wrong switchtype was selected, would the service possible work fine for a week, then all the sudden get unreliable, or would it always be flawed? |
| 21:21.52 | ManxPower | talntid: I don't know. |
| 21:21.58 | talntid | ok, thanks :) |
| 21:33.47 | Corydon76-dig | talntid: it would always be flawed |
| 21:34.11 | Corydon76-dig | talntid: that said, most switchtypes are more similar than different |
| 21:35.24 | *** join/#asterisk mtown_nerd (n=JHester@fileserver.ghruaim.net) |
| 21:53.14 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-71-252.socal.res.rr.com) |
| 21:54.22 | frieze | anyone have an ubuntu asterisk init.d startup script they can send me? Somehow mine didn't get installed this time round and I don't want to jinx my finally working config by running the install again |
| 21:55.15 | TJNII | There's a make option for that. |
| 21:55.22 | TJNII | I don't remember what it is, though. |
| 21:56.21 | jblack | tjnii: there's a make option to reinstall the init.d script for ubuntu? |
| 21:56.22 | frieze | ah, it's in contrib in the source |
| 21:56.30 | jblack | frieze: I can give you mine if you want. |
| 21:57.33 | jblack | frieze: Dcc offer for you. |
| 21:57.35 | frieze | oops |
| 21:57.38 | frieze | typed right over it |
| 21:57.47 | jblack | want me to send again? |
| 21:57.54 | frieze | if you could |
| 21:58.46 | TJNII | jblack: Well, AFAIK ubuntu is a spinoff of Debian, and when I installed * on Debian (from source, though) the makefile had an option to build/install an init script. |
| 21:58.50 | Strom_C | i think you just run "make config" :) |
| 21:59.05 | jblack | strom_c: I don't think packages come with make config. ;) |
| 21:59.14 | Strom_C | ugh, packages |
| 21:59.18 | jblack | the dcc looks frozen. http://linuxguru.net/~jblack/asterisk |
| 21:59.24 | ManxPower | jblack: then complain to the package maintainer! |
| 21:59.50 | jblack | "HEY! This binary package doesn't have the source code and build framework!" |
| 22:00.07 | ManxPower | or the init scripts! |
| 22:00.36 | jblack | They do. I think he said something about him doing somethign to wipe it. |
| 22:00.52 | jblack | by the way, thanks for the answer earlier. |
| 22:01.17 | jblack | frieze: You find it ok on http? |
| 22:01.24 | frieze | yeah |
| 22:01.34 | frieze | apparently my current firewall does not like dcc |
| 22:01.38 | gitguy | is asterisk really open source, free software or is owned by digium? |
| 22:01.47 | jblack | Yeah. I think it's on your side. mine sent 204 bytes. |
| 22:02.07 | jblack | gitguy: yes to all three. |
| 22:02.40 | ManxPower | gitguy: Yes. |
| 22:02.50 | ManxPower | Asterisk is not, however, PUBLIC DOMAN |
| 22:02.54 | ManxPower | or DOMAIN either |
| 22:02.57 | gitguy | that sucks |
| 22:03.05 | Qwell | why does it suck? |
| 22:03.10 | gitguy | it's a nice way to stop innovation |
| 22:03.14 | frieze | ? |
| 22:03.15 | Qwell | ... |
| 22:03.16 | ManxPower | feel free to write a public domain PBX |
| 22:03.23 | gitguy | freeswitch is one |
| 22:03.27 | frieze | I think everone who wants to innovate in this area is pretty much working on asterisk |
| 22:03.30 | ManxPower | gitguy: The source to all open source projects are owned by SOMEONE |
| 22:03.32 | Qwell | No, it *IS NOT* |
| 22:03.37 | gitguy | it is |
| 22:03.50 | gitguy | well, whatever |
| 22:03.52 | Qwell | bkw and anthm would absolutely disagree with that point. |
| 22:03.52 | frieze | time to /join #asterisk-flamewar |
| 22:04.31 | jblack | gitguy: Many people think the licensing in things such as gpl, forcing things to stay public, is a good thing. |
| 22:04.33 | russellb | it is absolutely without a doubt not public domain |
| 22:05.02 | CrazyTux[m] | Say I want to do a mid-session change, activated on PSTN, end i.e. Asterisk -> Cell -> Cell Signals, back to Asterisk, which initiates some other action, with the current call, is this possible? and where should I look. |
| 22:05.03 | ManxPower | Asterisk is GPL, like many GPL projects the owner of the copyright on the code can do anything they like with it. |
| 22:05.38 | jblack | I'm not talking about what Digium can do. I'm talking about what gitguy can do. |
| 22:06.28 | jblack | my point would hold anyways. One can't retroactively relicense released versions of gpl software anyways. |
| 22:07.17 | ManxPower | jblack: gitguy can do anything that does not violate the GPL. |
| 22:07.55 | TJNII | Hmmm, Which * service would use port 2727? |
| 22:07.56 | florz | jblack: of course you can relicense |
| 22:08.11 | Qwell | TJNII: umm...h323 maybe? |
| 22:08.28 | TJNII | Possibly.... Let me check. |
| 22:08.30 | Qwell | mgcp |
| 22:08.33 | ManxPower | florz: but you can't revoke the existing license |
| 22:08.37 | jblack | florz: You can reship with a different license, but no, you absolutely can not retroactively relicense. |
| 22:09.48 | ManxPower | Freeswitch is not even the GPL, it's the MPL. |
| 22:10.14 | Qwell | (which isn't compat with the GPL, oddly enough) |
| 22:10.16 | florz | ManxPower: well, that not, of course |
| 22:10.25 | frieze | does sip show peers cur off very long name/usernames? |
| 22:10.30 | frieze | in the cli I mean |
| 22:10.58 | florz | jblack: well, depends on what you mean by "relicense", then - you simply can't revoke licenses where the contract doesn't allow you to do so |
| 22:12.52 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 22:12.52 | jblack | I'm not interested in linguistic gymnastics. If you understand what I meant now, then I'm happy |
| 22:14.37 | jblack | hell. According to my doctor, I'm apparently not interested in any gymnastics at all. |
| 22:14.53 | jblack | looks up the pancreas |
| 22:14.56 | gitguy | jblack: i don't have anything against digium or asterisk, i think they do a nice work... and i like them, i don't think proprietary is bad also... i just think that software is better developed on a truly open source fashion |
| 22:15.05 | ManxPower | jblack: *I* could have told you that and I would not have chraged $80 |
| 22:15.20 | ManxPower | gitguy: Asterisk is open source. |
| 22:15.26 | ManxPower | The source. It is open. |
| 22:15.56 | frieze | i think gitguy doesn't get the whole gpl vs public domain |
| 22:15.58 | jblack | Well, they told me that, and you got charged $80. =) |
| 22:16.29 | ManxPower | gitguy: You understand that the FSF owns the copyrights for much of the linux CLI stuff, Linus owns the copyright for much of the Linux kernel. |
| 22:16.59 | ManxPower | Oddly enough, the Mozilla Foundation owns the copyright to most of Mozilla stuff |
| 22:17.00 | russellb | the FSF requires you to assign copyright to them to contribute to their projects |
| 22:17.14 | jblack | gitguy: So, you think users of free software should have the right to derive and make the results proprietary. Check out the bsd license for that. Which Asterisk is not. |
| 22:17.20 | jblack | russellb: No, not always. |
| 22:17.22 | ManxPower | russellb: do that allow a simple forever license like Digium does? |
| 22:17.41 | ManxPower | does, not do |
| 22:18.18 | ManxPower | russellb: I never could understand the whole "controversy" about Digium's copyright stuff |
| 22:18.34 | jblack | Oh, I can explain that if you like. |
| 22:18.41 | russellb | the digium license agreement does not assign copyright |
| 22:18.56 | russellb | it grants digium an unlimited license to use it ... but the original author retains copyright ... |
| 22:19.12 | ManxPower | russellb: there are no longer 2 options? |
| 22:19.19 | Corydon76-dig | ManxPower: nope |
| 22:19.33 | ManxPower | Corydon76-dig: cool |
| 22:19.42 | russellb | neither of the older ones did that, either |
| 22:19.46 | Corydon76-dig | Technically, the second option, of releasing your work to the public domain is, in fact, on shaky legal ground |
| 22:19.58 | *** join/#asterisk sack (n=sack@183.Red-81-33-110.dynamicIP.rima-tde.net) |
| 22:20.03 | ManxPower | Huh? There was a license that specifically assigned copyrights to Digium |
| 22:20.08 | jblack | If I write free software, it's for the intent that it stays open. The rights granting gives digium the right to make a proprietary copy of that work. |
| 22:20.24 | Corydon76-dig | There is only one provision in US law for something being available under public domain, and the author releasing it as such is NOT that option |
| 22:20.33 | ManxPower | I had to decide between the two options when I sent in the code for INVALID_EXTEN |
| 22:20.59 | Corydon76-dig | ManxPower: No, the two licenses were always a) licensing a perpetual license, or b) releasing to public domain |
| 22:21.27 | gitguy | ManxPower: yes i understand that |
| 22:21.46 | gitguy | jblack: i agree more with the GPL than with BSD |
| 22:21.54 | Corydon76-dig | ManxPower: the second one was called the disclaimer |
| 22:22.03 | ManxPower | Corydon76-dig: Ah. |
| 22:22.05 | gitguy | ManxPower: so Digium only owns the COPYRIGHT right? |
| 22:22.11 | jblack | gitguy: The difference I just described is the single largest difference between them. |
| 22:22.28 | Corydon76-dig | gitguy: Digium owns the copyright for much, but not all of the code in Asterisk. |
| 22:22.29 | ManxPower | I went with a very slightly modified version of the perpetual license when I sent it in |
| 22:22.45 | russellb | s/much/most/ |
| 22:22.53 | Corydon76-dig | gitguy: what Digium does have, though, are the licensing provisions that permit them to relicense the entire codebase |
| 22:23.31 | Corydon76-dig | russellb: I say much, because I don't know how much, really |
| 22:23.35 | gitguy | Corydon76-dig: what about if joe comes and wants to improve asterisk, can he? |
| 22:23.35 | gitguy | Corydon76-dig: or he has to give his rights to digium? |
| 22:23.50 | russellb | you can improve it all you want without granting the license to digium |
| 22:23.52 | Corydon76-dig | gitguy: He only has to grant a license |
| 22:23.54 | russellb | if you want it included ..... |
| 22:23.58 | russellb | o.O |
| 22:24.02 | ManxPower | gitguy: joe can do anything he wants with Asterisk, but if he wants his code in the OFFICIAL DIGIUM SOURCE, then he must license the code to Digium |
| 22:24.02 | russellb | nm, then |
| 22:24.19 | *** join/#asterisk gitguy (n=diego@adsl-128-132.click.com.py) |
| 22:24.30 | ManxPower | (5:24:01 PM) ManxPower: gitguy: joe can do anything he wants with Asterisk, but if he wants his code in the OFFICIAL DIGIUM SOURCE, then he must license the code to Digium |
| 22:24.31 | Corydon76-dig | gitguy: He only has to grant a license |
| 22:24.37 | errr | Im using the manager to Originate a call. For my channel I am using a ring group I created. for my exten I am using a number provided by a web interface which has for example my home number. If my ring group doesnt answer I want for the exten to still be called anyway but I want their voicemail message to be played to the person who answers the exten so they can choose to leave a message. How can I make the voicemail message wait to be played for the exten to be |
| 22:24.51 | gitguy | Corydon76-dig: ah |
| 22:25.24 | Corydon76-dig | gitguy: you're heavily into IP law, which is probably why there is so much confusion |
| 22:25.31 | gitguy | Corydon76-dig: why do you have to sign a license? |
| 22:25.41 | Corydon76-dig | These are NOT intuitive concepts |
| 22:25.54 | gitguy | Corydon76-dig: i'm not into any kind of laws |
| 22:25.58 | Corydon76-dig | gitguy: purely for legal reasons |
| 22:25.59 | joe | but you can have your patches be gpl I thought? |
| 22:26.08 | mchou | <PROTECTED> |
| 22:26.08 | mchou | <PROTECTED> |
| 22:26.26 | sumasuma | Digium is selling business editions |
| 22:26.31 | sumasuma | they make money out it |
| 22:26.34 | gitguy | Corydon76-dig: i'm not into kind of IP laws... i'm not a lawyer too, i don't care about laws... |
| 22:26.39 | sumasuma | with the code which you have written |
| 22:26.47 | gitguy | Corydon76-dig: i care about free software |
| 22:26.55 | joe | unless it's part of the proprietary license part of the code base |
| 22:26.58 | Corydon76-dig | sumasuma: and the money goes back into open source Asterisk |
| 22:27.06 | ManxPower | gitguy: you have to sign a license so you can't go back and sue digium for using your code |
| 22:27.28 | sumasuma | Corydon76-dig: that is next, but the reason of the disclaimer or sign a license is for the same |
| 22:27.29 | errr | 3 |
| 22:27.35 | gitguy | ManxPower: that's fine |
| 22:27.47 | sumasuma | Corydon76-dig: are you saying digium is not making profit out of it ? |
| 22:28.00 | mchou | lol |
| 22:28.15 | Corydon76-dig | sumasuma: technically, Business Edition is about selling support |
| 22:28.15 | ManxPower | gitguy: you can always use your code in any way you see fit -- you own the copyright. But Digium wants to be protected when they take your code and put it in Asterisk. |
| 22:28.38 | gitguy | ManxPower: ok |
| 22:29.03 | Corydon76-dig | sumasuma: the whole reason for making it separate is to ensure that there is a specific set of code to support, so we're not supporting a moving target |
| 22:29.16 | russellb | it's the same code, though |
| 22:29.19 | russellb | nothing special about it |
| 22:29.32 | russellb | others could make packages that they support, too |
| 22:29.42 | Corydon76-dig | sumasuma: or worse, spending 3 hours only to discover that someone has modified their Asterisk in ways that were unsafe |
| 22:29.46 | russellb | (well, we added license control, but that's about it) |
| 22:30.07 | Corydon76-dig | sumasuma: that is truly the only reason |
| 22:30.56 | sumasuma | Corydon76-dig: http://www.digium.com/en/products/software/abe.php |
| 22:30.59 | sumasuma | this says more |
| 22:31.12 | ManxPower | What I wish we could get is ABE in source form, without support. |
| 22:31.24 | Corydon76-dig | sumasuma: Honestly, it's marketing. |
| 22:31.36 | ManxPower | ABE is the only version that seems to go thru strict regression testing and control |
| 22:31.55 | ManxPower | I point you to 1.4.21 for anyone that disagrees. |
| 22:32.02 | Corydon76-dig | sumasuma: people buy ABE for the support, just like they buy Red Hat for the support |
| 22:32.08 | sumasuma | ManxPower: yes, you are correct |
| 22:32.13 | sumasuma | Corydon76-dig: Nope |
| 22:32.29 | Corydon76-dig | You can get EVERYTHING you get from Red Hat Enterprise as open source |
| 22:32.43 | sumasuma | Corydon76-dig: <ManxPower> What I wish we could get is ABE in source form, without support. |
| 22:32.48 | sumasuma | did you get this statement ? |
| 22:33.03 | mchou | nice |
| 22:33.13 | anonymouz666 | Corydon76-dig: did the guy who mail the users- with the subject "1.4.21 stalls" opened a bug report for it? |
| 22:33.14 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-185-165.hsd1.wa.comcast.net) |
| 22:33.16 | Corydon76-dig | sumasuma: yep |
| 22:33.16 | frieze | it's hard. the people here on IRC tend to give you support right when you mention that you have a problem |
| 22:33.27 | frieze | denying you that "no support" option some people crave |
| 22:33.43 | Corydon76-dig | sumasuma: might actually happen after a version of ABE is EOLed, to avoid support issues |
| 22:33.47 | ManxPower | frieze: not really. If people use ABE they can't get support here. |
| 22:34.39 | frieze | really? |
| 22:34.45 | Corydon76-dig | sumasuma: but I can say 'might' all day long, as I don't get to make those decisions |
| 22:34.45 | frieze | hmm |
| 22:34.49 | ManxPower | frieze: we are UNABLE to support it here. |
| 22:35.11 | frieze | ahh |
| 22:35.13 | sumasuma | Corydon76-dig: got you |
| 22:35.20 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 22:35.22 | frieze | in any event, it's burger time for me |
| 22:35.24 | Corydon76-dig | ManxPower: why would somebody with Red Hat Enterprise seek volunteer support when they can get professional support? |
| 22:35.28 | frieze | look out shake shack, here I come! |
| 22:35.44 | ManxPower | Corydon76-dig: I have no idea? |
| 22:35.45 | errr | Corydon76-dig: it happens in #rhel all the time |
| 22:36.05 | mchou | Corydon76-dig: speed |
| 22:36.19 | ManxPower | Corydon76-dig: is Digium support open 24/7/365? |
| 22:36.26 | Corydon76-dig | mchou: ABE customers already get priority support |
| 22:36.39 | Qwell | ManxPower: 24/7/365 can be paid for, afaik. |
| 22:36.40 | Corydon76-dig | ManxPower: Officially or unofficially? |
| 22:36.55 | errr | IRC never rests :) |
| 22:37.00 | ManxPower | Corydon76-dig: that is a very silly question |
| 22:37.24 | Corydon76-dig | errr: I dunno, have you tried to get support in here at 4 am Central Time? |
| 22:37.34 | Corydon76-dig | Most of us are in bed by then |
| 22:37.37 | Strom_C | I'm awake sometimes |
| 22:37.40 | Qwell | most? |
| 22:37.43 | Qwell | like...half |
| 22:38.08 | errr | Corydon76-dig: nah I am sleeping then, but other rooms are booming then |
| 22:38.09 | ManxPower | Corydon76-dig: How about this: Can you call your ABE support phone number 24/7 and get a person? |
| 22:38.17 | Qwell | ManxPower: If you have 24/7 support. |
| 22:38.23 | Strom_C | but yeah, people come in here all the time looking for install support with their digium hardware, and some of them steadfastly refuse to call and use the support they've already paid for. It makes no sense to me. |
| 22:38.25 | ManxPower | so the answer is "yes" |
| 22:38.28 | Corydon76-dig | ManxPower: depends on the level of support you paid for |
| 22:38.31 | Qwell | ABE does not include 24/7 though, afaik. |
| 22:38.38 | Qwell | (but it can be obtained) |
| 22:39.05 | Qwell | angler: maybe you can clarify? :D |
| 22:39.07 | ManxPower | So there ARE reasons. |
| 22:40.54 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:41.28 | russellb | there are no reasons for anything |
| 22:42.04 | [TK]D-Fender | just look at the platypus! |
| 22:42.16 | Strom_C | or the Quebecois |
| 22:42.16 | errr | =) |
| 22:42.20 | [TK]D-Fender | builds another "horse by commitee" |
| 22:42.27 | *** join/#asterisk SexyKen (n=ken@c-67-160-193-210.hsd1.ca.comcast.net) |
| 22:42.48 | SexyKen | Hey all - I'm trying to setup DISA so that I can enter my callerid while on the phone... |
| 22:43.11 | SexyKen | ...in the extensions.conf - how do I make it accept digits and then place those digits into SetCallerID? |
| 22:43.25 | Strom_C | before we can answer that, you will have to prove your claim of sexiness |
| 22:43.38 | Corydon76-dig | SexyKen: you probably don't want DISA. You want to use Read(). |
| 22:43.44 | SexyKen | Hrm. ... I'm too sexy for my shirt, too sexy for my shirt. |
| 22:43.44 | errr | lol |
| 22:43.51 | [TK]D-Fender | SexyKen: "core show application read" "core show application set", "core show function CALLERID" |
| 22:43.58 | Strom_C | SexyKen: in other words...photos :) |
| 22:46.22 | jeev | Fender, sup |
| 22:48.05 | *** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com) |
| 22:51.37 | *** join/#asterisk RoyK (n=roy@ip-158-24-149-91.dialup.ice.no) |
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| 22:53.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 23:02.14 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
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| 23:07.26 | TJNII | ~sipnat |
| 23:07.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:07.54 | *** part/#asterisk fogo (n=fogo@72.8.104.15) |
| 23:11.57 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 23:18.53 | [TK]D-Fender | BBIAB |
| 23:27.15 | nauticalthinker | have you guys ever run into issues with placing an ftp server on a asterisk server? |
| 23:28.36 | Corydon76-dig | Nope |
| 23:28.55 | TJNII | I think a lot of people run * & TFTP for phone configs.... |
| 23:29.02 | Corydon76-dig | but then again, the FTP server was only for phone configs, not for large downloads |
| 23:29.23 | TJNII | Though, TFTP != FTP |
| 23:32.29 | *** part/#asterisk nuonguy (n=john@c-24-6-187-202.hsd1.ca.comcast.net) |
| 23:34.01 | *** join/#asterisk erojasv (n=erojasv@190.40.94.108) |
| 23:38.26 | *** join/#asterisk loood (n=gman@64.201.247.2) |
| 23:38.28 | loood | anyone know why i wouldn't be able to check voicemail if dtmfmode=info is set? |
| 23:42.00 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 23:50.03 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 23:54.54 | *** join/#asterisk [cfdisk] (n=cfdisk@68-116-156-85.dhcp.ftwo.tx.charter.com) |
