| 00:05.14 | JamesJRH | This 3rd party website appears to be more useful than the official Digium website: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 00:06.18 | [TK]D-Fender | Except realizing that it's talking all about * 1.2 which is over 5 years old... |
| 00:06.26 | [TK]D-Fender | More like 7 really |
| 00:06.42 | [TK]D-Fender | And things have changed. |
| 00:06.57 | [TK]D-Fender | Live CLI gives you at least the syntax relative to yours |
| 00:07.14 | file | as will the wiki |
| 00:07.58 | [TK]D-Fender | Yup, *'s official wiki does seem active... |
| 00:08.07 | [TK]D-Fender | What was the previous website issue scope exactly?> |
| 00:08.38 | file | www is completely separate |
| 00:09.10 | [TK]D-Fender | It does look like it's back no so far as I can see... |
| 00:09.13 | [TK]D-Fender | now3* |
| 00:10.03 | file | yeah I'm going to poke the relevant people tomorrow just in case |
| 00:10.55 | [TK]D-Fender | JamesJRH: https://wiki.asterisk.org/wiki/display/AST/Home <- far better place to start |
| 00:11.35 | [TK]D-Fender | JamesJRH: voi-info is mostly dated and needs to be taken with a brick of salt |
| 00:12.13 | JamesJRH | [TK]D-Fender: Aah! Thank you. |
| 00:12.26 | JamesJRH | file: I did search the wiki before but got lost. |
| 00:14.50 | [TK]D-Fender | Start with the BOOK... then use the official wiki as reference for the technical bits |
| 00:14.52 | [TK]D-Fender | ~book |
| 00:14.53 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:14.55 | JamesJRH | [TK]D-Fender: The www seems to come and go. Different pages seem to be mangled at any one time. I tried back a few times over an hour the other day, and it would changed a couple of times. |
| 00:14.55 | [TK]D-Fender | ^^^ |
| 00:17.04 | file | I'll nudge tomorrow |
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| 00:20.14 | JamesJRH | Also annoying, is that many of the things I clicked on on the Asterisk homepage when I could get it to work would lead me to a form (or a mangled dialogue depending on your luck). For example, ‘Watch the Video’ pops-up a form or pops-up what looks like a binary file. |
| 00:20.20 | ChannelZ | it is/was spitting out gzipped pages but seemingly without saying it has in the header |
| 00:20.42 | file | let's look through my email and see if I knew of anything... |
| 00:21.10 | JamesJRH | ChannelZ: But it's on and off. |
| 00:21.21 | ChannelZ | Yup |
| 00:21.21 | file | nada |
| 00:21.30 | JamesJRH | Maybe there are multiple servers and only one is affected. |
| 00:22.46 | ChannelZ | yeah I never got that far into poking around, it was like 4am and I had to go to bed. Seemed like the same IP though. |
| 00:23.17 | file | it is certainly possible |
| 00:23.19 | ChannelZ | though could still be a misbehaving CDN or some caching |
| 00:25.03 | JamesJRH | Yes. |
| 00:25.22 | JamesJRH | Though that is still multiple servvers. ;-) |
| 00:25.28 | JamesJRH | servers* |
| 00:25.55 | ChannelZ | yeah I just meant 'behind the IP I can see' |
| 00:26.56 | file | it's interesting |
| 00:27.22 | file | it is using gzip encoding, HTTP headers are fine, it's hitting the drupal cache, and it's returning an HTML document but the body is nonsense |
| 00:27.56 | JamesJRH | ChannelZ: Okay, yes, so you may be seeing multiple servers through one IP. I see. |
| 00:27.59 | file | unless this is lying to me, which is also possible |
| 00:28.56 | ChannelZ | file: I was doing some wgets last night and the barf it was spewing was definately gzip data, I could unzip it and get the proper html |
| 00:29.06 | JamesJRH | file: What's “this”? |
| 00:29.23 | ChannelZ | but then pages were working I was trying to check and then I went to bed. ;) |
| 00:29.27 | file | web dev tools |
| 00:30.52 | JamesJRH | file: If the tools are in a browser, it could be due to some dynamic HTML fixup that the browser is doing. |
| 00:31.08 | file | browser extensions, actually |
| 00:31.32 | ChannelZ | I can't get anything to not work now |
| 00:31.44 | JamesJRH | So yeah, they'll be affected by the browser's fixup probably. |
| 00:31.54 | file | http://www.asterisk.org/products/ip-phones |
| 00:32.01 | JamesJRH | ChannelZ: Me neither. :-/ |
| 00:32.05 | file | that does not work for me. |
| 00:32.29 | ChannelZ | yup here either. |
| 00:32.53 | JamesJRH | file: Thanks. |
| 00:33.54 | ChannelZ | So it's gzip data, but I see yes the headers claim so. I wonder if it's somehow being double-gzipped? |
| 00:34.11 | file | quite |
| 00:34.13 | ChannelZ | like it's getting cached compressed for some reason |
| 00:34.29 | pabelanger | HACK THE PLANET |
| 00:42.11 | JamesJRH | file: When you said “but the body is nonsense”, that made me initially think that the browser had fixed-up a HTML <head> and enclosed the gzip file with <body> tags. I've seen similar things happen before, but I see no fixup here. I wonder if you were actually talking about the ‘body’ of the HTTP transfer if that's what it's called? |
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| 00:46.00 | JamesJRH | Hmm, ‘body’ is also the correct term in HTTP, just to be confusing. :-/ |
| 00:46.42 | WIMPy | doesn't see anything unusual. |
| 00:47.27 | JamesJRH | WIMPy: This page seems to be gone now: http://www.asterisk.org/products/ip-pbx |
| 00:47.52 | WIMPy | yes |
| 00:47.57 | JamesJRH | (The other one is back.) |
| 00:48.26 | ChannelZ | There's a glitch in the matrix |
| 00:48.36 | JamesJRH | :-O |
| 00:50.35 | WIMPy | Interesting. If I use wget, I get HTML. With a browser I get gzipped HTML. |
| 00:51.28 | WIMPy | lynx -dump works, but not without -dump. |
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| 04:32.20 | FennecVox | So, I've obtained an AA50 from a thrift store. How obsolete is it? |
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| 05:28.22 | [TK]D-Fender | <PROTECTED> |
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| 06:09.21 | FennecVox | [TK]D-Fender: Like, not worth messing with old, or "don't connect it straight to the Internet" old |
| 06:09.35 | FennecVox | aw, he left |
| 06:09.36 | FennecVox | damn |
| 06:09.50 | FennecVox | figured out what files should've been on the missing CF card though! |
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| 08:59.13 | bitwize | Hi guys, does anybody know where I can get hold of audio files with normal beeping ring tones? |
| 09:05.51 | Chainsaw | bitwize: Do you mean progress tones? |
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| 09:13.26 | andi | Hi |
| 09:14.02 | andi | I'm trying to setup faxreceive for one of my numbers. I get this error: Cannot reserve FAX session - session limit exceeded (max: 0). |
| 09:14.36 | andi | After some googling I found out that there is any license needed. Can you tell me more about this license and what this is and where I can get it? |
| 09:17.40 | Chainsaw | andi: You can get a single free license from this page: http://www.digium.com/en/products/software/fax-for-asterisk |
| 09:19.13 | Chainsaw | andi: The explanations are further down the page. |
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| 09:31.56 | Chainsaw | bitwize: Did you mean progress tones? |
| 09:35.43 | bitwize | Chainsaw: yes - progress tones |
| 09:35.48 | bitwize | (sorry for late reply) |
| 09:36.12 | Chainsaw | bitwize: Generally you have Asterisk generate these as required (see indications.conf) rather then playing pre-recorded files. |
| 09:38.29 | bitwize | Chainsaw: Thanks, this setup is a little bit different doh. I need to start/stop playing (progressing) ringtones into a 3rd party conference room (app_konference) with already connected channels |
| 09:40.41 | andi | Chainsaw: Thanks, it's working... Until the point where my sipgate trunking refuses T.38. |
| 09:45.42 | Chainsaw | bitwize: core show application Playtones. |
| 09:46.03 | Chainsaw | bitwize: Or you can use the r argument to Dial. |
| 09:46.44 | bitwize | Chainsaw: Thank, I will look into that right away |
| 09:49.26 | Chainsaw | bitwize: That and the Ringing, Congestion & Progress applications. |
| 09:49.39 | Chainsaw | bitwize: Between those 5 options I'm sure you'll find one that is just right :) |
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| 09:52.52 | zamba | has MeetMe been removed? |
| 09:56.56 | Chainsaw | Morning Faustov. |
| 10:04.50 | kaldemar | zamba: no. |
| 10:13.23 | Faustov | moin mr Chainsaw |
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| 11:19.11 | Ice_Strike | Anyone got experience with PoE switches for the phones? |
| 11:19.16 | Ice_Strike | Is it stable |
| 11:19.43 | WIMPy | As stable as your cable :-) |
| 11:19.50 | WIMPy | Why shouldn't it? |
| 11:20.29 | Ice_Strike | Just thought power and data going through a cable can be sneeky |
| 11:20.29 | zamba | kaldemar: how can i check if it's there? |
| 11:20.30 | Ice_Strike | maybe not. |
| 11:20.56 | zamba | kaldemar: [2014-03-31 13:20:44] WARNING[8268]: pbx.c:4218 pbx_extension_helper: No application 'MeetMe' for extension (sentralbord, 8150, 5) |
| 11:21.07 | Ice_Strike | WIMPy My understanding that if I get PoE switches, the phones will not longer have power pin connected? :) |
| 11:21.36 | WIMPy | There are many ways to power your phones. |
| 11:21.47 | Ice_Strike | I want to reduce the plugs. |
| 11:22.03 | Ice_Strike | power sockets |
| 11:22.14 | WIMPy | zamba: You need dahdi for meetme. Or you forget about the old MeetMe and switch to the new ConfBridge. |
| 11:22.39 | Ice_Strike | WIMPy What are other way? |
| 11:23.07 | WIMPy | Ice_Strike: My preferred way, although it requires some custom cabling, is to get the power for the phone from the PC. |
| 11:24.16 | Ice_Strike | WIMPy Is there already made that do that? |
| 11:24.22 | WIMPy | Allthough you can question why you need a phone if you have a PC. |
| 11:24.29 | WIMPy | Err, what? |
| 11:24.44 | zamba | WIMPy: is that a drop-in replacement? |
| 11:25.24 | WIMPy | zamba: No. It's different, but it can do a lot more. And it doesn't need DAHDI. |
| 11:25.25 | Ice_Strike | WIMPy I meant can I buy cable from somewhere that get the power for the phone from the PC |
| 11:25.45 | davlefou | hi, it works |
| 11:26.29 | WIMPy | Ice_Strike: I don't think so. I have seen slot brackets with a power plug being sold. |
| 11:26.46 | WIMPy | davlefou: What works? |
| 11:27.24 | davlefou | my agi script, i have an question but i find the soluce before i have write it |
| 11:27.43 | WIMPy | Nice |
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| 11:28.18 | pbxman | hello |
| 11:29.26 | pbxman | can anybody get into this url-> https://jai-imageio.dev.java.net/binary-builds.html#Release_builds??? how can I download the JAI_core and jai_codec libraries, the maven repos don't seem to have permissions to distribute this libraries? any ideas? |
| 11:30.05 | pbxman | sorry wrong channel |
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| 11:35.42 | pbxman | all that was to send a FAX i need to parse pdfs and stuff to tiffs, does anybody know any other alternatives to SendFax that allow other formats than tiff? |
| 11:37.48 | WIMPy | Fax is tiff, so it has to be converted. |
| 11:41.03 | pbxman | I think I'm gonna use GhostScript to convert that PDF into TIFF, I wanted to do it without calling the OS |
| 11:42.00 | davlefou | Somes one have already use SIPAddHeader with an softphone? i try to use it with sflphone. |
| 11:44.39 | computer22 | Is there any good website comparing voip phones? |
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| 11:50.40 | davlefou | computer22, no |
| 11:51.48 | Ice_Strike | what do you think of this switch http://www.misco.co.uk/product/195247/Netgear-ProSafe-48-Port-Gigabit-PoE-Smart-Switch |
| 11:52.24 | WIMPy | Why do you need gigabit to connect phones? |
| 11:53.07 | Ice_Strike | Not just phones |
| 11:53.11 | Ice_Strike | computers as well |
| 11:53.53 | WIMPy | You want to take yourselt the time to make sure they coexist on the same switch? |
| 11:54.32 | Ice_Strike | What do you mean |
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| 11:55.27 | WIMPy | If you use the same network for phones and PC you will have to do some traffic management. |
| 11:55.50 | Ice_Strike | Yes, we use same network - no vlan. |
| 11:55.57 | Ice_Strike | Will do traffic management. |
| 11:56.41 | WIMPy | It's probably a lot cheaper to just use two seperate switches. |
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| 11:57.21 | Ice_Strike | 2 x 24 you mean |
| 11:57.33 | Ice_Strike | I would need two x 48 switches |
| 11:57.36 | WIMPy | Whatever size you need. |
| 11:57.58 | WIMPy | But one with gigabit and one with POE should be cheaper than one with both. |
| 11:58.19 | WIMPy | And you don't have to do the traffic manangement. |
| 11:58.53 | Ice_Strike | WIMPy Than that mean I would need seperate network for computers |
| 11:59.01 | Ice_Strike | I like phones to be chained with computer |
| 11:59.32 | Ice_Strike | Switch <> A Network Cable <> Phone <Chained> computer |
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| 12:00.58 | WIMPy | Good luck with the phones handling traffic management then. |
| 12:02.44 | Ice_Strike | We don't have enough network ports on the wall to keep seperate for the phones. |
| 12:03.13 | Ice_Strike | We have about 90 computers and 100 ports. |
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| 13:03.57 | file | for those who were seeing asterisk.org problems - I've sent an email off to the right person |
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| 13:32.38 | Groenleer | anyone here who could help me with an queue problem on an old asterisk build 1.8 (which i can't upgrade) |
| 13:33.11 | Groenleer | The queue returns an 'UNKNOWN' status, and i need to reroute the call to an external number which doesn't seem to happen right now. |
| 13:34.00 | Chainsaw | Ice_Strike: If you're after relatively cheap gig PoE switches I'd look at ProCurve. |
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| 13:34.05 | cusco | hi folks |
| 13:34.10 | Chainsaw | Ice_Strike: Rather then NetGear. |
| 13:34.13 | cusco | say.. in asterisk cli: features show |
| 13:34.15 | Chainsaw | Good afternoon Cubber. |
| 13:34.18 | cusco | shows default and current |
| 13:34.20 | Chainsaw | And hi cusco. |
| 13:34.30 | cusco | hi Chainsaw :) |
| 13:34.39 | cusco | so, that means default is replaced by 'current' |
| 13:34.40 | cusco | right? |
| 13:34.56 | cusco | I'm wondering why I'm getting the transfer feature everytime I press # on a outbound call |
| 13:35.18 | cusco | and it hit me, that it might be asterisk-gw ... checking... |
| 13:35.32 | Chainsaw | Groenleer: You're on debug 10 & verbose 10, and still no information on the actual queue event is forthcoming? |
| 13:35.52 | Groenleer | well the queue is empty (no members) and has leave on empty on. |
| 13:36.14 | Groenleer | so that is why it is 'UNKNOWN' the verbose is at 39, debug i don't know |
| 13:36.17 | Ice_Strike | Chainsaw Why? |
| 13:36.32 | Ice_Strike | Oh |
| 13:36.48 | Ice_Strike | Missed your msg.. you suggested ProCurve |
| 13:36.54 | Chainsaw | Ice_Strike: Correct. |
| 13:37.08 | Ice_Strike | I thought ProCurve is more expensive than Netgear |
| 13:37.17 | Chainsaw | Ice_Strike: Do you think there is a reason for that? |
| 13:37.33 | Groenleer | I would rather go with ProCurve than Netgear ;) |
| 13:37.34 | cusco | hmm there is no features.conf .. how does one disable features? |
| 13:38.04 | cusco | only module with features in its name is bridge_builtin_features.so |
| 13:38.46 | [TK]D-Fender | [09:37]cuscohmm there is no features.conf .. how does one disable features? <- only reason it's there is because you specified the dial options |
| 13:38.56 | [TK]D-Fender | "core show application dial" <- |
| 13:39.05 | cusco | right, the T or t option |
| 13:41.27 | cusco | thank you TK |
| 13:43.52 | Ice_Strike | Chainsaw ProCurve POE 48 ports costing £5,279.96 http://www.broadbandbuyer.co.uk/Shop/ShopDetail.asp?ProductID=8425 |
| 13:44.05 | Ice_Strike | Or Im I looking at wrong one? |
| 13:44.28 | Chainsaw | Ice_Strike: You're comparing an unmanaged Netgear against a managed ProCurve? |
| 13:44.49 | Ice_Strike | Did I? |
| 13:44.50 | Ice_Strike | lol |
| 13:45.15 | Chainsaw | Ice_Strike: Yes, you did. |
| 13:45.35 | Ice_Strike | Netgear is Managed |
| 13:45.57 | Ice_Strike | NETGEAR ProSafe GS752TP Gigabit Smart Switch - switch - 48 ports - Managed - desktop, rack-mountable |
| 13:47.37 | Chainsaw | Ice_Strike: If you want to buy a cheap Taiwanbox, I cannot stop you. |
| 13:47.47 | Ice_Strike | :) |
| 13:47.49 | Groenleer | Ok, made sure debug was at level 10, but does not return any useful information to me. |
| 13:48.14 | Chainsaw | Ice_Strike: Look at a something like an Extreme Networks X450-48p and get it refurb. |
| 13:48.15 | *** join/#asterisk alami (~ialami@unaffiliated/alami) |
| 13:48.43 | [TK]D-Fender | Groenleer: show us the queue dump prior, and the call itself at verbose 10. |
| 13:48.46 | [TK]D-Fender | ~pb |
| 13:48.47 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:48.48 | [TK]D-Fender | ^^^ |
| 13:49.01 | Ice_Strike | I have a look |
| 13:49.24 | Chainsaw | Ice_Strike: That's actually better than any ProCurve, but because it doesn't have the brand recognition you might luck out on price. |
| 13:49.37 | [TK]D-Fender | Ice_Strike: the netgear will probably work fine. I've done well with D-Link for PoE switches as well, and there are plenty of Linksys that'll do the job |
| 13:50.11 | Chainsaw | [TK]D-Fender: Anything involving eternal handset reboots or intermittent packet loss goes to you. |
| 13:50.53 | Ice_Strike | PoE switches will be used for the phones which will be chained to computers |
| 13:51.05 | Ice_Strike | not a seperate network. |
| 13:51.17 | Groenleer | http://pastebin.com/AE2Lz127 <== is this enough? I am quite new to asterisk debugging, so still learning all the commands. |
| 13:51.18 | Kobaz | http://cdn.themetapicture.com/media/funny-quote-think-while-its-legal.jpg |
| 13:51.34 | Groenleer | not sure i have the queue dump etc. |
| 13:51.46 | [TK]D-Fender | Chainsaw: Never had any issues |
| 13:52.05 | [TK]D-Fender | Chainsaw: Why do you think a local LAN switch is just going to lose packets out of nowhere? |
| 13:52.10 | [TK]D-Fender | This isn't the open internet... |
| 13:53.00 | Chainsaw | [TK]D-Fender: Because he's hanging a whole office on it, and a Netgear doesn't have the backplane to carry all that reliably. |
| 13:53.10 | [TK]D-Fender | Groenleer: that dialplan bit was useless, and I don't see the queue dump... |
| 13:54.18 | Ice_Strike | Chainsaw I found this: http://www.broadbandbuyer.co.uk/Shop/ShopDetail.asp?ProductID=13244 |
| 13:54.37 | Ice_Strike | This seem to be managed right? |
| 13:54.58 | [TK]D-Fender | Ice_Strike: features 24 PoE+ port <---------------- |
| 13:55.12 | Ice_Strike | Yep |
| 13:55.14 | Ice_Strike | Ohh |
| 13:55.28 | Chainsaw | Ice_Strike: Yes, but only half the ports are PoE, and they max out at 100mbit/sec. |
| 13:56.21 | Ice_Strike | Yep |
| 13:56.26 | Ice_Strike | What do you think of this: http://www.misco.co.uk/product/195049/ZyXel-GS1910-48-48-Port-Gigabit-PoE-Smart-Switch |
| 13:56.31 | Ice_Strike | ZyXel GS1910-48 48 Port Gigabit PoE Smart Switch |
| 13:58.01 | Chainsaw | Ice_Strike: First non-horrendous one you've posted. |
| 13:58.47 | Chainsaw | Ice_Strike: It's not a mainstream brand, but I would trust them for an office deployment. |
| 13:59.17 | Ice_Strike | Aha |
| 14:00.24 | WIMPy | ZyXEl used to be the top end many years ago. Now they feel more like the dollar store of computing. |
| 14:02.07 | *** join/#asterisk leifmadsen (~free-mads@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:02.07 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:02.39 | Chainsaw | Ice_Strike: Value for money is quite good on that one, and it ticks all the boxes. Go with that over a Netgear. |
| 14:04.20 | Ice_Strike | I need to see if include any type of loop protection. |
| 14:05.37 | Groenleer | eitherway, it seems that the queue dump should be written to /var/log/asterisk/queue_log, but it isn't for the calls i am trying to fix. However, when agents are in the queue, the log file is being used. |
| 14:05.41 | Chainsaw | Ice_Strike: It does. |
| 14:06.51 | Ice_Strike | In the past, one of the agent plugged a cable from port to port on the wall. |
| 14:07.12 | Ice_Strike | Whole network went down, took a while to figure out what went wrong. |
| 14:09.18 | [TK]D-Fender | Groenleer: No... the queue STATUS dump... "queue show" <- |
| 14:09.37 | [TK]D-Fender | Groenleer: And your actual queue config. Youa re stating probelms with them and we're not actually looking at it yet |
| 14:13.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:13.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:15.08 | Groenleer | ok, sorry my misunderstanding. |
| 14:16.16 | *** join/#asterisk valentinmu (~webdev@91.197.19.104) |
| 14:16.29 | valentinmu | hi guys |
| 14:16.41 | valentinmu | need help with compiling from source |
| 14:17.13 | valentinmu | lock.o:/usr/src/asterisk-11.8.1/utils/lock.c:1239: more undefined references to `ast_bt_get_addresses' follow |
| 14:18.20 | valentinmu | as i understand, "ast_bt_get_addresses" is a part of backtrace structure that is obviously necessary for asterisk |
| 14:18.46 | valentinmu | but i don't know how to fix problem |
| 14:20.05 | *** join/#asterisk imox (~imox@p57A97291.dip0.t-ipconnect.de) |
| 14:22.14 | WIMPy | 11.8.1? Maybe you should get a sensible version first. |
| 14:22.36 | WIMPy | Oh, shit. Cahn't read. |
| 14:22.41 | WIMPy | Even when copying. |
| 14:22.55 | WIMPy | Multitasking broken :-( |
| 14:25.58 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:27.01 | valentinmu | WIMPy: m? sensible version? |
| 14:27.06 | *** join/#asterisk jploh (~textual@49.146.166.169) |
| 14:27.26 | WIMPy | Just forget what I wrote. |
| 14:27.33 | WIMPy | That wa complete rubbish. |
| 14:27.40 | Groenleer | ok, tried to gather some info, still not sure this is all that is needed for the troubleshooting (i am learning). http://pastebin.com/q07Y6p6E |
| 14:28.30 | Groenleer | The issue is the queue returns with UNKNOWN status (which seems valid based on the config) and i need to redirect the calls if the queue returns UNKNOWN, or any other error |
| 14:28.41 | valentinmu | WIMPy: :) so you have nothing to advice? |
| 14:29.14 | WIMPy | Doesn't seem to be a good day to get advice from me. |
| 14:29.23 | [TK]D-Fender | Groenleer: Your queue has NO members and you have "joinempty = no". Your caller cannot join your queue. |
| 14:29.26 | WIMPy | So I better shut up. |
| 14:29.29 | valentinmu | WIMPy: ok %) |
| 14:31.54 | Groenleer | [TK]D-Fender, true, i need to figure out how to redirect a caller if a queue has no members |
| 14:32.24 | [TK]D-Fender | Groenleer: You aren't looking at your dilaplan near where you call the queue at all. |
| 14:32.26 | Groenleer | actually i should try to prevent them getting in the queue if there is no one, but i have no clue how to. |
| 14:32.37 | [TK]D-Fender | Groenleer: thatIVR context JUMPS to another context. |
| 14:32.42 | [TK]D-Fender | ~book |
| 14:32.42 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:32.43 | [TK]D-Fender | ^^^^^^^^^^ |
| 14:32.56 | [TK]D-Fender | Time to actually learn the Asterisk dialplan.... |
| 14:33.00 | *** join/#asterisk evilman_home (~evilman_h@89-179-77-66.broadband.corbina.ru) |
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| 14:56.28 | *** join/#asterisk eth00 (~John@unaffiliated/eth00) |
| 14:59.20 | eth00 | I am running asterisk 1.8 and trying to use an AGI perl script for outbound calls. I have it calling but it immediately hangs up. I straced the process and when asterisk tries to execve the perl file it gets: -1 EACCES (Permission denied) . Permissions are fine and using sudo -u asterisk-user perl-file.pl works fine. The AGI debug does not give any errors. Any ideas on where to look? |
| 14:59.30 | eth00 | I appreciate any help or being pointed in the right direction! |
| 15:07.28 | alami | hello, i want to set the language for asterisk is de, so i have download the language package and place it into /var/lib/asterisk/sounds/de, in sip.conf->general i set Langiage = de |
| 15:07.52 | alami | also at extension.conf with the Setlanguage func.. but still not work |
| 15:12.17 | alami | i'm using asterisk 11.8.1 |
| 15:14.50 | [TK]D-Fender | ~pb |
| 15:14.50 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 15:14.52 | [TK]D-Fender | ^^^ |
| 15:14.56 | [TK]D-Fender | alami: Show us |
| 15:22.37 | alami | http://pastebin.com/KceNGThA |
| 15:26.07 | *** join/#asterisk evilman_home (~evilman_h@89-179-77-66.broadband.corbina.ru) |
| 15:26.59 | [TK]D-Fender | same => n,Set(LANGUAGE()=de) <- this is not valid the way you did it. |
| 15:27.15 | [TK]D-Fender | There is nothing to be the "same" as, and it is in the CHANNEL() function |
| 15:28.23 | alami | [TK]D-Fender>: sorry that was only a copy paste mistake |
| 15:29.41 | [TK]D-Fender | Show actual configs, and actual cal debug. |
| 15:29.43 | [TK]D-Fender | call* |
| 15:30.47 | alami | [TK]D-Fender: Sorry i'm new to asterisk, wich configs do you mean? and wich debug mode? |
| 15:30.53 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 15:31.22 | [TK]D-Fender | alami: What did you mean by "a copy paste mistake"? |
| 15:31.42 | [TK]D-Fender | alami: and for debug, show us an actual call at * CLI, "core set verbose 10" |
| 15:31.45 | *** join/#asterisk jiku (~jiku@182.71.136.242) |
| 15:34.26 | alami | here it's http://pastebin.com/CFXsS4Vy |
| 15:35.17 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:35.49 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 15:38.05 | [TK]D-Fender | http://pastebin.com/KceNGThA <--- [genereal] is wrong. |
| 15:38.40 | [TK]D-Fender | http://pastebin.com/CFXsS4Vy <--- extend => _.,,Hangup() extend => _.,,Set(LANGUAGE()=de) <- both never used and multiple mistakes in there |
| 15:39.17 | alami | [TK]D-Fender: when i put at user language = de is working |
| 15:39.35 | [TK]D-Fender | <PROTECTED> |
| 15:39.39 | [TK]D-Fender | and broken |
| 15:42.14 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
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| 15:52.41 | alami | [TK]D-Fender:i can set up language for each user, but not generaly |
| 15:53.58 | [TK]D-Fender | alami: Ok, I've told you twice, I'm going to try one last time and hopefully it will be clear this time . ---> YOU SPELLED [general] WRONG <--- |
| 15:54.11 | [TK]D-Fender | [11:39][TK]D-Fender[11:38][TK]D-Fenderhttp://pastebin.com/KceNGThA <--- [genereal] is wrong. <-- MIS-SPELLED |
| 15:54.28 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 15:55.27 | alami | [TK]D-Fender: i understand that only if you explain it one time, and i have also change context=users |
| 15:55.36 | [TK]D-Fender | alami: Your configs are full of spelling and other syntax errors. You need to pay more attention to what you are doing. |
| 15:56.08 | [TK]D-Fender | alami: What you've shown should give the results we wee what is a lack of gloabl language settings |
| 15:58.18 | [TK]D-Fender | global* |
| 15:59.13 | *** join/#asterisk hellc2 (~elio@100.Red-88-26-250.staticIP.rima-tde.net) |
| 15:59.24 | hellc2 | Hi everyone! |
| 16:02.01 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 16:03.30 | alami | [TK]D-Fender: and now http://pastebin.com/diZ91UFx |
| 16:03.51 | [TK]D-Fender | root@tktest:/etc/asterisk# cat sip.conf |
| 16:03.52 | [TK]D-Fender | [genereal] <- NO |
| 16:04.03 | hellc2 | I have a newbie question that I can't fix it: I have created a sample queue with 2 SIP members SIP/A and SIP/B. I'm trying to call to this queue, it rings A during 5 secs, B durings 5 secs, A during 5 secs, and go on forever... |
| 16:04.25 | hellc2 | my queue couldn't be more simple.. |
| 16:04.54 | hellc2 | but always get 5 seconds of wait between A to B... and B to A... |
| 16:05.05 | [TK]D-Fender | Sure it could.... remove B from the picture :) |
| 16:05.38 | [TK]D-Fender | hellc2: PASTEBIN your queue config. |
| 16:05.39 | [TK]D-Fender | ~pb |
| 16:05.40 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:05.41 | [TK]D-Fender | ^^^^^^^^^ |
| 16:06.15 | hellc2 | http://pastebin.com/2FECDanj |
| 16:07.00 | *** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net) |
| 16:07.52 | [TK]D-Fender | hellc2: and the call attempt as well: |
| 16:08.04 | [TK]D-Fender | "core set verbose 10" |
| 16:08.30 | [TK]D-Fender | hellc2: While you're at it, the full queue config file |
| 16:08.46 | [TK]D-Fender | hellc2: that looks a little thin and probably has general overrides |
| 16:08.51 | alami | [TK]D-Fender: i don't understand what's wrong at [general] on sip.conf |
| 16:09.36 | [TK]D-Fender | [12:03]alami[TK]D-Fender: and now http://pastebin.com/diZ91UFx <- LINE # 23 IS STILL WRONG. |
| 16:10.54 | alami | ohhh now i get it |
| 16:11.05 | alami | ohh my god sorry |
| 16:11.09 | alami | Fender |
| 16:11.56 | hellc2 | [TK]D-Fender, queues.conf: http://pastebin.com/WKUAm9aK |
| 16:12.02 | hellc2 | [TK]D-Fender, the log: http://pastebin.com/zcg6FvRG |
| 16:12.28 | hellc2 | You can see the "seconds" between dial SIP/400 and SIP/401 |
| 16:12.53 | hellc2 | line .116 |
| 16:13.47 | [TK]D-Fender | hellc2: -- Executing [999@outgoing:1] Queue("Console/dsp", "prueba") in new stack <- this is not [testqueue] that we see in your queues.conf config |
| 16:13.56 | [TK]D-Fender | [queuetest] rather |
| 16:14.07 | hellc2 | dialplan reload.. |
| 16:14.29 | [TK]D-Fender | your CLI output does NOT match the config you showed us |
| 16:14.46 | hellc2 | right... |
| 16:14.56 | michael_work | hmmm |
| 16:14.57 | hellc2 | I try to clean sample code... |
| 16:15.22 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:15.44 | michael_work | i'm using same way to originate call. on 1.8 i use SIP_CAUSE and it works fine, on 11 i se HANGUP_CAUSE and it says that there nothing found |
| 16:16.59 | [TK]D-Fender | michael_work: "same way" .. as what? |
| 16:17.11 | michael_work | same way on 1.8 as on 11 |
| 16:17.18 | michael_work | on 1.8 with sip_cause and it works fine |
| 16:17.26 | [TK]D-Fender | Ah, I did miss the later reference |
| 16:17.35 | hellc2 | queues: http://pastebin.com/6YBN0sAQ |
| 16:17.39 | hellc2 | log: http://pastebin.com/NtLVJngj |
| 16:18.30 | hellc2 | line 116 : ringing SIP/401.. and wait 5 secs... |
| 16:18.57 | hellc2 | line 167: ringing SIP/400 ... and wait 5 secs... |
| 16:20.14 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 16:25.08 | hellc2 | these 5 seconds are the time that I don't understand why happen and how I can remove. |
| 16:30.04 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 16:34.08 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 16:34.44 | [TK]D-Fender | hellc2: try setting to 1 s |
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| 16:35.37 | ChannelZ | is puzzled.. your timeout is 5 seconds |
| 16:36.59 | hellc2 | [TK]D-Fender, it will sound SIP/400 during 1 sec... 5 secs waiting... ringing SIP/401 during 1 sec... 5 secs waiting... |
| 16:43.39 | puzzled | hi |
| 16:43.55 | hellc2 | I found the problem... |
| 16:44.09 | hellc2 | app_queue.c sets DEFAULT_RETRY to 5 |
| 16:44.51 | hellc2 | although I change retry parameter to 0, it doesn't change.. |
| 16:45.18 | hellc2 | but I change DEFAULT_RETRY parameter in source, it works. |
| 16:47.41 | puzzled | so how do I upload a patch to the Jira issue I'm filing? My eyes must be getting really bad as I don't see a patch field somewhere |
| 16:48.01 | hellc2 | If someone could confirm me it, I will send a patch |
| 16:48.06 | file | submit it and then go to More -> Attach Files |
| 16:48.30 | puzzled | file: thanks. glad I'm not going blind/bonkers :) |
| 16:50.16 | *** join/#asterisk ralphmazio (~ralphmazi@nc-76-5-180-51.dhcp.embarqhsd.net) |
| 16:50.47 | ralphmazio | If you modify sip.conf and extensions.conf can you just do a dialplan reload to activate changes? |
| 16:51.00 | Qwell | sip.conf isn't dialplan, so no |
| 16:51.19 | ralphmazio | What do you have to do to get sip.conf reread. |
| 16:51.26 | Qwell | sip reload |
| 16:51.38 | ralphmazio | Will that affect active connections? |
| 16:51.48 | Qwell | no |
| 16:51.52 | ralphmazio | thank you sir |
| 16:52.05 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:52.50 | michael_work | [TK]D-Fender, found the reason to previous question about HANGUPCAUSE but no idea for a solution yet |
| 16:54.33 | puzzled | file: do I also need to request a review of a patch I submitted? https://issues.asterisk.org/jira/browse/ASTERISK-23564 |
| 16:56.57 | file | doesn't hurt. |
| 17:06.25 | navaismo | :'( Compiled wanpipe for raspberry pi, but didn't generate the .ko kernel module, wanrouter cant start :'( :'( but wancfg_ utils exists |
| 17:12.48 | puzzled | file: thanks |
| 17:13.43 | coppice | navaismo: you have a raspberry pi with PCI slots? :-\ |
| 17:14.26 | navaismo | coppice, no, trying to use the U100 |
| 17:14.42 | coppice | I thought they canned that device |
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| 17:23.30 | navaismo | U just need to find the gcc line to generate the kernel module to see the error :'( |
| 17:23.42 | navaismo | s/U/I/ |
| 17:24.11 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 17:25.15 | drmessano | http://wiki.sangoma.com/sangoma-wanpipe-usbfxo <-- Lots of RED TEXT |
| 17:25.35 | drmessano | "You might make it work.." |
| 17:25.46 | drmessano | Glad I sold the one I had |
| 17:26.34 | navaismo | Hi drmessano |
| 17:26.43 | drmessano | hi |
| 17:26.48 | navaismo | yeah this like a challenge to do |
| 17:27.38 | drmessano | Are you using an older version of DAHDI? |
| 17:28.31 | drmessano | DAHDI 2.6 Currently not supported, please use DAHDI 2.5.0.2 or below |
| 17:28.32 | drmessano | IMPORTANT LIMITATIONS: The USB FXO drivers are only guaranteed to work on Centos Installations.Due to rapidly changing kernel USB core we have to limit the USB FXO drivers to Centos installations only. |
| 17:28.50 | navaismo | so ar i get the binaries for wancfg_* wanrouter wanpipemon but the .ko module isnt present and cant find where is the compilation section. Yes I have DAHDI 2.6.1 compiled |
| 17:29.07 | navaismo | s/ar/far/ |
| 17:29.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:30.00 | drmessano | The note says not only does it not work on 2.6 and above, but CentOS only. |
| 17:30.36 | drmessano | If you can't get the .ko module compiled, that kinda agrees |
| 17:31.48 | drmessano | I never even tried to install the one I had. I won it. Sold it for $100, which I guess is $100 more than it was worth |
| 17:31.56 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 17:32.12 | navaismo | well this isnt a normall install i have edited the source file many times, and not using the Setup sangoma script make directly |
| 17:32.20 | navaismo | LOL |
| 17:32.59 | navaismo | Im just trying to compile it like a dare, i dont have one MEGALOL |
| 17:33.09 | drmessano | Hopefully you've edited more than s/2.5/2.6/ |
| 17:35.14 | *** join/#asterisk samuelharden (~gaurav@106.51.219.85) |
| 17:37.10 | navaismo | hahaha |
| 17:37.11 | navaismo | yes |
| 17:38.48 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 17:41.42 | navaismo | but I guess this is the far way i can reached with that, everything started year ago with dahdi compilation and end here :'( |
| 17:45.44 | drmessano | SOmetimes when things dont work, they dont work |
| 17:46.04 | drmessano | A Pi probably needs a good SIP FXO ATA or small channel bank |
| 17:46.15 | drmessano | I doubt the Pi could power the U100 anyway |
| 17:49.28 | *** join/#asterisk ralphmazio (~ralphmazi@nc-76-5-180-51.dhcp.embarqhsd.net) |
| 17:51.45 | *** join/#asterisk Milarepa_ (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 17:52.03 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 17:52.09 | navaismo | yeah thats another thing the crappy usb on the pi |
| 17:52.18 | *** join/#asterisk Milarepa_ (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 17:57.12 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 18:02.18 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 18:06.45 | *** join/#asterisk ghost75 (~quassel@ipservice-092-211-047-242.pools.arcor-ip.net) |
| 18:07.20 | eirirs | a SIP Pi only need ethernet :P |
| 18:07.41 | drmessano | </obvious> |
| 18:13.18 | drmessano | Anyone here using EC2? |
| 18:18.54 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.34) |
| 18:23.13 | *** join/#asterisk igcewieling1 (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 18:24.27 | navaismo | not me to poor for that |
| 18:24.48 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 18:25.05 | igcewieling1 | Has anyone tried (or even thought about) using MulticastRTP (new in Asterisk 10?) to do a kind of "clustered" condbridge, this might avoid some of the issues with the current way of doing clustered confrences |
| 18:30.50 | *** join/#asterisk polysics (~Adium@host207-131-dynamic.9-87-r.retail.telecomitalia.it) |
| 18:30.53 | polysics | hello! |
| 18:31.03 | polysics | chacnes anyone knows the status of the ICES app? |
| 18:31.13 | polysics | the wiki is not exactly helpful |
| 18:33.47 | JamesJRH | file, ChannelZ, WIMPy: The www website seems to be fixed, though it's still broken in the sense that clicking ‘Watch Video’ gets a form. :-P |
| 18:33.54 | JamesJRH | polysics: Agreed. |
| 18:33.57 | file | ack |
| 18:34.09 | file | polysics, hasn't been touched in years but I used it for AstriDevCon and it worked fine |
| 18:34.39 | JamesJRH | polysics: I find the wiki really confusing to search or otherwise find what I'm looking for. |
| 18:35.28 | polysics | the official wiki is not bad, unless it's missing some thing like in this case :) |
| 18:39.59 | [TK]D-Fender | [14:31]polysicschacnes anyone knows the status of the ICES app? <- It's there. |
| 18:42.07 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 18:42.07 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:42.40 | navaismo | newtonr, the webrtc bug is fixed in the 11.9.0-rc1 |
| 18:43.11 | JamesJRH | polysics: I guess it's okay once you find stuff; I probably just didn't have much luck with the search function. I can probably make better use of the wiki by using ‘site:wiki.asterisk.org’ in DDG (or whatever general-purpose search engine). |
| 18:43.19 | newtonr | navaismo, nice, i'll ping the reports of the various issues and ask them all to re-test with 11.9.0-rc1, thanks for the heads up |
| 18:43.25 | newtonr | *reporters |
| 18:47.55 | *** join/#asterisk twanny796 (~twanny796@c67-109.i07-17.onvol.net) |
| 18:48.08 | JamesJRH | pabelanger, WIMPy, [TK]D-Fender, whoever else: Thank you for your help about understanding PSTN channels. |
| 18:48.19 | twanny796 | any help with configuring calling out to sipgate with asterisk? |
| 18:49.48 | navaismo | have you tried wiki of sipgate |
| 18:50.44 | [TK]D-Fender | twanny796: Show us your failure with SIP debug enabled, and your config masking only the secret |
| 18:51.08 | twanny796 | [TK]D-Fender, ok |
| 18:52.11 | twanny796 | [Mar 31 20:51:41] NOTICE[21863][C-00000003]: chan_sip.c:23018 handle_response_invite: Failed to authenticate on INVITE to '"101" <sip:101@192.168.0.2>;tag=as668e285e' |
| 18:52.20 | *** join/#asterisk [sr] (~kvirc@pal-213-228-163-73.netvisao.pt) |
| 18:52.22 | [sr] | hi |
| 18:52.48 | twanny796 | <PROTECTED> |
| 18:52.51 | [sr] | i have the most wierd situation going on, i have an voip gsm as usual, and it appears call's from nowhere, from a CID 101 |
| 18:53.03 | [sr] | the only relation i see is the internal codec number, 101 |
| 18:53.27 | twanny796 | [TK]D-Fender, are you following ;) |
| 18:53.29 | [TK]D-Fender | twanny796: PASTEBIN <- |
| 18:53.30 | [TK]D-Fender | ~pb |
| 18:53.30 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:53.32 | [TK]D-Fender | ^ |
| 18:53.40 | twanny796 | [TK]D-Fender, ok |
| 18:54.38 | *** part/#asterisk polysics (~Adium@host207-131-dynamic.9-87-r.retail.telecomitalia.it) |
| 18:55.11 | igcewieling1 | [sr]: someone is trying to hack your PBX or thinks your phone is a PBX and is trying to hack it |
| 18:56.30 | [TK]D-Fender | Or your description is bad and we have no debug to base a real opinion on. |
| 18:56.40 | [TK]D-Fender | hedges his bet.... |
| 18:58.19 | twanny796 | http://pastebin.com/28RxaBtS |
| 18:59.00 | [sr] | igcewieling1: i have the voip gsm with 2x SIM's, and only happens to one of the SIM's |
| 19:00.23 | [sr] | igcewieling1: there's no external access, only if there's some cleaver guy inside the network |
| 19:00.40 | [TK]D-Fender | twanny796: Waiting for the CALL DEBUG as requested... |
| 19:01.25 | twanny796 | [TK]D-Fender, it? in the first line of patebin? |
| 19:02.02 | twanny796 | [TK]D-Fender, no it? not ok |
| 19:02.06 | [TK]D-Fender | LINE? |
| 19:02.21 | [TK]D-Fender | ASTERISK CLI <- this should be HUNDREDS of lines show actual comm flow |
| 19:02.36 | [TK]D-Fender | "sip set debug on" <- |
| 19:02.43 | twanny796 | [Mar 31 21:02:28] NOTICE[21863][C-00000005]: chan_sip.c:23018 handle_response_invite: Failed to authenticate on INVITE to '"101" <sip:101@192.168.0.2>;tag=as5f5819e8' |
| 19:02.45 | [TK]D-Fender | "core set verbose 10" <-\ |
| 19:03.40 | twanny796 | Console verbose was OFF and is now 10. |
| 19:03.40 | twanny796 | <PROTECTED> |
| 19:03.40 | twanny796 | <PROTECTED> |
| 19:03.40 | twanny796 | <PROTECTED> |
| 19:03.40 | twanny796 | <PROTECTED> |
| 19:03.40 | twanny796 | [Mar 31 21:03:17] NOTICE[21863][C-00000006]: chan_sip.c:23018 handle_response_invite: Failed to authenticate on INVITE to '"101" <sip:101@192.168.0.2>;tag=as3346ccfb' |
| 19:03.43 | twanny796 | <PROTECTED> |
| 19:03.45 | twanny796 | <PROTECTED> |
| 19:03.47 | twanny796 | <PROTECTED> |
| 19:03.49 | navaismo | PASTEBIN!!!! |
| 19:03.51 | navaismo | Or kick |
| 19:05.48 | twanny796 | http://pastebin.com/dGDDLq7r |
| 19:08.13 | *** part/#asterisk igcewieling1 (~igcewieli@ip98-183-26-100.pn.at.cox.net) |
| 19:08.47 | ChannelZ | flicks twanny796's nipples |
| 19:09.48 | [TK]D-Fender | twanny796: [15:02][TK]D-Fender"sip set debug on" <- |
| 19:09.50 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:10.23 | navaismo | twanny796, i bet that you need to change your trunk password i already registered on my asterisk test machine |
| 19:11.43 | ChannelZ | oops |
| 19:12.53 | navaismo | unregistered already |
| 19:13.34 | navaismo | twanny796, seriously dude what the.... You are pasting your passwords |
| 19:14.06 | twanny796 | navaismo, I will change it no prob. |
| 19:14.31 | twanny796 | [TK]D-Fender, cannot catche the output, is there a way to pipe it to a file? |
| 19:15.28 | navaismo | | tee filename |
| 19:16.26 | [TK]D-Fender | twanny796: What are you using to get to the shell? |
| 19:17.11 | twanny796 | rasterisk |
| 19:17.39 | [TK]D-Fender | no, how are yougetting to the *OS* shell |
| 19:18.19 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 19:18.55 | twanny796 | bash |
| 19:19.06 | twanny796 | putty |
| 19:19.09 | [TK]D-Fender | ^ |
| 19:19.21 | twanny796 | ok |
| 19:19.40 | [TK]D-Fender | right click on the bar -> Copy All To Clipboard. DONE |
| 19:19.49 | computer22 | How easy is it to upgrade asterisk if you've compiled from src? |
| 19:20.11 | [TK]D-Fender | computer22: wipe modules folder. recompile. install. Done |
| 19:20.21 | computer22 | Woo. That's pretty straight forward. |
| 19:20.28 | computer22 | If I'm using swift, I have to recompile that too? |
| 19:20.49 | [TK]D-Fender | depends if things it relies on changed |
| 19:20.59 | computer22 | Alright, sounds good. |
| 19:24.28 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 19:24.55 | twanny796 | http://pastebin.com/Urf1DNnf |
| 19:25.05 | twanny796 | [TK]D-Fender, ^ |
| 19:25.33 | [TK]D-Fender | I am clearly not seeing the SIP DEBUG I asked for 3 times not... |
| 19:25.36 | [TK]D-Fender | now* |
| 19:26.59 | *** join/#asterisk joecool (~joecool@no-sources/joecool) |
| 19:27.59 | navaismo | ++ask |
| 19:28.29 | twanny796 | [TK]D-Fender, thank you for your time, but I am sorry I have to go, maybe I will post in the asterisk forum? |
| 19:29.22 | joecool | hey, i'm trying to set up 2 IP phones in my office with an external asterisk provider (outside the LAN), one phone has 2 extensions on 2 accounts, the other is just a single extension |
| 19:29.43 | joecool | phone with single extension always works (swissphone ip10s) |
| 19:30.06 | joecool | the phone with 2 extensions (cisco 7940) only one extension works and no inbound calls work |
| 19:30.32 | joecool | do i need to run a different UDP port for every extension? |
| 19:31.18 | joecool | i've had some weirdness where both phones ring when trying to call an extension and i'm assuming it has to do with packets getting mixed up |
| 19:32.22 | [TK]D-Fender | [15:28]twanny796[TK]D-Fender, thank you for your time, but I am sorry I have to go, maybe I will post in the asterisk forum? <- just come back here withit |
| 19:51.58 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.unitymediagroup.de) |
| 20:02.04 | jameswf | joecool: every active sip channel uses 2 udp ports |
| 20:03.21 | [TK]D-Fender | or more |
| 20:04.44 | navaismo | ~book |
| 20:04.44 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:06.46 | joecool | jameswf: any idea how i can stop this from happening? do i need to switch port ranges? |
| 20:08.10 | jameswf | joecool: you want it to stop using udp ports? |
| 20:10.24 | joecool | jameswf: no, i described my issue up above, it seems like the regitration is getting confused between phones because i'm using the same server but different extensions from WAN to LAN, i was assuming that it's mapping to the same port and causing things to get confused |
| 20:10.44 | joecool | s/regitration/registration/ |
| 20:11.23 | joecool |
|
| 20:12.57 | *** join/#asterisk bsdice (~bsdice@meran.embinet.com) |
| 20:17.12 | jameswf | ~extip |
| 20:17.22 | jameswf | meh |
| 20:17.53 | jameswf | ~nat |
| 20:17.54 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 20:18.46 | bsdice | my phone... it still works |
| 20:25.39 | *** part/#asterisk JamesJRH (~james@ps87731.dreamhost.com) |
| 20:27.06 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 20:28.02 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 20:34.57 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 20:46.16 | *** join/#asterisk FuriousGeorge (182c2966@gateway/web/freenode/ip.24.44.41.102) |
| 20:47.36 | FuriousGeorge | hey all |
| 20:49.07 | FuriousGeorge | I've been away since 1.4, so it's been a while, and forgive me if this is a dumb question, but: is this normal output for sip show channels when no one is on the phone? |
| 20:49.07 | FuriousGeorge | http://pastebin.com/dqR93ksR |
| 20:51.23 | FuriousGeorge | i remember in the 1.2 days I would sometimes have issues on another server with sip <-> zap channels that would not hangup, and i'd have to go in there and destroy the channels. this doesn't look like that though. plus it's all sip end to end, so maybe this is normal? |
| 20:51.27 | navaismo | yes, SIP dialogs between server and endpoints |
| 20:52.01 | *** join/#asterisk mokmeister (~mokmeiste@86.41.97.128) |
| 20:52.52 | FuriousGeorge | navaismo: how does one check to see how many active calls are going on these days? |
| 20:53.21 | navaismo | core show channels verbose at the end it say how many channels and how many active calls |
| 20:54.54 | FuriousGeorge | navaismo.karma++ |
| 20:55.11 | FuriousGeorge | ty |
| 20:55.40 | *** join/#asterisk bulkorok (~Adium@gw1.pinguin.ag) |
| 20:56.02 | navaismo | bitcoins please :P |
| 21:00.56 | *** part/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:10.12 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 21:19.01 | drmessano | Bitcoins |
| 21:19.07 | drmessano | I have like 3 of them |
| 21:25.44 | FennecVox | So, is anyone familiar with the AA50? |
| 21:28.10 | FennecVox | happened into one |
| 21:31.58 | FennecVox | is about to contact Digium to ask about purchasing a firmware update for it, but is afraid of getting either laughed at or 4 figures demanded of him |
| 21:49.47 | drmessano | Doesn't seem like it's worth it |
| 21:49.59 | _Corey_ | FennecVox: I believe that it's EOL'd |
| 21:50.02 | drmessano | It's only going to be based on a much older codevase |
| 21:50.05 | drmessano | It's only going to be based on a much older codebase |
| 21:50.23 | drmessano | All you're doing is adding newer, insecure code to it |
| 21:50.43 | drmessano | (newer than what it's running) |
| 21:50.58 | drmessano | That's like upgrading from Windows 95 to 98 |
| 21:54.55 | FennecVox | drmessano: I'm concerned about that it seems to have a random reboot issue. I'm going to keep it inside my LAN and use it as an IAX FXO/FXS gateway, lol |
| 21:55.29 | drmessano | How often is it rebooting? |
| 21:56.17 | FennecVox | drmessano: mean is 26 hours, min is 13, max is 31 |
| 21:56.29 | FennecVox | according to my firewall's dhcp server log |
| 21:57.17 | FennecVox | I did some googling, and there are a lot of posts about the 1.3.0.5 firmware having reboot issues and 2.0.0.5 not having them |
| 21:57.34 | FennecVox | but I can't find 2.0.0.5's uImage anywhere, and I'd rather not pirate it if I can at all help it |
| 21:57.52 | drmessano | I would call them. They may even give it to you |
| 21:57.57 | drmessano | or tell you where to pirate it |
| 21:58.24 | FennecVox | their support line seemed to be closed when i tried to call earlier |
| 21:58.25 | drmessano | That thing is as extinct as an IAXy |
| 21:58.45 | FennecVox | I could drive over to their office and look |
| 21:58.49 | FennecVox | it's only like 3 minutes from here |
| 21:59.02 | FennecVox | knock on the door. "hey, anyone got firmware for this?!" |
| 21:59.09 | drmessano | I would |
| 21:59.35 | FennecVox | Tomorrow during business hours. x3 |
| 22:00.03 | FennecVox | but yeah, i mainly intend to use it just as a fxo/fxs box |
| 22:00.15 | FennecVox | because my audiocodes gateway has been being "funny" lately |
| 22:01.13 | FennecVox | and the AudioCodes box is more dead than this bad thing. |
| 22:01.58 | FennecVox | Audiocodes EOLed the MP-108 2 years before the AA50 was even made lol |
| 22:09.23 | *** join/#asterisk petris (~petris@192.184.93.147) |
| 22:26.42 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 22:27.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:28.09 | FennecVox | So, any major security problems I should be aware of with this AA50 |
| 22:28.29 | FennecVox | keeping in mind it's going to be behind my network firewall, and I don't intend to point it at anything Internet-facing |
| 22:28.57 | [TK]D-Fender | FennecVox: plenty of OS vulnerabilities, as wel Asterisk and every single thing on it potentially. |
| 22:29.03 | [TK]D-Fender | It is YEARS out of date |
| 22:29.21 | FennecVox | Yeah, that's why I have a rule in pfsense saying "deny all from 192.168.5.181" |
| 22:30.47 | FennecVox | I intend to use it as a SIP/IAX FXS/FXO box to a newer asterisk VM I already have around my LAN |
| 22:31.08 | FennecVox | Since my AudioCodes MP-108 is dying, lol |
| 22:31.42 | [TK]D-Fender | [18:29]FennecVoxYeah, that's why I have a rule in pfsense saying "deny all from 192.168.5.181" sounds like you be resdtricting it TO a single IP (your VM server) |
| 22:32.02 | FennecVox | The vm-server and the box are on the same subnet |
| 22:32.06 | FennecVox | it'd never cross pfsense |
| 22:32.19 | FennecVox | this is for my home "play" network |
| 22:32.44 | FennecVox | but having a decent FXO would be nice to have, since the current FXO I have is some kind of winmodem, lol |
| 22:33.20 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 22:33.25 | [TK]D-Fender | Well if you lock down the hosts that can get to it it could do the job. (FXS/FXO -> SIP) |
| 22:34.16 | FennecVox | nods |
| 22:34.37 | FennecVox | second question: is pbx in a flash a decent pre-assembled asterisk distribution, or should I look for others |
| 22:35.37 | [TK]D-Fender | what do you actually want to do? |
| 22:36.32 | FennecVox | I got a giant box of polycom phones. i have a single analog phone line coming into my house now. I was using PIAF with Google Voice, but I'm going to be able to use the AA50 as an FXO over SIP now. |
| 22:36.48 | FennecVox | I want to be able to plonk Polycom phones down around my house to confuse my roommates. |
| 22:37.35 | FennecVox | "Why do we have to dial 9 to call for pizza now?" |
| 22:38.00 | [TK]D-Fender | Oh, so 1980 is calling.. they want their dialing prefixes back... |
| 22:38.16 | FennecVox | THAT is entirely retaliation for them putting a coin box on the toilet. |
| 22:39.21 | [TK]D-Fender | Well if you don't actually care to learn or anything.. then just go with the FreePBX provided ISO |
| 22:52.59 | *** join/#asterisk ralphmazio (~ralphmazi@nc-76-5-180-51.dhcp.embarqhsd.net) |
