| 00:31.36 | ani216 | apb1963: thank you :) |
| 00:38.47 | *** join/#asterisk Micc (~Micc@static-50-125-113-34.frr01.both.wa.frontiernet.net) |
| 00:38.57 | Micc | anyone have any experience with Aastra 612d's? |
| 00:41.55 | igcewieling | Reminds me of the old telecom saying. When you solve a problem with DTMF by using inband, now you have two problems. |
| 00:44.53 | coppice | igcewieling: quoting Pyrrhus seems appropriate here |
| 01:03.42 | killown | what's the best asterisk distro? |
| 01:06.43 | apb1963 | killown: The one you roll yourself of course. |
| 01:07.44 | killown | apb1963, I mean something similar to elastix? |
| 01:07.47 | apb1963 | But I think the local constituency prefers the one by shmooze |
| 01:08.25 | apb1963 | for some reason the folks here get agitated when you mention the E word. |
| 01:09.49 | WIMPy | If you want something to click, you're in the wrong channel. |
| 01:10.42 | killown | I want something to manage |
| 01:11.11 | WIMPy | What does that mean? |
| 01:11.20 | apb1963 | He wants a supermarket |
| 01:11.36 | apb1963 | Manage THIS! |
| 01:11.44 | apb1963 | highfives himself |
| 01:11.47 | WIMPy | Not a football club? |
| 01:12.10 | apb1963 | Do you mean football, or soccer? |
| 01:12.25 | WIMPy | Depends on location. |
| 01:12.42 | WIMPy | But I guess the game principle would be the same anyway. |
| 01:12.46 | apb1963 | Whose location? |
| 01:13.58 | WIMPy | I mean football in the european meaning. So I guess that's soccer to you. |
| 01:14.26 | apb1963 | Yes it is |
| 01:14.33 | apb1963 | That's why I asked |
| 01:14.47 | apb1963 | One does not use their head in football |
| 01:15.02 | apb1963 | at least not to block |
| 01:15.04 | WIMPy | But for my comment it shouldn't matter. |
| 01:15.31 | apb1963 | Well you have to know the rules of the game if you're going to manage a team sport. |
| 01:15.51 | apb1963 | Or a sports team. |
| 01:15.56 | WIMPy | You are completely unprofessional. |
| 01:16.04 | apb1963 | am not |
| 01:16.20 | WIMPy | A ral manager doesn't care what he manages. |
| 01:16.30 | WIMPy | real |
| 01:16.34 | apb1963 | Well he may not care, but he has to know. |
| 01:17.11 | WIMPy | What do they always say: Knowledge might be beneficial but is not necessary. |
| 01:17.37 | apb1963 | I have no knowledge of that saying. |
| 01:18.11 | WIMPy | You never had to deal with managers? |
| 01:18.23 | apb1963 | No... they have to deal with me. :) |
| 01:18.54 | WIMPy | So you don't give them a chance to speak? |
| 01:19.29 | apb1963 | Doesn't mean I have to listen :) |
| 01:19.32 | WIMPy | Sounds sensible, but very exhausting. |
| 01:19.47 | WIMPy | Good point. |
| 01:20.00 | apb1963 | That's why I wear a hat |
| 01:20.30 | WIMPy | A noise cancelling hat? |
| 01:20.38 | apb1963 | a point covering hat |
| 01:21.09 | apb1963 | to cover my good pointy head |
| 01:21.37 | apb1963 | although a noise cancelling hat sounds highly beneficial |
| 01:22.24 | apb1963 | I think we lost the OP |
| 01:22.37 | WIMPy | Pointy haired boss? |
| 01:22.51 | apb1963 | Shhhh.... I'm incognito. |
| 01:23.16 | WIMPy | In the Wiener district? |
| 01:23.35 | apb1963 | I have a terrible sense of direction. |
| 01:24.19 | apb1963 | and apparently, I now have to watch out for Ninjas. |
| 01:24.42 | apb1963 | Luckily, I have Ninja repellent. It seems to be working. |
| 01:25.00 | apb1963 | Haven't seen a single Ninja all day. |
| 01:25.38 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 01:25.42 | saint_ | back |
| 01:25.50 | apb1963 | prays |
| 01:25.50 | saint_ | is there a way to have an extension with an option first digit ? |
| 01:26.02 | saint_ | like... exten => [1]3345 |
| 01:26.14 | saint_ | I want this exten to be dialed whether or not the 1 is in front of it |
| 01:26.27 | saint_ | i understand that [1] is no good, of course. |
| 01:26.36 | saint_ | but is there a solution to have an optional digit ? |
| 01:27.42 | WIMPy | Not possible. |
| 01:27.55 | WIMPy | You have to have to extensions. |
| 01:28.07 | apb1963 | how about _X3345 and 3345 ? |
| 01:28.13 | WIMPy | two |
| 01:28.17 | apb1963 | yes two |
| 01:28.42 | *** join/#asterisk timeshell (~chatzilla@69-165-165-76.dsl.teksavvy.com) |
| 01:29.36 | timeshell | After changing the bindport on my asterisk server to something different from default, when I try to make a call with a device connect to the new port to another extension I get the message Call from '7022' (192.168.2.105:54876) to extension '7012' rejected because extension not found in context 'DLPN_DialPlan1' |
| 01:29.58 | timeshell | Both these devices are registered with the server and worked before changing the port |
| 01:30.14 | timeshell | Version 1.8.18 |
| 01:30.17 | timeshell | Any thoughts? |
| 01:30.59 | WIMPy | doesn't see how the stories could be related. |
| 01:31.12 | apb1963 | my thought is pastebin your context and let WIMPy have a look at it. |
| 01:31.30 | timeshell | Stories? |
| 01:31.50 | carrar | heh |
| 01:31.52 | timeshell | Not sure even why context would matter. Worked before port change, doesn't work after port change |
| 01:32.00 | timeshell | ?? |
| 01:32.15 | apb1963 | The context will tell the story |
| 01:32.21 | timeshell | meh, just a sec |
| 01:33.12 | timeshell | http://www.pastebin.ca/2403641 |
| 01:33.27 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 01:34.28 | WIMPy | That doesn't tell us much. |
| 01:34.41 | timeshell | No kidding |
| 01:34.54 | timeshell | Configured mostly using asterisk-gi |
| 01:34.56 | timeshell | gui |
| 01:35.00 | apb1963 | if I had a context like that... I suppose I'd grep for 7012 in all the conf files to see where it was. |
| 01:35.08 | timeshell | Changed the bindport manually in sip.conf |
| 01:35.35 | WIMPy | dialplan show 7012@DLPN_DialPlan1 |
| 01:35.35 | timeshell | Look, simple question. Works using 5060, doesn't work using something else. |
| 01:35.38 | carrar | Ah I see the issue |
| 01:35.42 | apb1963 | that too |
| 01:35.44 | carrar | Are you using Asterisk from source? |
| 01:35.48 | timeshell | Yes |
| 01:35.51 | timeshell | 1.8.18 |
| 01:36.08 | carrar | heh |
| 01:36.22 | apb1963 | well no wonder... you have it configured for "evil". You need to switch it to "good". |
| 01:36.33 | timeshell | sighs |
| 01:36.39 | carrar | Whats with this gui talk? |
| 01:36.56 | timeshell | So dummies can configure it when I'm not around |
| 01:37.11 | carrar | Aren't you the one having the issue? |
| 01:37.13 | timeshell | What's that have to do with the bind port? |
| 01:37.22 | timeshell | hah hah |
| 01:38.36 | apb1963 | that's exactly right... what DOES it have to do with the bind port? Hence the request to see the dialplan because that's what matters... AFAIK |
| 01:38.40 | WIMPy | >>dialplan show 7012@DLPN_DialPlan1 |
| 01:39.10 | carrar | did you read {source}/configs/sip.conf.sample to see where you bind your SIP Ports? |
| 01:39.11 | timeshell | There is no existence of 'DLPN_DialPlan1' context |
| 01:39.42 | WIMPy | What? You showed us that context. |
| 01:39.44 | timeshell | Um? I already said devices registered on new port |
| 01:39.47 | timeshell | EXACTLY |
| 01:39.51 | apb1963 | well then that's the problem I would think |
| 01:40.12 | apb1963 | is confuzzled |
| 01:40.31 | timeshell | The context exists, but appears to be invisible to devices registered on non-default port |
| 01:40.45 | carrar | invisible=no |
| 01:40.57 | apb1963 | [DLPN_DialPlan1] |
| 01:41.01 | WIMPy | That makes no sense. |
| 01:41.11 | carrar | ~book |
| 01:41.11 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:41.12 | timeshell | That's why I'm here asking questions |
| 01:41.16 | WIMPy | Devices don't see anyhing. Your Asterisk does. |
| 01:41.23 | WIMPy | Or you broke it. |
| 01:41.24 | timeshell | Whatever, you know what I mean |
| 01:42.22 | timeshell | In sip.conf I set bindaddr=0.0.0.0:mynewport |
| 01:42.36 | timeshell | And tcpbindaddr=0.0.0.0:mynewport |
| 01:42.47 | timeshell | Not rocket science |
| 01:42.59 | timeshell | And then this happened. |
| 01:43.24 | timeshell | I could put it back to 5060, but I want to make it less visible to the outside world. |
| 01:43.42 | carrar | Do you're phones know to use that none standard port? |
| 01:43.51 | WIMPy | Maybe that makes your device not supply the correct credentials any more and end up in another context? |
| 01:43.57 | timeshell | Well if they didn't they could register with the server could they |
| 01:44.19 | carrar | one would hope |
| 01:44.20 | timeshell | sorry COULDN"T |
| 01:44.43 | timeshell | lWell if they didn't they couldn't register with the server could they |
| 01:44.50 | carrar | You tell me |
| 01:44.59 | WIMPy | register!=call |
| 01:45.06 | apb1963 | puts on his [TK]D-Fender hat & t-shirt |
| 01:45.12 | apb1963 | Prove that they register |
| 01:45.19 | carrar | Looking at your detailed sip trace I don't see that |
| 01:45.21 | timeshell | They couldn't register with the server until I changed the port |
| 01:45.27 | timeshell | On the sip phone that is |
| 01:45.45 | apb1963 | He had a detailed sip trace? I didn't see it. |
| 01:45.48 | carrar | no |
| 01:45.58 | timeshell | 7022/7022 192.168.2.105 D N 54876 Unmonitored |
| 01:46.04 | WIMPy | Or show us the verbose output of a working call and a non working call. |
| 01:46.09 | timeshell | 7012/7012 192.168.2.126 D N 50911 Unmonitored |
| 01:46.44 | WIMPy | You should read the security best practices. |
| 01:46.45 | timeshell | TCP peers |
| 01:46.58 | apb1963 | TCP? |
| 01:47.15 | timeshell | Yah. |
| 01:47.20 | timeshell | Bria for iPhone |
| 01:47.27 | timeshell | Bria for iPad |
| 01:47.39 | apb1963 | Bria with crackers? |
| 01:47.40 | timeshell | TCP is better, keeps battery usage low |
| 01:47.46 | carrar | heh |
| 01:47.47 | apb1963 | say what? |
| 01:48.02 | timeshell | iOS manages TCP in background, doesn't manage UDP in background |
| 01:48.10 | apb1963 | chucks everything he knows out the window |
| 01:48.13 | timeshell | Better battery management when using TCP |
| 01:48.30 | apb1963 | another management guy |
| 01:48.45 | timeshell | Well, I don't want my iphone battery dead in 6 hours |
| 01:49.01 | timeshell | Because I have to keep UDP app active in the background |
| 01:49.09 | timeshell | TCP more stable anyway |
| 01:49.43 | timeshell | Anyhow, this is off topic. |
| 01:49.59 | timeshell | Why does changing the default port kill asterisk? |
| 01:50.16 | apb1963 | kill is such a harsh word |
| 01:50.18 | WIMPy | You tell us. |
| 01:50.23 | apb1963 | What does your firewall say? |
| 01:50.44 | WIMPy | Or actually, you don't tell us. |
| 01:50.45 | carrar | Asterisk worked fine for me on none standard ports |
| 01:50.55 | apb1963 | does it allow TCP packets on that new port? |
| 01:51.07 | carrar | I'm not crazy enough to use TCP |
| 01:51.08 | apb1963 | grasps at straws |
| 01:51.20 | apb1963 | i know... right? |
| 01:51.23 | timeshell | Would asterisk say that context didn't exist and see the call attempt if it didn't? |
| 01:51.37 | timeshell | <PROTECTED> |
| 01:51.40 | apb1963 | we're in uncharted waters here |
| 01:51.58 | timeshell | yah, I always seem to be able to find where things are broken |
| 01:52.04 | carrar | timeshell: guessing you have something missconfigured |
| 01:52.17 | carrar | based on the details provided |
| 01:52.22 | *** join/#asterisk n8ideas (~joshelson@216.241.45.82) |
| 01:52.49 | WIMPy | Unless you start to give us somethign to work on, I choose to believe that it was not the only change made. |
| 01:52.51 | timeshell | Yah, cause contexts always become invisible when you change the default port |
| 01:53.05 | n8ideas | has anyone seen function CHANNEL not work for setting faxdetect or t38support, even though documentation would seem to indicate that is possible? |
| 01:53.08 | apb1963 | we're guessing timeshell... check these things whether they make sense or not. Improper configuration can have undefined results. |
| 01:53.39 | n8ideas | oh.... nevermind. I may have read documentation wrong |
| 01:56.53 | apb1963 | hmmm...I ordered a seal an hour ago.. wonder what happened to it. |
| 01:57.14 | carrar | escaped back to the ocean |
| 01:57.41 | apb1963 | dammit... what am I gonna eat? |
| 02:08.29 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.157) |
| 02:20.58 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 02:25.24 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 02:25.39 | saint_ | What is the best way to have a 0 to call all extension, but the caller's extension ? |
| 02:26.03 | saint_ | Right now, my 0 calls everybody. If I dial it from extension 300 , I see the call coming on the 2nd line of 300, which I do not want .. |
| 02:32.12 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 02:57.14 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 03:13.50 | *** join/#asterisk tehrabbitt (~tehrabbit@unaffiliated/tehrabbitt) |
| 03:13.58 | tehrabbitt | join #reddit-sysadmin |
| 03:14.01 | tehrabbitt | erm |
| 03:14.24 | tehrabbitt | anyway lol trying to set up asterisk for the first time in a few years now |
| 03:14.46 | tehrabbitt | I did it a few years back, I'm just rusty :\ I forget how to accept inbound calls for multiple DIDs |
| 03:15.08 | tehrabbitt | can anyone point me to a good sample / resource I could use to brush up? |
| 03:15.17 | tehrabbitt | (google has been returning nothing but complicated dialplans) |
| 03:15.23 | WIMPy | ~book |
| 03:15.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:16.04 | tehrabbitt | WIMPy: Yes, I know the book... I mean like if there was a sample dialplan anyone knew of that wasn't completely crazy lol |
| 03:16.15 | tehrabbitt | even the samples that come with asterisk are so long it's crazy |
| 03:16.37 | WIMPy | YOU write YOUR dialplan. |
| 03:17.03 | WIMPy | As far as your question goes there is nothing that could be recycled from any example. |
| 03:17.45 | tehrabbitt | WIMPy: I understand this.... maybe i'm saying it wrong... I understand what needs to go into the dialplan, I just forget how to listen to a single SIP registration and then depending on the DID coming in, send it to a different internal SIP extension |
| 03:18.18 | WIMPy | By definig multiple extensions. |
| 03:18.19 | tehrabbitt | if I remember, it was just exten => 15555551234,n,Answer() for instance |
| 03:18.37 | WIMPy | Unless your ITSP doesn't tell you what number was called. |
| 03:18.39 | tehrabbitt | and then the second line would be the Dial() for dialing hte internal line |
| 03:19.06 | WIMPy | You shouldn't Answer() before a Dial(). |
| 03:19.08 | tehrabbitt | WIMPy: it's flowroute, i'm pretty sure they do because I did have t working in the past... I just haven't touched asterisk in close to 2 years now sadly :\ |
| 03:19.16 | tehrabbitt | WIMPy: hm |
| 03:19.33 | tehrabbitt | so it would be just a straight dial? |
| 03:19.56 | WIMPy | If you just want to pass the call to one or more devices, yes. |
| 03:20.07 | tehrabbitt | ah |
| 03:20.18 | tehrabbitt | only time for Answer() is when using the IVR right? |
| 03:20.55 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 03:20.59 | WIMPy | Not even then. Many Applications such as Playback will do it implicitly. |
| 03:21.09 | tehrabbitt | WIMPy: so what's the point of Answer() |
| 03:21.32 | WIMPy | But not all do, obe example would be VoiceMail which doesn't. |
| 03:21.44 | WIMPy | s/obe/one/ |
| 03:21.46 | tehrabbitt | Ah |
| 03:22.37 | WIMPy | But you don't want to answer a call going to a phone before the callee answers. |
| 03:22.42 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 03:27.47 | tehrabbitt | WIMPy: that makes sense |
| 03:36.42 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 03:47.21 | saint_ | tehrabbitt: you use flowroute ? I just switched to them. |
| 03:50.12 | saint_ | going to bed |
| 03:50.12 | saint_ | c ya |
| 03:51.17 | tehrabbitt | any reason why sip reload doesn't work? |
| 03:51.18 | tehrabbitt | :\ |
| 03:51.42 | tehrabbitt | as in SIP cannot be found |
| 03:52.22 | WIMPy | sip.conf was too wrong so the module didn't load? |
| 03:52.37 | tehrabbitt | WIMPy: it says there is no such module. |
| 03:52.41 | tehrabbitt | how can I check? |
| 03:52.52 | WIMPy | module load chan_sip |
| 03:52.53 | tehrabbitt | and I don't see how it's too wrong... |
| 03:53.15 | tehrabbitt | [Jun 20 23:52:44] WARNING[13953]: loader.c:423 load_dynamic_module: Error loading module 'chan_sip': /usr/lib/asterisk/modules/chan_sip.so: cannot open shared object file: No suc |
| 03:53.29 | tehrabbitt | so yeah, apparently the module doesn't exist and I have no idea why |
| 03:53.44 | WIMPy | You built it without sip support? |
| 03:54.14 | tehrabbitt | WIMPy: no idea how... I built it the way i've always done it... I didn't unselect SIP from make menuconfig |
| 03:54.22 | tehrabbitt | in fact, SIP you can't unselect |
| 03:54.31 | tehrabbitt | unless... maybe I missed a step |
| 03:54.40 | tehrabbitt | make modules? |
| 03:54.47 | WIMPy | I'm sure you can deselect it. |
| 03:54.55 | tehrabbitt | it shows up as XXX |
| 03:55.06 | WIMPy | Maybe you deselected websocket support. |
| 03:55.18 | WIMPy | It should tell you what's missing at the bottom. |
| 03:55.34 | tehrabbitt | WIMPy: I didn't change anything but add support for the music on hold packages. |
| 03:56.00 | tehrabbitt | i've uninstalled and reinstalled 3 times now |
| 03:56.04 | WIMPy | No information why you can't select it? |
| 03:56.38 | tehrabbitt | <PROTECTED> |
| 03:56.55 | tehrabbitt | XXX chan_sip |
| 03:57.18 | WIMPy | So do you have those enabled? |
| 03:57.23 | tehrabbitt | first time around, I didn't even do make menuconfig |
| 03:57.33 | tehrabbitt | chan_local, yes. |
| 03:57.40 | tehrabbitt | gotta check the other 1 |
| 03:57.42 | tehrabbitt | erm 2 |
| 03:57.56 | tehrabbitt | yes they are |
| 03:58.16 | WIMPy | Interesting. |
| 03:58.38 | tehrabbitt | this is 11.4.0 btw |
| 03:58.48 | WIMPy | I guess, you should re-run configure and pay close attention to its output. |
| 03:58.48 | tehrabbitt | Linux voip 3.2.0-4-amd64 #1 SMP Debian 3.2.46-1 x86_64 GNU/Linux |
| 03:59.04 | tehrabbitt | i'm thinking you're right |
| 03:59.25 | tehrabbitt | doing a make uninstall-all and then a make clean |
| 04:00.24 | tehrabbitt | package is configured it says |
| 04:00.34 | tehrabbitt | WIMPy: should I just do a "make' or should I do the make menuconfig |
| 04:01.05 | WIMPy | If you're comfortable with the defaults... |
| 04:01.24 | tehrabbitt | WIMPy: defaults *should* work, no? |
| 04:01.28 | WIMPy | But running menuconfig should tell you faster if it will build chan_sip. |
| 04:01.37 | WIMPy | yes |
| 04:02.16 | tehrabbitt | i'm watching the make... it didn't make chan_sip.so |
| 04:02.18 | tehrabbitt | :\ |
| 04:02.24 | tehrabbitt | bad package maybe? |
| 04:02.49 | WIMPy | More likely a missing dependency. |
| 04:03.04 | tehrabbitt | WIMPy: how would I find that out? |
| 04:03.17 | tehrabbitt | this is debian jessie btw |
| 04:03.20 | WIMPy | config.log |
| 04:03.40 | tehrabbitt | hm okay |
| 04:03.47 | WIMPy | Jessie? Not heard before. |
| 04:03.49 | tehrabbitt | just look for anything "SIP" related? |
| 04:04.08 | tehrabbitt | WIMPy: it's the next "Testing" debian... goes hand in hand with basically the stable version of Ubuntu |
| 04:04.15 | WIMPy | But I know I got pissed before because everything you need is missing on Debian by default. |
| 04:04.23 | tehrabbitt | lol |
| 04:04.28 | tehrabbitt | yeah, i've dealt with that before |
| 04:04.49 | tehrabbitt | nothing in there about SIP /sip |
| 04:05.18 | WIMPy | Any "not found"s or "no"s? |
| 04:05.53 | tehrabbitt | /usr/bin/ld: cannot find -lvorbis |
| 04:05.56 | tehrabbitt | etc |
| 04:06.16 | WIMPy | That one isn't important. |
| 04:08.14 | tehrabbitt | :| |
| 04:08.16 | tehrabbitt | openssl.... |
| 04:08.21 | tehrabbitt | seriously? lol |
| 04:08.27 | tehrabbitt | res_crypto requires it. |
| 04:08.39 | *** join/#asterisk LokiScarlet (~loki@crantrap.cranberrytrap.org) |
| 04:08.40 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-tjakkeczqmwnrahr) |
| 04:08.41 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 04:08.53 | tehrabbitt | but it's isnstalled.... |
| 04:08.54 | tehrabbitt | weird |
| 04:09.14 | WIMPy | Also the famous -dev? |
| 04:09.25 | tehrabbitt | hm good point lmao |
| 04:09.36 | *** join/#asterisk jkister (~chatzilla@pool-108-36-87-4.phlapa.fios.verizon.net) |
| 04:10.02 | tehrabbitt | libssl-dev |
| 04:10.48 | tehrabbitt | installed it, doing a ./configure again |
| 04:11.09 | tehrabbitt | if suddenly SIP shows up... i'm going to laugh my ass off :| always those stupid -dev packages :\ |
| 04:11.37 | tehrabbitt | 0_o THat was it. |
| 04:11.40 | tehrabbitt | <PROTECTED> |
| 04:12.02 | WIMPy | concludes: Verson doesn't matter. If you see Debian, run for your life. |
| 04:12.09 | tehrabbitt | hahahahaha |
| 04:13.01 | tehrabbitt | making again... hopefully this time, SIP is installed haha |
| 04:13.49 | tehrabbitt | WIMPy: btw, I found this... see how he has a "simple" dialplan? with answer()? you're saying that's bad practice? |
| 04:13.52 | tehrabbitt | http://letitknow.wordpress.com/2011/05/16/how-to-install-asterisk-1-8-4-on-debian-6-0-1/ |
| 04:14.30 | WIMPy | Yes. |
| 04:14.40 | WIMPy | And so is using extensions as usernames. |
| 04:15.06 | WIMPy | Check the security best practices. |
| 04:18.45 | tehrabbitt | WIMPy: where would said best practices be? (book?) |
| 04:18.53 | tehrabbitt | (for security) |
| 04:19.37 | WIMPy | In your source dir. |
| 04:19.44 | tehrabbitt | ah |
| 04:19.48 | WIMPy | readmeseriousely or something. |
| 04:20.00 | tehrabbitt | lol it's actually called something like that? :o |
| 04:27.13 | *** join/#asterisk petris (znc@ip-50-62-86-130.ip.secureserver.net) |
| 04:28.16 | tehrabbitt | WIMPy: so basically, rather than calling a deviece by the extension #, give it an actual name, or use the MAC address, so someone who knows the extension, can't easially brute-force the secret and add their *own* client heh |
| 04:28.46 | tehrabbitt | (because if you know the device username is 1000 for extension 1000, you're halfway there basically) |
| 04:28.57 | WIMPy | Well, MACs are witin certain ranges. |
| 04:29.08 | WIMPy | yeah |
| 04:29.24 | tehrabbitt | WIMPy: so basically it just makes it that much harder to find the credentials |
| 04:29.35 | WIMPy | yes |
| 04:31.51 | tehrabbitt | WIMPy: hm, it won't let me do a "dialplan reload" from the cli |
| 04:32.02 | tehrabbitt | SIP loaded though :) |
| 04:32.21 | *** join/#asterisk gigo1980 (~Adium@178-24-166-237-dynip.superkabel.de) |
| 04:32.32 | WIMPy | What happened? |
| 04:32.45 | tehrabbitt | WIMPy: no config files |
| 04:32.51 | tehrabbitt | i'm guessing that's part of it |
| 04:32.52 | tehrabbitt | lol |
| 04:32.59 | tehrabbitt | should I just do "make samples" |
| 04:33.01 | gigo1980 | hi all. i have to build an IVR. which "packages" from APT does i have to install on Ubuntu 12.04LTS? |
| 04:33.01 | tehrabbitt | or not worth it |
| 04:33.52 | WIMPy | The samples are not exately meant to be used. Mor like linkine documentation. |
| 04:34.13 | tehrabbitt | WIMPy: gotcha |
| 04:34.22 | tehrabbitt | so I have to make an asterisk.conf from scratch, etc. |
| 04:34.42 | WIMPy | That's one of the files you can just copy over. |
| 04:34.46 | gigo1980 | or is there an manual / howto to setup how to setup an IVR (only) system on ubuntu ? |
| 04:34.57 | tehrabbitt | WIMPy: from where? |
| 04:35.03 | WIMPy | And make sure you have a modules.conf with autoload=yes. |
| 04:35.23 | WIMPy | From the source. |
| 04:35.25 | tehrabbitt | ah |
| 04:35.26 | tehrabbitt | found it |
| 04:35.40 | WIMPy | What did you think where the samples would come from? |
| 04:36.02 | tehrabbitt | lol make samples |
| 04:36.13 | tehrabbitt | figured it downloaded them or such |
| 04:36.32 | WIMPy | gigo1980: Shouldn't matter if you just want IVR or more. |
| 04:37.10 | tehrabbitt | WIMPy: i'm guessing i'm better off making my own voicemail.conf as well right? |
| 04:37.20 | WIMPy | gigo1980: If you build from source you can choose not to build the things you don't need. |
| 04:38.04 | WIMPy | You probably don't need too much in there. |
| 04:38.32 | tehrabbitt | WIMPy: so voicemail.conf should be able to be brought over too? |
| 04:38.47 | WIMPy | Not having config file for stuff you don't need will often prevent those parts from being loaded. |
| 04:39.13 | WIMPy | You CAN use them all, if you want and dare. |
| 04:39.35 | tehrabbitt | awesome, now asterisk isnt' starting up :\ |
| 04:39.36 | tehrabbitt | lol |
| 04:39.39 | tehrabbitt | i think i know why |
| 04:39.44 | WIMPy | But make sure you understand the dialplan and anythin network related if the box has an internet connection. |
| 04:40.24 | tehrabbitt | WIMPy: yeah, I made that mistake 2 years ago by using simple passwords etc... cost me $500 in calls to overseas painful lesson to learn |
| 04:40.27 | tehrabbitt | lol |
| 04:41.04 | WIMPy | The samples are meant to give you an easy feeling of success. That also means they probably allow much more than you want. |
| 04:41.31 | tehrabbitt | WIMPy: yeah lol |
| 04:42.07 | tehrabbitt | WIMPy: all I can say is that's why I was trying to figure out what the "best" way of setting up the dial rules were, because I was using old notes from 2 years ago that most likely had mistakes Heh |
| 04:44.56 | gigo1980 | and is there an webpannel for managing the IVR ? |
| 04:45.18 | WIMPy | gigo1980: Not included. |
| 04:46.04 | gigo1980 | and excluded ? |
| 04:46.35 | WIMPy | Third party stuff not supported here. |
| 04:46.36 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 04:47.06 | gigo1980 | i dont want support. only an tip what i can use ;) |
| 04:47.38 | tehrabbitt | WIMPy: any reason why this is returning "404" exten => _800NXXXXXX,1,Dial(SIP/${EXTEN}@tollfreeOut) |
| 04:48.41 | WIMPy | Non-existant extensions return 404. So in that way the question doesn't make sense. |
| 04:49.16 | tehrabbitt | ah i figured it out I think |
| 04:49.21 | tehrabbitt | forgot to include the context |
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| 05:27.23 | Kobaz | [2013-06-21 01:21:10] WARNING[7401][C-00000002]: chan_sip.c:10171 process_sdp: Ignoring video stream offer because port number is zero |
| 05:27.33 | Kobaz | trying to make a video call and getting no video |
| 05:27.39 | Kobaz | both clients are lan, no nats |
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| 05:51.01 | tehrabbitt | any reason why core set verbose 10 isn't showing anything in the console? |
| 05:51.35 | WIMPy | logger.conf |
| 05:51.41 | tehrabbitt | ah |
| 05:53.13 | tehrabbitt | still nothing :\ |
| 05:54.13 | WIMPy | logger reaload |
| 06:00.03 | tehrabbitt | [Jun 21 01:59:35] WARNING[49709][C-00000007]: chan_sip.c:7280 sip_write: Asked to transmit frame type ulaw, while native formats is (g729) read/write = ulaw/ulaw |
| 06:00.09 | tehrabbitt | just flooded the terminal with that |
| 06:00.26 | tehrabbitt | what does that mean? |
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| 06:01.03 | WIMPy | You tried to use G.729? You need a licence for that. |
| 06:01.30 | tehrabbitt | I didn't though 0_o as far as I know |
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| 06:02.56 | tehrabbitt | figured it out... apparently there was a 729 there by accident |
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| 08:10.12 | MicW | hi |
| 08:10.52 | MicW | we installed a new asterisk a few days ago. everything works except my homeoffice phone. i cannot hear the other end from my homeoffice phone. what we have set up: |
| 08:11.37 | MicW | office: asterisk box on local network. port forwarding for external port 5060 to asterisk box |
| 08:12.07 | MicW | homeoffice: sip phone in local network. port forwarding for my external port 5160 to my sip phone (it listens on 5160) |
| 08:12.25 | MicW | i can register with my sip phone to the office (using office's external address) |
| 08:12.48 | MicW | with "sip show peers" i can see my homeoffice sip phone registered with my homeoffice external address, port 5160 |
| 08:12.55 | MicW | i can do calls and receive calls |
| 08:13.05 | MicW | people in office can hear me. i cannot hear them |
| 08:13.11 | *** join/#asterisk seik0 (5f34879e@gateway/web/freenode/ip.95.52.135.158) |
| 08:13.13 | MicW | what am i missing? |
| 08:13.45 | seik0 | hi again. Question: how is it possible to get on which app/appdata channel is now via AMI ? |
| 08:14.49 | seik0 | if it's hard over pure AMI, maybe some CLI command (AMI allows to execute them, though) |
| 08:14.55 | seik0 | ? |
| 08:17.30 | seik0 | anybody? |
| 08:20.02 | nix8n82 | MicW proper NAT settings. May have to use a stun server. Could be a lot of things |
| 08:20.29 | MicW | my office sip phone uses an external stun server (sipgate which is my phone provider) |
| 08:20.44 | MicW | there is no NAT at all invoked for sip |
| 08:21.08 | MicW | are there other ports to forward? |
| 08:23.09 | nix8n82 | RTP |
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| 08:26.51 | MicW | nix8n82, on my homeoffice are some ports forwarded for rtp. on the office not |
| 08:27.06 | MicW | which ports are used here and how can they be configured? |
| 08:29.06 | MicW | ah, found it |
| 08:29.13 | MicW | 10.000-20.000 |
| 08:29.21 | MicW | ist that txp or udp? |
| 08:29.23 | MicW | tcp |
| 08:29.46 | nix8n82 | UDP |
| 08:31.59 | MicW | ok, i changed this as well. i tried with/without stun server |
| 08:32.05 | MicW | its always the same |
| 08:32.46 | creativx | so how was this pattern matching setup again.. i have a range pattern defined, "_111174[5-9]X" - can i put "exten=>1111745599,1" _before_ this pattern and have it match? |
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| 08:33.31 | nix8n82 | On the homeoffice you have both rtp and sip forwarded to your phone?? |
| 08:34.27 | MicW | yes |
| 08:34.44 | MicW | this i did ~1 year before and is always worked |
| 08:35.03 | nix8n82 | What version of asterisk? |
| 08:37.07 | MicW | 11 |
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| 08:38.49 | nix8n82 | Nat is set to no? |
| 08:39.09 | MicW | it is |
| 08:39.11 | MicW | "no" |
| 08:39.31 | MicW | i checked the "full" log and found no error |
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| 08:45.54 | nix8n82 | Turn on sip debugging and you will be able to tell what ports are being used for the media streams |
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| 08:52.40 | MicW | how? with asterisk client? |
| 08:53.37 | MicW | i'll do it tonight. atm there are too many calls in the office |
| 08:53.44 | MicW | and i need to go to there now ^^ |
| 08:53.50 | nix8n82 | Yeah in the asterisk cli |
| 08:54.09 | MicW | thank you for your help. i'll need to go in deeper debugging |
| 08:54.23 | MicW | btw. i can hear the ringtone. so there is at least minimal audio |
| 08:54.41 | nix8n82 | No that is signalling |
| 08:55.15 | nix8n82 | Unless you are hearing music on hold |
| 08:58.44 | MicW | i see |
| 09:00.02 | MicW | bye |
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| 09:09.18 | fling | What is the name of the feature when I dial some numbers while talking with someone to reroute the call to somebody else? |
| 09:10.03 | fling | redirect? |
| 09:10.26 | fling | It is transfer, right? |
| 09:11.41 | sekil | transfer blind or attended |
| 09:11.55 | fling | sekil: what config file do I need? |
| 09:12.06 | sekil | it's usually done by a phone...but pbx can do it too |
| 09:12.44 | fling | Where are the default numbers for actions like transfer? |
| 09:13.03 | sekil | check features.conf |
| 09:13.08 | fling | sekil: thanks |
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| 12:45.22 | carrar | moo |
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| 13:42.30 | johnwigley | Hi, can I ask a question about the expected behaviour of MixMonitor with AudioHook_Inherit=Yes. I've tried asking this on the forum with no response. I have Caller A, call into a Queue and connect to Member B with MixMonitor & AudioHook_Inherit set on Member B's local channel dialplan or member connect Macro and I record the call as expected. If Member B then performs a SIP attended transfer to Member C, the call ceases to b |
| 13:43.15 | johnwigley | AudioHook_Inherit documentation, I would expect the call to be continued to be recorded? Is this expected behaviour, or do I need to do something different to continue recording the call through a SIP attended transfer? |
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| 13:49.08 | Kobaz | soooo |
| 13:49.17 | Kobaz | trying to make a video call and getting no video |
| 13:49.20 | Kobaz | [2013-06-21 01:21:10] WARNING[7401][C-00000002]: chan_sip.c:10171 process_sdp: Ignoring video stream offer because port number is zero |
| 13:49.29 | leifmadsen | Kobaz: oh I've seen that.. :) |
| 13:49.34 | Kobaz | both clients are lan, no nats |
| 13:49.36 | leifmadsen | something is invalid in the sdp or something |
| 13:49.54 | Kobaz | it's bria ipod talking to bria mac |
| 13:49.56 | leifmadsen | I forget why.... I feel like I asked file about this a few weeks back |
| 13:50.11 | Kobaz | i've seen posts on the web with people with the same problem all over |
| 13:50.25 | Kobaz | and one guy says in one reply 'oh i fixed it' or something similar |
| 13:50.29 | Kobaz | and never posted what he did to fix ot |
| 13:50.51 | file | port number zero = media stream is rejected/not being used |
| 13:51.20 | Kobaz | i've disabled all video except for h264 on both sides |
| 13:51.26 | Kobaz | and set allow=ulaw,h264 |
| 13:52.32 | file | if it's an incoming SDP... then it's the client |
| 13:52.59 | file | configuring codecs and all that won't matter if the client offers a video stream with no port |
| 13:54.29 | Kobaz | soooo |
| 13:54.33 | Kobaz | you fix this by doing.... |
| 13:54.42 | file | that's a client question |
| 13:54.51 | file | I did not write the client. |
| 13:55.06 | Kobaz | oh okay |
| 13:55.16 | Kobaz | wow that sounds bad that it's a messed up client |
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| 14:01.17 | carrar | file when are you writting a client ;) |
| 14:01.41 | carrar | call it... fileview |
| 14:01.51 | file | that'd be amusing |
| 14:02.20 | carrar | might clash with something the giv has already written |
| 14:02.24 | carrar | gov |
| 14:04.02 | Kobaz | so weird |
| 14:04.07 | Kobaz | so about this port number zero crap |
| 14:04.20 | Kobaz | someone wrote: However finally I made it. nat=yes |
| 14:04.40 | file | that wouldn't do it, likely a different issue |
| 14:05.03 | Kobaz | yeah especially since both phones are lan |
| 14:05.26 | Kobaz | so maybe there's a checkbox in the bria client "dont send port as zero" ? |
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| 14:07.10 | Kobaz | i think sdp needs a better error message |
| 14:09.26 | carrar | like what |
| 14:10.06 | Kobaz | like saying 'port is zero, we are ignoring the codec, fix your client' |
| 14:10.19 | file | it's not ignoring the codec |
| 14:10.23 | file | it's ignoring the video stream, as it says |
| 14:10.24 | Kobaz | rather than me needing to dig for two days to figure out what that means |
| 14:10.29 | Kobaz | video codec |
| 14:10.50 | Kobaz | it's not accepting video |
| 14:11.05 | file | it's not accepting the video stream because the port number is zero, exactly as it says |
| 14:11.22 | file | codecs aren't even examined |
| 14:12.52 | Kobaz | so how can it offer a stream without a codec |
| 14:13.00 | jmetro | think of network layers |
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| 14:13.04 | jmetro | data first, then protocol for the data. |
| 14:14.13 | Kobaz | well in network layers you have agreed upon communication methods end to end prior to sending |
| 14:14.22 | file | I didn't say there is no codec... I said that the client offered a video stream with no port, since that video stream isn't being used the code doesn't examine the codecs because it's useless to |
| 14:14.39 | Kobaz | ah |
| 14:14.49 | jmetro | ^ better explanation |
| 14:15.18 | Kobaz | so maybe the client is configured to not allow video or something |
| 14:15.26 | file | to quote the RFC: |
| 14:15.26 | file | A port number of zero in the offer indicates that the |
| 14:15.27 | file | <PROTECTED> |
| 14:15.30 | Kobaz | yeah |
| 14:15.32 | Kobaz | reading that now |
| 14:15.43 | carrar | RFC3264 5.1 2nd paragraph |
| 14:15.49 | Kobaz | yeap |
| 14:16.12 | Kobaz | maybe the warning should spit out the rfc and section |
| 14:16.23 | carrar | every warning should :) |
| 14:16.28 | Kobaz | true |
| 14:16.30 | file | oh god |
| 14:16.33 | carrar | hahah |
| 14:17.08 | Kobaz | Port set to 0 is perfectly legal and part of SDP offer/answer model. In fact, it probably means that there was something wrong with your SDP offer. For example, if you support PCMA codec and the peer only support PCMU, he will reject the SDP offer with the port set to 0. There can be quite some reasons with the offer was rejected but codec incompatibility is probably the most common. |
| 14:17.46 | file | actually that'd be rejected with a 488 in SIP |
| 14:17.55 | Kobaz | oh |
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| 15:32.15 | ipalmer | Hi all, I'm using * 1.8 and have a stuck channel which is causing my hints to show the endpoint as buys so it can't take any calls. I've tried using channel request hangup but the channel is still stuck afterwards. Any ideas how this can be reolved with restarting Asterisk |
| 15:34.40 | [TK]D-Fender | ipalmer: AMI Redirect -> off a cliff |
| 15:35.21 | *** join/#asterisk navaismo (~navaismo@189.241.95.6) |
| 15:35.40 | ipalmer | [TK]D-Fender: thanks for your response, I'm not aware of AMI Redirect? |
| 15:35.55 | [TK]D-Fender | Is that .... a question? |
| 15:36.03 | ipalmer | Yeah sorry |
| 15:36.15 | ipalmer | just googling AMI Redirect now |
| 15:36.32 | [TK]D-Fender | Use the AMI Redirect command .... to toss the call off a cliff |
| 15:36.47 | ipalmer | can I do this from the cli? |
| 15:37.28 | [TK]D-Fender | TELNET <- |
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| 15:39.17 | ipalmer | Ok sounds good, what would the command be to do this if for instance the stuck channel was SIP/1234 |
| 15:39.39 | Greenlight | Hangup the channel and make it go away, presumably |
| 15:40.21 | Greenlight | Although you might want to investigate the underlying cause of the channel getting stuck, and prevent it in the first place |
| 15:40.56 | igcewieling | Greenlight: Why!? Just cover up smallpox with some makeup and move on! |
| 15:41.17 | Greenlight | Ahh yes, of course, what what I thinking?! |
| 15:41.30 | Greenlight | adds "investigate" to his list of banned words |
| 15:42.12 | ipalmer | You are right Greenlight but as this is the only time it's happened in a year and a half, think I'll jsut cover it with some makeup for now :) |
| 15:42.43 | Greenlight | Ahh, in that case, makeup away :) |
| 15:43.03 | igcewieling | if it happens so seldom wouldn't it be easier to use the CLI hangup commands? |
| 15:43.10 | Greenlight | They didn't work apparently |
| 15:43.32 | igcewieling | then I doubt anything else will work. restart the pbx |
| 15:43.34 | Greenlight | Guessing from [TK]D-Fender's advice that the AMI redirect is more likely to succeed |
| 15:43.50 | Greenlight | It may masqurade the channel away |
| 15:43.57 | Greenlight | And maybe leave the masqurade behind |
| 15:44.14 | Greenlight | Which might fix his hint issue |
| 15:44.20 | [TK]D-Fender | [11:39]ipalmerOk sounds good, what would the command be to do this if for instance the stuck channel was SIP/1234 <- this would be the point where you read the book/wiki which explains the syntax for all of this already |
| 15:44.23 | igcewieling | I'm doubtful, but [TK]D-Fender generally know what he is talking about |
| 15:44.31 | [TK]D-Fender | ~asterisk-wiki |
| 15:44.34 | [TK]D-Fender | ~asteriskwiki |
| 15:44.35 | infobot | hmm... asteriskwiki is http://wiki.asterisk.org |
| 15:44.39 | [TK]D-Fender | ~book |
| 15:44.40 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 15:49.35 | igcewieling | Greenlight: what is wrong with "investigate"? Makes it sound like you are doing something when you are really the trouble report because it is less coherent than the frickin tax code. |
| 15:49.53 | igcewieling | s/when you are really/when you are really ignoring the report/ |
| 15:50.32 | igcewieling | I got a trouble report today involving a number which was disconnected in April 8-| |
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| 15:54.40 | Greenlight | What sort of trouble report ? |
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| 15:57.41 | igcewieling | Greenlight: can't call out |
| 15:58.40 | Greenlight | Sounds like an easy one to solve, if it was disconnected in April :) |
| 16:00.11 | igcewieling | Greenlight: indeed if I didn't spend an hour looking through our systems for the reported calls. |
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| 16:13.59 | rayz24_ | Is there anyway to get a list of available sounds to play via the AMI? |
| 16:14.21 | Greenlight | You can play any sound you want |
| 16:14.31 | Greenlight | There is no "list" ? |
| 16:14.35 | rayz24_ | I know, but I mean a list so I know what I can play. |
| 16:14.48 | rayz24_ | Dynamically. |
| 16:15.01 | igcewieling | rayz24_: no. |
| 16:15.05 | Greenlight | You can play any sound file you want |
| 16:15.06 | [TK]D-Fender | rayz24_: Look in your "sounds" folder |
| 16:15.18 | [TK]D-Fender | rayz24_: That is typically what you "have".... |
| 16:15.21 | rayz24_ | So I'd need to have shell access then |
| 16:15.29 | rayz24_ | I'm accessing the AMI remotely. |
| 16:15.30 | Greenlight | Not neccissari;y |
| 16:15.32 | igcewieling | rayz24_: the sound files are by default located in /var/lib/asterisk/sounds. You can build a dynamic list using an AGI if you need. |
| 16:16.03 | igcewieling | you could also parse the *.txt files located in your sounds directory |
| 16:16.25 | johnwigley | Is there anyway to make callers leave a queue at a fixed time, for example, at 5.30pm any callers left in the queue are dumped out to an exit context or extension? |
| 16:16.32 | igcewieling | rayz24_: if you don't have shell access to the server, then we cannot help you |
| 16:16.46 | rayz24_ | I have shell access, but my application wouldn't. |
| 16:17.13 | igcewieling | rayz24_: I still don't understand what the problem is. |
| 16:17.16 | rayz24_ | But before I do something complex, just wanted to make sure there wasn't an easy way to do it via AMI |
| 16:17.17 | Greenlight | Me neither |
| 16:17.36 | igcewieling | log into the server, do an ls -l /var/lib/asterisk/sounds > /path/to/your/soundslist.txt DONE! |
| 16:17.38 | Greenlight | The AMI is not designed to list directory contents |
| 16:17.52 | Greenlight | Using it for such, is going to be hacky |
| 16:17.56 | rayz24_ | I am going to have a list of sounds someone can whisper to an agent. IF someone comes along and adds a new sound to the /sounds directory, I want that file to show up in my list. |
| 16:18.01 | Kobaz | Greenlight: and *really* slow |
| 16:18.08 | Greenlight | mmhmm |
| 16:18.15 | Greenlight | So, monitor the directory |
| 16:18.16 | igcewieling | rayz24_: you should not add stuff to sounds/ |
| 16:18.35 | rayz24_ | custom directory still gives me teh same issue. |
| 16:18.38 | Greenlight | Wait, so your users have access to the directory, but your app doesnt? |
| 16:18.42 | igcewieling | rayz24_: exactly how are they "adding sounds"? |
| 16:18.42 | Greenlight | That's silly |
| 16:18.50 | rayz24_ | No, not the users. The guy I'm making the app for. |
| 16:19.01 | Greenlight | Make him maintain a database then |
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| 16:19.18 | [TK]D-Fender | [12:17]rayz24_I am going to have a list of sounds someone can whisper to an agent. IF someone comes along and adds a new sound to the /sounds directory, I want that file to show up in my list. <- these are just files. There is no magic to knowing that files are being dropped in behind the scenes. Lean linux. This has nothing to do with Asterisk. |
| 16:19.24 | igcewieling | rayz24_: We wish you the BEST of luck. |
| 16:19.28 | [TK]D-Fender | rayz24_: Asterisk "detects" nothing |
| 16:19.45 | [TK]D-Fender | rayz24_: It plays what you tell it to play and you'd better know that what you want is actually there |
| 16:19.51 | rayz24_ | gotcha |
| 16:19.54 | Greenlight | Yup this is more general filesystem stuff... nothing Asterisk related |
| 16:20.08 | rayz24_ | I understand that |
| 16:20.30 | igcewieling | in the dialplan you have a bit more flexibility |
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| 16:59.31 | growse | Anyone know if it's possible to force asterisk to connect to a peer over IPv6 if the peer has both IPv4 and IPv6 addresses? |
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| 17:14.28 | mmlj4 | use the ipv6 IP? |
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| 17:56.33 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:47.03 | *** join/#asterisk infobot (~infobot@184.232.247.220) |
| 18:47.03 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 21:56.37 | *** join/#asterisk infobot (~infobot@174-27-225-67.slkc.qwest.net) |
| 21:56.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |