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| 00:07.38 | WIMPy | Features don't break native bridging. |
| 00:07.47 | WIMPy | Not waiting for them anyway. |
| 00:09.33 | bipolar | This particular sets the call to be recorded with an in-call feature code. If I enable it, it breaks the native bridge. Might be a bug in the macro. We don't use it anyway, so I'll just leave it turned off. |
| 00:10.31 | WIMPy | Recording will break the native bridge. |
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| 00:19.38 | bipolar | Yep. It makes sense. Just sucks I can't have a global option like that without breaking a native bridge that would never use it. |
| 00:21.28 | WIMPy | That doesn't make sense to me. As long as you don't use it, it should be able to use the native bridge. |
| 00:21.59 | WIMPy | Unless that bridge decides to opt out. What channel are we talking about? |
| 00:24.26 | WIMPy | Ah. Looks like chan_dahdi. Yes, that declines a native bridge when features are activated, most probably because of the issues I'm seeing. |
| 00:24.40 | bipolar | yep |
| 00:24.47 | WIMPy | But the comment in chan_dahdi about optiong out is not very informative. |
| 00:25.21 | bipolar | Again, if I could just turn off DTMF mangling I'd be golden. lol |
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| 00:28.57 | WIMPy | Maybe this wasn't disabled inolder versions. |
| 00:29.21 | WIMPy | The code to handle it clearly exists. |
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| 02:32.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:48.12 | gavimobile | my telco company wants me to open rtp ports from 10000-50000. I currently only have 10000-10200 I never have more than 2 concurrent calls. do I need to open all 50,000? |
| 06:53.27 | wdoekes | gavimobile: your end decides on which ports you *receive*, so you only need to open up *inbound* ports that you like |
| 06:54.01 | Demon_VoIP | gavimobile, rtp.conf rtpstart & rtpend |
| 06:54.02 | wdoekes | as for *outbound*, you may need to allow many/all. but you probably allow all outbound ports anyway |
| 07:00.21 | gavimobile | so will I have audio problems if I only open up 200 ports? |
| 07:00.37 | gavimobile | all outbound is allowed, yes |
| 07:01.37 | wdoekes | no you won't |
| 07:01.45 | wdoekes | assuming your firewall and rtp.conf are in sync |
| 07:01.49 | gavimobile | thanks wdoekes |
| 07:01.55 | gavimobile | yes they are |
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| 10:41.37 | madduck | if I Goto(i,1), I lose the content of ${EXTEN}. Is there another way to just handle an extension as if it was invalid? i.e. from a GotoIf? |
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| 14:03.53 | madduck | we are experiencing very intermittent issues with some calls and cannot discern a pattern or isolate them. One suggestion was to provide a means by which e.g. the user can press '*' when a problem occurs and be dropped into a context where debug info are saved and the user is prompted for a problem description |
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| 14:04.27 | madduck | do I need the G dial flag? |
| 14:05.48 | madduck | also, from a context, can I just jump to treating an extension as invalid, e.g. after running a few tests and using GotoIf? I thought I'd use "i" as target (which works), but then I lose the ${EXTEN} info useful in the logs and for debuggung |
| 14:07.20 | WIMPy | Your questions are pretty confusing at best. |
| 14:07.32 | madduck | sorry :( |
| 14:07.39 | WIMPy | If you decide that you don't want to accept a call, you use Hangup(). |
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| 14:39.38 | madduck | WIMPy: pertaining to the other question, let's say I use Dial(â¦) and something doesn't work, e.g. I cannot hear the other side. I'd like to be able to press '*' now and have Asterisk $DoSomething, i.e. jump into another context without disconnecting the channel. Is this possible? |
| 14:39.55 | madduck | I could Dial(â¦, g) and after the callee hangs up, then WaitExten etc. |
| 14:40.06 | madduck | but what if I don't want to wait for the other side to hang up? |
| 14:41.24 | [TK]D-Fender | madduck: "core show application dial" <- These are your options.... plus whatever you create in features.conf |
| 14:41.48 | [TK]D-Fender | madduck: This shouldn't be a question. The instructions saying what Dial can do are right there |
| 14:42.22 | madduck | oh wow, so far I always loaded voip-info.org for docs ;) |
| 14:42.26 | madduck | i did not know there are online docs |
| 14:43.06 | [TK]D-Fender | Voip-info is amongst the LAST places to look |
| 14:43.13 | [TK]D-Fender | * always had internal instruction. |
| 14:43.19 | [TK]D-Fender | And there is an official wiki |
| 14:43.32 | [TK]D-Fender | and semi-official book series |
| 14:43.54 | madduck | i have the o'reilly book but i don't recall it telling me about this. |
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| 14:47.19 | superscrat | madduck, look on page 74 of the 4th edition |
| 14:47.54 | madduck | "Upgrading Asterisk"? |
| 14:47.55 | superscrat | that is the first time it is mentioned anyway. it's at the top of the page |
| 14:48.02 | madduck | aaah |
| 14:48.13 | madduck | i skipped the whole chapter due to Debian |
| 14:48.26 | madduck | page 203 too |
| 14:48.30 | superscrat | then again at 203 |
| 14:48.34 | superscrat | yes, that's correct |
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| 14:57.24 | madduck | so I looked through Dial() and I cannot find anything really. features.conf can invoke applications, but no logic e.g. WaitExten or Goto. |
| 14:57.48 | madduck | The thing I am unsure about is the G flag, which puts caller/callee into a dialplan context. I will research this. |
| 14:57.59 | madduck | Does anyone have a good example to illustrate this feature? |
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| 14:58.43 | [TK]D-Fender | It is EXACTLY what it says. |
| 14:58.47 | [TK]D-Fender | What part is unclear? |
| 14:58.57 | madduck | i just can't wrap my head around it I guess |
| 14:59.01 | madduck | I will try it out. |
| 14:59.15 | madduck | sorry, don't mean to be sounding thick, but it's been a while since I did real asterisk work |
| 14:59.52 | [TK]D-Fender | "If the call is answered, transfer the calling party to the specified <priority> and the called party to the specified <priority> plus one." |
| 14:59.55 | [TK]D-Fender | How is that not clear? |
| 15:00.08 | [TK]D-Fender | When they answer... BOTH ends are redirected ELSEWHERE in the dialplan |
| 15:00.10 | [TK]D-Fender | They are NOT connected |
| 15:00.15 | [TK]D-Fender | It is EXACTLY what it says |
| 15:00.25 | madduck | ah okay |
| 15:00.37 | madduck | i was thinking that they'd be connected and then the dialplan continues |
| 15:00.50 | madduck | which would be interesting, just not something I was able to imagine |
| 15:00.52 | [TK]D-Fender | That clearly has nothing to do with "let me interrupt this call in the middle based on DTMF" |
| 15:01.05 | [TK]D-Fender | You're inventing things it is not saying. |
| 15:01.16 | [TK]D-Fender | Don't imagine that it does "more". |
| 15:01.17 | madduck | i'll stop ;_) |
| 15:01.42 | [TK]D-Fender | Everything else that happens is you doing more dialplan apps, etc |
| 15:01.51 | madduck | though it is a bit unclear. "if the call is answered" does imply a little bit that there is a connection |
| 15:01.53 | cbeyerlein | Hey there. I cleaned up the sample-configuration of Ast 13.3.2 to remove all the default stuff that I don't want. |
| 15:02.10 | cbeyerlein | But now when I start Asterisk, the CLI prompt loops: *CLI> *CLI> *CLI> *CLI> .... |
| 15:02.17 | cbeyerlein | and CLI thread consumes 100% cpu :( |
| 15:02.31 | cbeyerlein | any idea which of the dozens config-files could be the trigger? :) |
| 15:03.08 | [TK]D-Fender | madduck: Nope. Just because they answered does not mean the call is BRIDGED |
| 15:03.15 | superscrat | cbeyerlein, which samples did you 'clean up'? all of them? |
| 15:03.35 | madduck | [TK]D-Fender: have patience with us newbies! ;) |
| 15:03.40 | [TK]D-Fender | madduck: See M() |
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| 15:04.06 | superscrat | madduck, what are you trying to accomplish? |
| 15:04.24 | cbeyerlein | superscrat: I started out with moving the config files of modules I don't need... And then iteratively moving them back (and cleaning up) when on startup some ERROR is logged... my goal is to remove default config, but not having any ERROR in the logs |
| 15:04.37 | cbeyerlein | because I want to monitor logfiles for ERROR in the future |
| 15:04.58 | superscrat | okay, what I think you should do, is start back with the default state of the logs, and move one log at a time. |
| 15:05.10 | superscrat | too many moving parts can make debugging difficult |
| 15:05.29 | madduck | [TK]D-Fender: that does not seem to work *during* the bridged call |
| 15:05.55 | cbeyerlein | yep, will probably do that. just wanted to check if someone happend to have looping CLI too and found a culprit |
| 15:06.02 | madduck | superscrat: to put simply: during a call, I'd like to be able to execute another context |
| 15:06.04 | superscrat | err, logs, sorry cbeyerlein, I meant, configs |
| 15:06.09 | [TK]D-Fender | madduck: So your options are to use the DTMF option to hangup the call and capure at the end, or a features.conf applicationmap |
| 15:06.11 | madduck | even shift control to the other context |
| 15:06.23 | superscrat | ah, okay. |
| 15:06.30 | madduck | alright, I will play around more. |
| 15:06.35 | madduck | must go make dinner now and then be back later |
| 15:06.39 | [TK]D-Fender | madduck: I never said it worked DURING a call. It was a sample of things that happen AFTER the called channel ANSWERS ... and clearly before it is BRIDGED |
| 15:06.44 | madduck | superscrat: if you have any ideas, I will read scrollback |
| 15:07.04 | [TK]D-Fender | madduck: To exemplify that you shouldn't assume things happen only when bridged |
| 15:07.23 | superscrat | okay, looking now madduck. tk is probably correct, but I will read through the previous messages. |
| 15:12.13 | superscrat | to execute another extension inside of another context, you can inside of an extension, use GoSub. |
| 15:12.19 | superscrat | madduck |
| 15:13.00 | *** part/#asterisk cervajs2 (~cervenka@gatekeeper.bm.ipex.cz) |
| 15:13.29 | superscrat | sorry that was written odd, grammatically. https://wiki.asterisk.org/wiki/display/AST/GoSub |
| 15:14.46 | [TK]D-Fender | And nope, neither of these are what he's looking for |
| 15:16.09 | superscrat | cbeyerlein, I have not ever seen that - but I have seen some odd behavior when I messed up configs. so, it wouldn't surprise me if you have some settings that are stepping on each other. |
| 15:17.02 | cbeyerlein | ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0x7fff0185ebb0) = -1 ENOTTY (Inappropriate ioctl for device) |
| 15:17.03 | cbeyerlein | write(1, "\0", 1) = 1 |
| 15:17.03 | cbeyerlein | write(1, "*CLI> ", 6) = 6 |
| 15:17.06 | cbeyerlein | this scares me... |
| 15:17.33 | cbeyerlein | thats all the cli thread is doing |
| 15:17.40 | superscrat | okay, that's fair. I was answering "to put simply: during a call, I'd like to be able to execute another context" - which is what I understood the question to be. [TK]D-Fender, can you restate what he is asking, because I joined the party a bit late and I think I am lacking enough context. |
| 15:18.20 | [TK]D-Fender | superscrat: That's why they invented scroll-back and logging. So you can like .... read what he was asking for and stuff and get context ;) |
| 15:18.31 | cbeyerlein | with FD 0 being 0 -> [timerfd], 1 -> /dev/pts/0 |
| 15:18.46 | [TK]D-Fender | superscrat: He want's to INTERRUPT a call in progress via DMTF so he can fill in logging info as to crap quality issues, etc |
| 15:18.56 | [TK]D-Fender | superscrat: With no mention of having to return to that call. |
| 15:20.47 | superscrat | cbeyerlein, is it your dahdi configuration? |
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| 15:21.18 | cbeyerlein | superscrat, I had it in question, but but noload => chan_dahdi.so into modules.conf... |
| 15:21.20 | superscrat | [TK]D-Fender, ahhhh, okay. Thank you. |
| 15:23.08 | marceloamorim | guys, using asterisk 13.3.2 when I get a mp3 file and use the audacity to change to wav, and use the command sox file_audacity.wav -r 8000 -e signed-integer -c 1 file_sox.wav. After that I try to app Playback(file_sox), the ast_openstream_full send to me the log, file file_sox does not exist in any format. The sox -V option to this file http://pastebin.com/LC4w4E0L |
| 15:23.53 | marceloamorim | any suggestion ? |
| 15:24.15 | cbeyerlein | just tried with the original sample-configuration, it does not occur.. |
| 15:24.35 | cbeyerlein | I will iteratively cleanup the config, and when I find the culprit I will come back :) |
| 15:26.03 | superscrat | [TK]D-Fender, just curious, why couldn't he use WaitExten - caveat, I am sure this is not a viable solution, this is for my information purposes only. |
| 15:26.51 | superscrat | cbeyerlein =) great. glad to hear it wasn't anything worse than a setting (or settings) somewhere |
| 15:28.30 | madduck | [TK]D-Fender: no need to return to the call really. I think one option would be to force-hangup the other side and then to continue after Dial() |
| 15:36.25 | [TK]D-Fender | superscrat: You can't go running Waitexten WHILE talking to the other person in a connected call |
| 15:37.05 | [TK]D-Fender | madduck: Test the hangup option to see if it allows you to just dump the callee and leave a variable you can detect the reason for so you can catalog it. |
| 15:38.16 | cbeyerlein | I got it. Its triggered by an empty hep.conf |
| 15:40.05 | superscrat | [TK]D-Fender, <insert face-palm here>, yes, of course. Thanks for clearing that up for me. |
| 15:41.32 | superscrat | cbeyerlein, interesting! I don't think that I have ever included that file in my development configurations. Good to know =) |
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| 15:42.56 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 15:44.06 | cbeyerlein | superscrat, well, I consider this a BUG... |
| 15:49.53 | cbeyerlein | luckily easy to reproduce :) empty or incomplete hep.conf |
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| 16:04.17 | cbeyerlein | hm. looking in res_hep.c load_module() I wonder if the hep_queue_tp should be derefenced when goto error; like in unload_module.... |
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| 16:21.29 | themrrobert | since adding MALLOC_DEBUG to the compile flags, asterisk seems to be crashing more often lol |
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| 16:42.38 | superscrat | sorry that I seemed to disappear; I was driven outside by some repairmen. at least it's nice outside today. |
| 16:46.16 | superscrat | cbeyerlein, you still here? |
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| 17:38.56 | hiroprot | Hey everyone. So I'm trying to sketch out the hardware and software requirements for a small phone system in a permanent art installation. I want to use vintage analog phones, play recorded dial tone, real DTMFs the whole shebang. I'm pretty new to PBXs, but have a rudimentary understanding and I'm trying to fill in the wholes. I want a central server to connect calls within the system of maybe 8 phones. It does not necessarily have to be ab |
| 17:38.56 | hiroprot | le to connect to the outside world, but there may be a larger PBX for the building that if it could connect to that would be cool. |
| 17:39.35 | hiroprot | The main conditions are that 1) I want to be able to use analog phones 2) I don't want to spend $500 on a card for 8 lines |
| 17:39.51 | hiroprot | is there any way to do this with asterisk using older telephony cards? |
| 17:47.12 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:47.12 | robmal | hiroprot: Cisco SPA8000 should be enough. |
| 17:48.06 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 17:48.09 | drmessano | You need a card for 8 lines or an ATA/channel bank.. Asterisk has no magic to make that not happen |
| 17:50.59 | drmessano | hiroprot, http://www.ebay.com/itm/HT-882-FXS-VoIP-Gateway-with-8-FXS-Ports-VoIP-Gateway-HT-882-ATA-Gateway-SIP-/121584686733?pt=LH_DefaultDomain_0&hash=item1c4f03168d |
| 17:51.09 | drmessano | $153.79 |
| 17:51.15 | eppigy | hello |
| 17:51.17 | eppigy | i am dave |
| 17:51.18 | drmessano | Looks like junk |
| 17:51.20 | drmessano | But hey |
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| 17:57.42 | hiroprot | and those units are totally compatible with asterisk? I haven't been able to get a sense of just what hardware is compatible. The asterisk site does not list much. |
| 17:58.45 | *** part/#asterisk riess82 (~riessma@188-23-127-19.adsl.highway.telekom.at) |
| 17:59.30 | hiroprot | that was directed at you, robmal |
| 17:59.49 | drmessano | The SPA8000 is SIP |
| 17:59.52 | drmessano | and works with Asterisk |
| 18:00.04 | drmessano | Though you can wait for him to answer and tell you the same |
| 18:00.26 | hiroprot | drmessano, the unit you posted a link for, will that work with any analog phone? |
| 18:00.35 | drmessano | Yes |
| 18:00.51 | hiroprot | also, if i wanted to up the number of phones to 12 or so, are there any affordable options? |
| 18:01.01 | drmessano | Nope |
| 18:01.11 | hiroprot | anyway to use two of the 8 line units? |
| 18:01.20 | drmessano | Yep, buy two, configure them |
| 18:03.11 | drmessano | I would probably recommend that anyway for $153. I can't imagine the quality will be high.. If you expect to need 12 or so, you should invest in a decent 16 channel channel bank |
| 18:03.20 | drmessano | But it will cost you more than $500 |
| 18:03.48 | hiroprot | got a product recommendation on that? |
| 18:04.25 | drmessano | Mediatrix 4116 |
| 18:04.31 | hiroprot | and when you refer to quality, are you referring to audio quality, or overall bugginess of the product |
| 18:04.48 | drmessano | Both |
| 18:05.37 | drmessano | Little under $700 for the Mediatrix |
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| 18:16.56 | hiroprot | Thanks for you help. I think I've got a good place to start. Any good documents explaining the setup procedure in a little greater detail than some of the rather superficial articles I've seen about asterisk? I also would like if possible to have a number dialable from these phones that has a automated menu, know where I could find some info on that? |
| 18:19.06 | drmessano | ~book |
| 18:19.07 | 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 |
| 18:29.17 | superscrat | hiroprot, |
| 18:30.28 | superscrat | I was new to asterisk about 7 mos ago. I found the ~book to be very useful. Basically, I just walked through setting up a pbx following the author the whole time. |
| 18:30.39 | superscrat | that is also how I learned how to manipulate the dialplan |
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| 19:11.51 | nixhr | does asterisk mysql dialplan command support SSL? |
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| 19:58.00 | cervajs2 | <PROTECTED> |
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| 20:21.05 | *** join/#asterisk Quo-fan (quofan@unaffiliated/quo-fan) |
| 20:21.45 | Quo-fan | hey I need quick howto on how to add phone number extensions and SIP accounts to asterisk manually |
| 20:21.55 | Quo-fan | cos theres no such quick howto available |
| 20:22.19 | Quo-fan | and I'm using debian and both freepbx and asterisk-gui refuse to work with debian |
| 20:22.22 | [TK]D-Fender | Because there is no such thing as "add phone number extensions" |
| 20:22.26 | Quo-fan | so im out of options |
| 20:22.54 | Quo-fan | all howtos tell me to install some stupid webgui that fails to run on debian |
| 20:22.55 | [TK]D-Fender | AsteriskGUI has been dead for years, and FreePBX has guides for Debia |
| 20:23.22 | [TK]D-Fender | FreePBX's official WIKI has instructions for it |
| 20:23.30 | Quo-fan | I tried to freepbx |
| 20:23.34 | [TK]D-Fender | As for DIY ... well you need to elan Asterisk itself. |
| 20:23.47 | [TK]D-Fender | And no "asterisk from scratch" guide would ever mention a GUI |
| 20:23.54 | [TK]D-Fender | So I have no idea what you were googling for |
| 20:23.57 | Quo-fan | the install script gives error cannot connect to mysql even tho i have mysql and userid and database added to mysql |
| 20:24.07 | [TK]D-Fender | Keeping in mind we've had a series of nice books for a long time now |
| 20:24.10 | [TK]D-Fender | ~book |
| 20:24.10 | 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 |
| 20:24.12 | [TK]D-Fender | ^^^ |
| 20:24.32 | [TK]D-Fender | If you're down to a MySQL error then that should be petty to fix |
| 20:25.02 | Quo-fan | i deleted freepbx cos it didnt work |
| 20:25.08 | [TK]D-Fender | It works |
| 20:25.12 | Quo-fan | and theres no freepbx package for debian even |
| 20:25.16 | [TK]D-Fender | sounds like 1 step got fudged |
| 20:25.31 | [TK]D-Fender | but if you're going DIY... you'r going to have to learn how to actually configure Asterisk |
| 20:25.37 | Quo-fan | debian hasnt got it in the repositories cos the developers of debian are stiff ass lamos |
| 20:26.15 | Quo-fan | i tried removing freepbx and installing it again |
| 20:26.30 | Quo-fan | but it has some hidden config file somewhere that cocks it up |
| 20:27.17 | Quo-fan | those webguis are pretty much all a big failure |
| 20:27.23 | [TK]D-Fender | Thre's nothing magical hidden |
| 20:27.23 | Quo-fan | and big disappointed |
| 20:27.30 | [TK]D-Fender | but if you've already wiped it then lets move on |
| 20:27.52 | Quo-fan | I want the sip server to work |
| 20:27.54 | Quo-fan | but no |
| 20:27.57 | [TK]D-Fender | FreePBX works fine for pretty much everybody else. Your call. |
| 20:28.05 | Quo-fan | doesnt wanna work |
| 20:28.17 | [TK]D-Fender | Well if you aren't looking to actively debug it... lets move on |
| 20:28.27 | Quo-fan | looks like debian doesnt fully support it |
| 20:28.41 | Quo-fan | cos they dont even have freepbx as a debian package |
| 20:29.02 | [TK]D-Fender | Packaging means nothing |
| 20:29.12 | [TK]D-Fender | Using this as a basis is faulty |
| 20:29.17 | Quo-fan | it would make work with lot less effort |
| 20:29.24 | Quo-fan | if the debian package existed |
| 20:29.54 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 20:30.07 | [TK]D-Fender | Yes, but that has never been the case |
| 20:30.17 | [TK]D-Fender | So again, your call |
| 20:30.50 | sgriepentrog | Quo-fan: http://wiki.freepbx.org/display/HTGS/Installing+FreePBX+on+Debian+Wheezy -- although I recommend using the freepbx distro instead, much easier. |
| 20:31.51 | Quo-fan | sgriepentrog: I have debian already installed in a virtualization host... and centos is the wrong base distro for freepbx anyway |
| 20:31.57 | Quo-fan | debian is much more versitile |
| 20:32.14 | [TK]D-Fender | How is it "wrong for FreePBX"? |
| 20:32.40 | Quo-fan | debian would be the right distro for freepbx cos its more commonly used |
| 20:32.53 | Quo-fan | centos is a marginal distro |
| 20:32.56 | [TK]D-Fender | That's what the dev team actually built their distro around. Seems rather "right" in that they did it that way and support it. |
| 20:33.07 | sgriepentrog | There are several who would argue with you on that, and I will decline. |
| 20:34.05 | [TK]D-Fender | Yeah we could get into the full "CentOS is effectively RHEL so the entire corporate world and a billion dollar company as a basis makes that arguement null and void" comes to mind |
| 20:34.21 | [TK]D-Fender | But whatever |
| 20:34.26 | [TK]D-Fender | Don't want to run FreePBX? Fine |
| 20:34.32 | [TK]D-Fender | You decision. So go learn Asterisk |
| 20:34.57 | [TK]D-Fender | And have fun with the power (and responsibility) that DIY gives you |
| 20:35.41 | *** part/#asterisk Quo-fan (quofan@unaffiliated/quo-fan) |
| 20:35.41 | [TK]D-Fender | Checkout time.. heading home, BBIAB |
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| 21:31.49 | Quo-fan | i got the asterisk working no thanks to this channel |
| 21:31.58 | Quo-fan | but freepbx wasnt the answer |
| 21:32.18 | Quo-fan | asteriskgui was the only one that would work with debian |
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| 21:32.39 | Iamnacho | "no thanks to this channel" LOL |
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| 21:46.53 | carrar | NO |
| 21:46.55 | carrar | THANK YOU |
| 21:47.34 | carrar | such sadness |
| 21:47.37 | carrar | heh |
| 21:48.04 | carrar | I must have missed that persons initial channel questions |
| 21:48.19 | carrar | Oh it's in my scroll back |
| 21:49.40 | carrar | oh there we have it |
| 21:49.41 | carrar | <Quo-fan> hey I need quick howto on how to add phone number extensions and SIP accounts to asterisk manually |
| 21:50.35 | carrar | ~solder |
| 21:50.51 | carrar | guess that was removed |
| 21:51.31 | carrar | https://www.osburn.com/asterisk-churchsign.png |
| 21:51.51 | carrar | oh there is the one I was looking for https://www.osburn.com/asterisk-sign.png |
| 21:52.01 | carrar | haha |
| 21:52.06 | carrar | that never stops beign true |
| 21:52.19 | carrar | -rw-r--r-- 1 osburn www 393466 Nov 8 2008 asterisk-sign.png |
| 21:53.09 | [TK]D-Fender | ~savemoney |
| 21:53.09 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 21:53.11 | [TK]D-Fender | ^^^ |
| 21:53.18 | carrar | haha yeah there it is |
| 21:54.26 | [TK]D-Fender | Nah, this guy came in with distro-flamewar-FUD and a sense that just because he failed that you couldn't install FreePBX on Debian. And that CentOS wasn't a "real serious" distro and that FreePBX should have packages for Debian. |
| 21:54.42 | WIMPy | That's what they call a telco grade phone line these days... |
| 21:55.50 | carrar | yeah all the centos servers I have are not real |
| 21:56.03 | WIMPy | All virtualized? |
| 21:56.08 | WIMPy | scnr |
| 21:56.10 | carrar | heh |
| 21:56.12 | carrar | some are |
| 21:56.29 | carrar | actually most are |
| 21:57.17 | carrar | and now that systemd is in centos.... |
| 21:57.24 | carrar | *sigh* |
| 21:57.44 | WIMPy | That's somethig I don't want to see on any server. |
| 22:02.03 | WIMPy | is already pretty pissed about the fact that you need to have udevd running to use Digium cards. |
| 22:05.18 | carrar | People still use analog cards!!! :) |
| 22:05.39 | WIMPy | Analog? What's that? |
| 22:05.45 | carrar | well T1 cards |
| 22:05.54 | WIMPy | Oh, the cound cards. |
| 22:06.13 | carrar | none sip |
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| 23:57.14 | kplong | Hello- just configured TLS on my asterisk server and installed CA cert and client certs on my phones. When I call phone to phone it rings but when I pick up theres no audio and I get the error in my "full" log : chan_sip.c: Rejecting secure audio stream without encryption details: audio 11780 UDP/TLS/RTP/SAVP 0 101 |
| 23:57.24 | kplong | Any hints appreciated ^ |
| 23:57.47 | kplong | I'm trying to do TLS and SRTP |
