| 00:00.01 | voipeng | gotcha |
| 00:00.06 | voipeng | ill pick up here at home then |
| 00:00.09 | voipeng | thank ou |
| 00:00.15 | voipeng | thank you both* |
| 00:00.18 | dapsaille | p3nguin > does dahdi is compiled by asterisk or it's own sources ? |
| 00:00.22 | *** join/#asterisk kaushal (~kaushal@115.118.244.37) |
| 00:00.38 | navaismo | own sources |
| 00:00.43 | dapsaille | thanks |
| 00:00.49 | p3nguin | You have to build dahdi from its own source. |
| 00:00.54 | p3nguin | It is not part of asterisk. |
| 00:01.56 | *** part/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 00:04.50 | kaushal | Hi |
| 00:06.06 | kaushal | if i do dialplan reload at CLI> , would it break the current ongoing session and is there a way to verify it and what exactly behind the scene once i shoot dialplan reload ? |
| 00:06.19 | kaushal | ^happens |
| 00:06.42 | kaushal | little inquistive about this use case |
| 00:06.56 | p3nguin | No, it will not interrupt a call. |
| 00:07.27 | kaushal | p3nguin: ok |
| 00:07.30 | p3nguin | But if that current call progresses the dial plan, any changes you have made to that part of the dial plan WILL BE USED. |
| 00:08.14 | kaushal | ok |
| 00:08.21 | p3nguin | If you want to see the dial plan, use "dialplan show" to see it all. |
| 00:09.01 | kaushal | more examples about "current call progresses the dial plan" ? |
| 00:09.09 | kaushal | not sure i understand that bit |
| 00:10.02 | p3nguin | Let us say that your caller is sitting in a BackGround or Playback right now, listing to a long audio file playing... |
| 00:10.41 | *** join/#asterisk fofware (~fabian@host26.186-108-159.telecom.net.ar) |
| 00:10.54 | *** join/#asterisk darkskiez_ (~dz@cpc4-broo7-2-0-cust167.14-2.cable.virginmedia.com) |
| 00:11.17 | p3nguin | If the next step in the dial plan was Goto(someplace) |
| 00:11.24 | WIMPy | I never tried it, but didn't I read somewhere that existing calls will continue in the old dialplan? |
| 00:11.43 | p3nguin | But then you change it to Hangup() and run dialplan reload |
| 00:11.57 | p3nguin | When the audio file is done, the call will hangup, not Goto someplace. |
| 00:12.22 | p3nguin | The call will use the new existing dial plan. |
| 00:12.31 | p3nguin | It just will not interrupt the existing application. |
| 00:14.04 | p3nguin | Once you dialplan reload, there is no "old dialplan" left. It's gone. There is only the currently loaded dial plan. |
| 00:14.34 | kaushal | ok |
| 00:14.38 | *** join/#asterisk psharmor (~quassel@97-118-218-173.hlrn.qwest.net) |
| 00:15.48 | kaushal | p3nguin: is there a explanation on asterisk documentation or docs ? |
| 00:15.52 | kaushal | i mean wiki |
| 00:16.00 | p3nguin | ~wiki |
| 00:16.18 | p3nguin | well wtf |
| 00:16.23 | p3nguin | ~asteriskwiki |
| 00:16.23 | infobot | i heard asteriskwiki is http://wiki.asterisk.org |
| 00:17.58 | kaushal | p3nguin: thanks |
| 00:18.02 | kaushal | yes its mentioned |
| 00:18.13 | kaushal | If you change the dialplan, you can use the Asterisk CLI command "dialplan reload" to load the new dialplan without disrupting service in your PBX. |
| 00:18.29 | kaushal | as per https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Dialplan |
| 00:18.43 | kaushal | p3nguin: Thanks for the explanation |
| 00:26.12 | *** join/#asterisk TimeRider (~steve@92.40.247.208.threembb.co.uk) |
| 00:32.59 | patrickod | when using realtime extensions in a mysql DB what's the best way to have each context inherit a set of extensions |
| 00:33.02 | patrickod | ? |
| 00:42.28 | SeRi | p3nguin: to see if G722 is working on my new phone can I call your conf? |
| 00:42.41 | p3nguin | Sure. |
| 00:42.50 | krotos | guy |
| 00:42.52 | SeRi | Thanks p3nguin |
| 00:42.55 | SeRi | one sec |
| 00:42.58 | krotos | how can i launch asterisk |
| 00:43.01 | krotos | in core-dump |
| 00:43.01 | krotos | mode |
| 00:43.30 | krotos | my asterix boxes become unaccessibile ( 100%cpu) and i've got to restart the vm |
| 00:43.48 | krotos | i don't understand why |
| 00:47.36 | navaismo | krotos: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 00:49.44 | krotos | the core dump where is saved? |
| 00:52.36 | krotos | becasue if it save in /tmp |
| 00:52.45 | krotos | and i've got to reboot the machine |
| 00:52.48 | krotos | i loose the core-dump |
| 00:54.43 | WIMPy | Either the directory from where Asterisk was started or the users home directory. |
| 00:57.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 01:02.00 | *** join/#asterisk UnixDev (unixdev@unaffiliated/unixdev) |
| 01:03.23 | UnixDev | hi, I think I found a bug in asterisk… using 'SVN-branch-1.8-r345976M' … this seems to be an issue inside chan_sip and has to do with reinvites, where a peer becomes 'lagged' when it really is not |
| 01:15.12 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 01:16.51 | *** join/#asterisk tekoholic (~quassel@97-118-209-106.hlrn.qwest.net) |
| 01:26.43 | SeRi | UnixDev: go to #asterisk-dev to report |
| 01:27.29 | WIMPy | ... but don't expect an answer before monday. |
| 02:15.51 | SeRi | nom nom nom flautas |
| 02:16.55 | *** join/#asterisk TimeRider (~steve@92.40.247.208.threembb.co.uk) |
| 02:17.37 | p3nguin | steals seri's flutes |
| 02:19.33 | SeRi | ha! |
| 02:19.54 | SeRi | their mine! |
| 02:19.58 | SeRi | lol :P |
| 02:20.16 | SeRi | wrestle p3nguin for the flautas |
| 02:22.58 | *** join/#asterisk master_of_master (~master_of@p57B52AEB.dip.t-dialin.net) |
| 02:30.25 | *** join/#asterisk box (~box@unaffiliated/box) |
| 02:30.26 | box | hello |
| 03:31.52 | *** join/#asterisk bloudermilk (~bloudermi@pool-108-38-59-34.lsanca.fios.verizon.net) |
| 03:33.16 | bloudermilk | Evening all. Is it possible to get ISUP release codes in Asterisk? |
| 03:33.52 | WIMPy | HANGUPCAUSE |
| 03:34.18 | WIMPy | But no location. |
| 03:35.40 | bloudermilk | No location? (Just learning about ISUP... please forgive me) |
| 03:36.42 | WIMPy | You only get the cause, no further information. |
| 03:37.52 | bloudermilk | Got it |
| 03:38.17 | bloudermilk | Thanks for the info. Found a helpful voip-info page |
| 03:41.45 | Sean-Der | In dialplan can I have multiple step 1's? |
| 03:43.59 | WIMPy | Not sure what you mean. |
| 03:50.23 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 03:55.17 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 04:07.28 | Sean-Der | WIMPy: It is just so confusing.... |
| 04:07.50 | Sean-Der | exten => +123,n,Verbose(1,Someone is calling extension 123.) |
| 04:08.00 | Sean-Der | If put inside the main code block works perfectly |
| 04:08.08 | Sean-Der | If I put it inside the include it doesn't work at all |
| 04:08.22 | Sean-Der | include => ext-did-0002-custom |
| 04:08.35 | Sean-Der | [ext-did-0002-custom] |
| 04:08.47 | Sean-Der | I don't see any issues with my syntax :( |
| 04:09.30 | WIMPy | Every Extension starts with priority 1. All following ones have to be numbered consecutive. |
| 04:09.51 | Sean-Der | Or just use n? |
| 04:10.25 | WIMPy | The include looks ok. |
| 04:10.34 | Sean-Der | Lets say I have two functions with the priority of 1 |
| 04:10.48 | Sean-Der | Are they executed procedurally by order |
| 04:10.49 | WIMPy | Yes, n will use the priority of the previous line +1. |
| 04:10.55 | Sean-Der | or at the same time? |
| 04:11.08 | WIMPy | The previous line in your file that is, not te previous line in that extension. |
| 04:11.17 | Sean-Der | ohh |
| 04:11.23 | WIMPy | In order. |
| 04:11.43 | Sean-Der | Also does an include literally just drop the code in like a function? |
| 04:12.06 | Sean-Der | For some reason my include is failing? |
| 04:12.31 | p3nguin | core show functions <---- this is a list of functions |
| 04:12.56 | p3nguin | I think maybe you're using wrong terminology here. |
| 04:13.05 | Sean-Der | p3nguin: I mean like a function in procedural programming. |
| 04:13.18 | Sean-Der | Does an include act like a function in C or PHP? |
| 04:13.30 | WIMPy | An include inserts one context in to another but extensions are always searched before includes. |
| 04:14.03 | Sean-Der | Crap. Is there a way to raise precedence of an include? |
| 04:14.10 | p3nguin | "include => other-context" is also different from "#Include other/file' |
| 04:14.16 | p3nguin | s/'/"/ |
| 04:14.18 | WIMPy | No like an include but without the possibility to conflict with existing definitions. |
| 04:14.44 | WIMPy | Use multiple includes. They are searched in listed order. |
| 04:15.07 | Sean-Der | For some reason none of the code in my include is being ran though! |
| 04:15.17 | p3nguin | You're probably doing it wrong. |
| 04:15.22 | p3nguin | Pastebin what you've done. |
| 04:15.35 | WIMPy | Do you have an extension in the context itself that matches? |
| 04:16.46 | Sean-Der | http://pastebin.com/g3zjtgXb |
| 04:17.19 | Sean-Der | Btw thanks for the help guys! |
| 04:17.59 | WIMPy | What's that ext ext ext? |
| 04:18.45 | p3nguin | Don't filter. If you want help, give us all the bits. |
| 04:20.00 | Sean-Der | I was just showing that a bunch of other functions happen after |
| 04:20.05 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 04:20.14 | coppice | Don't filter. If you want help, give us all your cash. |
| 04:20.29 | p3nguin | Those are not functions. I've already tried to explain that to you. |
| 04:20.32 | p3nguin | (2212.31) <p3nguin> core show functions <---- this is a list of functions |
| 04:20.38 | WIMPy | They may well be the reason it doesn't work. |
| 04:21.02 | Sean-Der | WIMPy: ? |
| 04:21.02 | p3nguin | Show all the pieces of the puzzle if you want me to tell you what it is. |
| 04:21.47 | WIMPy | The bits you left out. |
| 04:22.55 | Sean-Der | http://pastebin.com/Qn5F83PN |
| 04:23.27 | p3nguin | You're sending the call to ext-did-0002? |
| 04:24.11 | WIMPy | Extension +123 is already defined in ext-did-0002, so the include will never be searched. |
| 04:24.13 | p3nguin | Extension "+123" in that context is going to match before the one in the included context. |
| 04:25.02 | WIMPy | >>An include inserts one context in to another but extensions are always searched before includes. |
| 04:25.07 | Sean-Der | I can't edit the [ext-did-0002] context because it is auto generated |
| 04:25.28 | p3nguin | Generated by what? Asterisk doesn't generate anything. |
| 04:25.36 | Sean-Der | Freepbx in a flash |
| 04:25.42 | p3nguin | ~freepbx |
| 04:25.42 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 04:25.50 | Sean-Der | It isn't my choice |
| 04:25.52 | p3nguin | Wrong channel for help on that. |
| 04:26.01 | Sean-Der | Uggh ok thank you |
| 04:26.02 | p3nguin | We can fix it in asterisk. |
| 04:26.09 | p3nguin | We can't fix it via FreePBX. |
| 04:26.39 | Sean-Der | FreePBX is irritating me. My work should have just setup manually the first time. Would have been a hell of alot easier down the line |
| 04:26.53 | p3nguin | That'll teach them. |
| 04:27.10 | Sean-Der | p3nguin: ? |
| 04:27.23 | p3nguin | PiaF is certainly not a product of the quality I would use for anything of importance. |
| 04:27.37 | p3nguin | It's a low-grade mash-up of crap. |
| 04:27.50 | Sean-Der | Ouch :( |
| 04:28.42 | p3nguin | I wouldn't use FreePBX on top of a pure Asterisk install, either, but at least it isn't crap. |
| 04:29.18 | WIMPy | Depends on what you think it is. |
| 04:29.42 | p3nguin | I don't like it and I wouldn't use it, but it isn't crap. PiaF is crap. |
| 04:30.44 | Sean-Der | p3nguin: You are idling over there though :D |
| 04:30.53 | p3nguin | I sure am. |
| 04:31.23 | Sean-Der | I actually picked up the asterisk book yesterday so I am still learning my way around. I am an intern at an IT company so slaving away to earn my keep |
| 04:31.53 | p3nguin | The best thing you could do is stop now with the crapbx before you get in too deep. |
| 04:32.26 | p3nguin | Get out now, before it is too late, and get yourself a normal system with a normal asterisk. |
| 04:33.03 | p3nguin | Or if you have to have FreePBX, at least use AsteriskNOW with the FreePBX option. We can't help you with it here, but at least you can get help for it. |
| 04:33.27 | Sean-Der | I have an install on debian that I installed from the 1.8 trunk at home. |
| 04:33.36 | p3nguin | Perfect. |
| 04:33.47 | Sean-Der | But since I just started I only have the SIP extensions nothing else |
| 04:34.02 | p3nguin | Extensions aren't SIP. |
| 04:34.16 | p3nguin | Phones are not extensions. Phones are phones. Extensions make up the dial plan. |
| 04:35.27 | Sean-Der | I am still getting used to all this terminology |
| 04:35.53 | Sean-Der | Its hard coming from programming as alot of things are close but not exactly. Contexts `feel` like functions |
| 04:36.28 | WIMPy | No. Macros or Gosubs are. |
| 04:36.47 | p3nguin | But in Asterisk, functions are func_*.so |
| 04:37.02 | WIMPy | Contexts are more like directories of your finished programs (extensions). |
| 04:37.42 | WIMPy | But better don;t try to compare it to programming. |
| 04:38.13 | p3nguin | Unless you want to write all your dial plan in a C app and run it through the AGI application. |
| 04:38.55 | p3nguin | (or PHP) |
| 04:39.02 | WIMPy | That would be programming, but not related to the dialplan, except for being called from there. |
| 04:39.03 | p3nguin | You mentioned those two langs. |
| 04:41.24 | Sean-Der | I want to integrate into as much as possible though `the asterisk way` |
| 04:45.40 | p3nguin | If I knew exactly what it was you were trying to do, it would be beneficial to achieving results. |
| 04:52.37 | *** join/#asterisk greenwolf (~greenwolf@cpe-67-241-181-90.buffalo.res.rr.com) |
| 04:53.10 | Sean-Der | Sorry! |
| 04:55.39 | p3nguin | looks at ${CDR(duration)} |
| 04:57.12 | Sean-Der | I had CDR duration and obdc all ready to go |
| 04:57.28 | Sean-Der | If I can fix this context issue I am golden |
| 04:57.42 | p3nguin | It's working correctly, though. |
| 04:58.28 | Sean-Der | The verbose isn't displaying? |
| 04:59.27 | p3nguin | I never saw any evidence of that. |
| 04:59.53 | p3nguin | You've never shown me any call to extension "+123" yet. |
| 05:00.31 | SeRi | p3nguin: this is what I was talking about where it would fail http://pastebin.com/GMTAmcJV |
| 05:01.59 | WIMPy | SeRi: What does the 2nd line of your paste tell you? |
| 05:02.18 | *** join/#asterisk god-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 05:02.32 | SeRi | WIMPy: gmime is there. I am not sure why it claims is not. |
| 05:02.54 | WIMPy | That exact file? |
| 05:03.08 | WIMPy | It's probably some -dev package. |
| 05:03.10 | SeRi | one sec |
| 05:03.27 | WIMPy | And if you installed it after trying, did you rerun configure? |
| 05:04.38 | SeRi | It has all ways been there |
| 05:04.43 | SeRi | gmime 2.4 |
| 05:05.23 | WIMPy | locate gmime/gmime.h |
| 05:06.24 | *** join/#asterisk hipitihop (~denis@122-149-140-217.static.dsl.dodo.com.au) |
| 05:07.15 | *** join/#asterisk ppc (ppc@cloudy.cmang.org) |
| 05:07.37 | hipitihop | getting the following trace on incoming call sip , can someone point me at solution: WARNING[17267]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'SIP' |
| 05:07.42 | SeRi | /usr/include/gmime-2.4/gmime/gmime.h |
| 05:07.47 | SeRi | <PROTECTED> |
| 05:08.27 | WIMPy | Ok, that's looking good. |
| 05:08.47 | WIMPy | hipitihop: Do you have chan_sip loaded? |
| 05:10.01 | hipitihop | WIMPy, still pretty new to *, can you tell me how to tell, something in console ? |
| 05:10.11 | WIMPy | SeRi: Look at your configure.log for clues. |
| 05:10.25 | SeRi | WIMPy: Thanks I am on it |
| 05:10.28 | WIMPy | hipitihop: 'module show like sip' |
| 05:11.05 | WIMPy | seri: Or you just try to add an -I/usr/include/gmime-2.4 |
| 05:11.14 | hipitihop | WIMPy, chan_sip.so & app_adsprog.so loaded |
| 05:12.32 | WIMPy | hipitihop: They the peer you're trying to call is either non existant or unreachable. |
| 05:13.06 | WIMPy | You can check, what you've got with 'sip show peers'. |
| 05:14.02 | hipitihop | Wimpy, so time to check my extnesions.conf ? |
| 05:15.23 | WIMPy | Or your sip.conf. |
| 05:16.01 | hipitihop | so it's just a soft warning that not all peers I'm trying to call are currently registered ? |
| 05:16.27 | WIMPy | That is a possible cause, yes. |
| 05:17.29 | hipitihop | ah makes, sense, since I have a couple of soft phones, android and iphone currently not registered, and incoming calls are setup to everithing |
| 05:17.46 | hipitihop | WIMPy, thanks for your help |
| 05:37.30 | *** join/#asterisk s[X] (~mark@ppp118-208-17-182.lns20.bne1.internode.on.net) |
| 05:55.36 | dijib | so cdr-stats |
| 05:56.01 | dijib | i need to keep my cdr in sql? |
| 06:00.04 | p3nguin | That's right. Lucky for you, that is very easy to do. |
| 06:10.13 | *** join/#asterisk s[X] (~mark@ppp118-208-17-182.lns20.bne1.internode.on.net) |
| 06:35.47 | *** join/#asterisk timgws[coding] (~tim@124-171-19-230.dyn.iinet.net.au) |
| 06:35.56 | *** join/#asterisk timgws (~inspircd@sharesource.org) |
| 06:40.56 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:45.24 | dijib | lucky for you i still havn;t read the ~book |
| 06:45.30 | dijib | ~book |
| 06:45.30 | 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 |
| 06:57.57 | [TK]D-Fender | ~osmosis |
| 06:57.58 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 07:01.00 | [TK]D-Fender | And on that note.. checkout time... |
| 07:02.48 | dijib | o man thats good |
| 07:03.47 | WIMPy | ~reverse osmosis |
| 07:04.09 | WIMPy | Hmm |
| 07:05.54 | dijib | so i am still perplexed as to how to configure asterisk to use mysql as its cdr backend |
| 07:08.43 | p3nguin | Don't. |
| 07:08.47 | p3nguin | Use pgsql. |
| 07:09.20 | dijib | y? |
| 07:09.32 | p3nguin | MySQL is kind of shitty. |
| 07:10.15 | p3nguin | Just configure your cdr_pgsql.conf, set up your db according to the documentation, then log away. |
| 07:10.17 | dijib | it would match my shit shaper |
| 07:11.24 | dijib | asterisk-pgsql? |
| 07:14.25 | ChannelZ | database wars! Almost as fun as distro wars! |
| 07:21.58 | *** join/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 07:24.37 | *** join/#asterisk Sean-Der (~Sean-Der@NW1-DSL-74-215-64-154.fuse.net) |
| 07:25.34 | Sean-Der | p3nguin: I figured out how to implement what I was looking for. How does the CDR function work? |
| 07:27.49 | p3nguin | core show function CDR |
| 07:28.35 | WIMPy | A function! |
| 07:29.08 | p3nguin | Finally! |
| 07:29.50 | Sean-Der | Woohoo Wohoo! |
| 07:30.56 | Sean-Der | The only issue is that when I call CDR won't the duration be 0? How can do loop until the phone is disconnected? |
| 07:31.09 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 07:31.40 | WIMPy | CDRs are written when the call has ended. |
| 07:31.46 | p3nguin | Like I said several hours ago, you do it in the h extension. |
| 07:31.55 | p3nguin | extension h runs when the call ends. |
| 07:32.11 | p3nguin | Or... |
| 07:32.19 | p3nguin | Just parse the CDR files. |
| 07:32.32 | Sean-Der | Sorry didn't mean to ignore that! Xchat ate my scrollback :( |
| 07:32.33 | p3nguin | CDR can log directly to your database. |
| 07:32.55 | p3nguin | Which database are you using? |
| 07:32.59 | Sean-Der | I am just debugging now. I will be writing a macro and using odbc tommorow |
| 07:33.06 | Sean-Der | Its 2:30 AM here so I am pretty tired |
| 07:33.07 | p3nguin | Which database are you using? |
| 07:33.24 | Sean-Der | I will be just making a new table |
| 07:33.29 | p3nguin | Which database are you using? |
| 07:33.35 | p3nguin | pgsql or mysql |
| 07:33.44 | Sean-Der | Ahh you mean engine |
| 07:33.46 | Sean-Der | MySQL |
| 07:33.55 | p3nguin | No, I mean DATABASE |
| 07:34.25 | p3nguin | Look at the cdr_mysql.conf.sample |
| 07:34.55 | *** join/#asterisk coppice (~chatzilla@globbits.tripleone.co.uk) |
| 07:34.59 | p3nguin | CDR will write directly to the database. |
| 07:34.59 | Sean-Der | Okee dokee. Hopefully Freepbx doesn't override any of this now |
| 07:35.36 | ChannelZ | chokes on his drink |
| 07:35.43 | p3nguin | It will have your billsec and duration for you. |
| 07:35.44 | WIMPy | The hope always dies last. |
| 07:37.03 | p3nguin | Since you haven't even started with CDR yet, you could go one step further and use CEL instead of CDR. |
| 07:37.08 | p3nguin | ~cel |
| 07:37.08 | infobot | CEL is Channel Event Logging, or http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html#Monitoring_id246970 |
| 07:37.19 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 07:37.56 | Sean-Der | exten => h,n,Verbose(1, The call lasted ${CDR(duration)}) |
| 07:38.23 | Sean-Der | I have this saved in my scratch pad. Is this any where near what you were referencing? |
| 07:38.46 | p3nguin | Yes, it is almost exactly what I had in mind. |
| 07:39.00 | p3nguin | But remember, all extensions will start with priority 1. |
| 07:39.28 | p3nguin | Also note there will be a difference in 'duration' and 'billsec'. |
| 07:39.58 | Sean-Der | Also working with channels sounds like a much better idea |
| 07:40.34 | Sean-Der | CHAN_END looks like what I am exactly looking for |
| 07:40.50 | p3nguin | duration is the seconds of the entire call, where billsec is billable seconds after the call has been answered. |
| 07:41.01 | Sean-Der | ok changing now |
| 07:42.09 | Sean-Der | Hmm for some reason my billsec wasn't echoed |
| 07:42.24 | p3nguin | Did you have an answer? |
| 07:42.37 | p3nguin | I don't remember seeing one in your extension that you showed me earlier. |
| 07:42.54 | Sean-Der | No I do not. |
| 07:43.04 | p3nguin | Do a Playback() of some file or an Answer(10) or something similar. |
| 07:43.11 | p3nguin | something to answer the channel and wait a few seconds. |
| 07:43.22 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:43.25 | p3nguin | actually Answer(3000) for three seconds |
| 07:43.34 | p3nguin | 10 would be immeasurable. |
| 07:43.47 | Sean-Der | http://pastebin.com/SEc3DKsE |
| 07:43.59 | p3nguin | fail |
| 07:44.03 | p3nguin | (0138.59) <p3nguin> But remember, all extensions will start with priority 1. |
| 07:44.15 | p3nguin | You have no priority 1 in extension h. |
| 07:44.54 | Sean-Der | Sorry I am not understanding the concept then? I thought that was following 123? |
| 07:44.59 | p3nguin | http://pastebin.com/Ln7k3Fz4 |
| 07:45.15 | p3nguin | Every extension will start with priority 1. |
| 07:45.39 | p3nguin | And you have extension +123, not extension 123. |
| 07:45.49 | p3nguin | How do you dial the + from your keypad? |
| 07:45.59 | p3nguin | Oh, softphone, probably. |
| 07:46.12 | Sean-Der | its just a testing line for now |
| 07:46.27 | Sean-Der | Can you re explain the concept of h for me? |
| 07:46.31 | p3nguin | I can't dial a + |
| 07:46.41 | Sean-Der | I am sorry that I didn't get it the first time? |
| 07:46.56 | p3nguin | Extension 'h' is what we cann the "hangup extension." It runs when a call hangs up. |
| 07:47.08 | p3nguin | s/cann/call/ |
| 07:47.41 | Sean-Der | Ahh ok! How come I have to use the Answer and Hangup also... |
| 07:47.45 | p3nguin | When you use an extension, it eventually ends. When it ends and the call dies, extension h runs. |
| 07:47.58 | p3nguin | billsec does not start until an answer. |
| 07:48.02 | p3nguin | No answer, no billsec. |
| 07:48.17 | Sean-Der | But then won't I be billing for unanswered calls? |
| 07:48.19 | p3nguin | I'm giving you a basic extension to test. |
| 07:48.35 | p3nguin | You should never be billed for a call which has not been answered. |
| 07:48.53 | Sean-Der | Ahh ok that is just for our little test |
| 07:50.06 | p3nguin | http://pastebin.com/c5xSbrg8 |
| 07:50.39 | p3nguin | That might help. |
| 07:51.30 | Sean-Der | Alot more verbose thanks! |
| 07:51.38 | p3nguin | ~alot |
| 07:51.38 | infobot | i guess alot is raping the English language, or http://hyperboleandahalf.blogspot.com/2010/04/alot-is-better-than-you-at-everything.html |
| 07:52.07 | Sean-Der | Oh my :| |
| 07:52.13 | Sean-Der | Thank you very much for your help today! |
| 07:52.16 | Sean-Der | I learned so much |
| 07:52.42 | p3nguin | In this dial plan, you should be able to see how the duration is 7 seconds (4 seconds before answer, and 3 seconds after answer). |
| 07:52.58 | p3nguin | But billsec is only the 3 after the answer. |
| 07:53.25 | p3nguin | It should have printed all that when the call hung up. |
| 07:53.58 | Sean-Der | The call lasted 7 total and 3 after answer |
| 07:54.00 | Sean-Der | perfect! |
| 07:54.35 | p3nguin | You can play around with all of the fields shown in "core show function CDR" to see if any others are useful for you. |
| 07:54.49 | Sean-Der | I think I should implement by channel though. Is that what you were showing me? It seems it would be more `future` proof |
| 07:55.08 | p3nguin | The actual CDRs have the channel information. |
| 07:55.31 | Sean-Der | So if I transfer the call around when it ends it will still have the duration? |
| 07:55.36 | p3nguin | I'm not talking about the crap we're echoing from the dial plan, I'm talking about the actual CDR that will be written to file or db. |
| 07:56.10 | p3nguin | CDR is written when the call ends. |
| 07:56.16 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 07:56.27 | IsUp | morning all |
| 07:57.07 | Sean-Der | Ok well I am going to actually start on that now just to get a head start |
| 07:57.41 | p3nguin | Look at CEL, too. You may want to use it also or instead. |
| 07:58.03 | p3nguin | I started the conversion to CEL, but I haven't actually made it go yet. |
| 07:58.04 | Sean-Der | Whats your opinion? |
| 07:58.11 | p3nguin | CEL is the new hotness. |
| 07:58.28 | p3nguin | CDR is actually becoming the "old way." |
| 07:59.46 | Sean-Der | cdr_mysql.conf is already defined by freepbx. |
| 07:59.55 | Sean-Der | So I will have to create a new table in my existing schema |
| 07:59.59 | p3nguin | Sucks to be using FreePBX. |
| 08:00.32 | p3nguin | It really ruins a guy's day when you go to edit the confs yourself. |
| 08:01.26 | Sean-Der | It really does. So it looks like I will have to conform to this table schema http://pastebin.com/gzUjMWXS |
| 08:01.38 | Sean-Der | This is really worth $10.50 an hour :| |
| 08:02.23 | p3nguin | That looks like a good table to me. |
| 08:02.40 | p3nguin | If it is already there, you can use it. |
| 08:02.42 | Sean-Der | Okee doke now for some insert magic! |
| 08:03.05 | Sean-Der | I don't know what software Query's from it though. Hope I don't break that interface |
| 08:03.33 | p3nguin | Expect anything auto-created by FreePBX to break when you do anything to it manually. |
| 08:04.22 | Sean-Der | What an optimist.... :D |
| 08:04.46 | Sean-Der | So I will replace my hangup with a CDR insert now. |
| 08:04.59 | p3nguin | Uh... |
| 08:05.00 | p3nguin | What? |
| 08:05.26 | p3nguin | You configure CDR and it is written FOR YOU. It's not something you force. |
| 08:05.42 | p3nguin | cdr.conf |
| 08:05.46 | WIMPy | ForkCDR :-) |
| 08:06.30 | p3nguin | cdr.conf, cdr_mysql.conf, cdr_adaptive_odbc.conf, cdr_custom.conf |
| 08:06.50 | p3nguin | cdr_odbc.conf |
| 08:07.02 | p3nguin | All sorts of CDR config files. |
| 08:08.00 | Sean-Der | It looks like CDR is set to /var/log/asterisk/Master.csv |
| 08:08.03 | p3nguin | Then you've got the cel_*.conf files |
| 08:08.37 | IsUp | hey p3nguin |
| 08:08.42 | p3nguin | hi |
| 08:09.42 | Sean-Der | So I just make a simple call on hangup and thats it? |
| 08:09.56 | p3nguin | no |
| 08:10.02 | p3nguin | You do nothing. |
| 08:10.10 | p3nguin | Configure the files. Make calls. |
| 08:10.16 | p3nguin | CDR is written automatically. |
| 08:11.01 | p3nguin | Just look at /var/log/asterisk/Master.csv |
| 08:11.10 | p3nguin | tailf /var/log/asterisk/Master.csv |
| 08:11.12 | p3nguin | make calls |
| 08:11.16 | p3nguin | watch. |
| 08:11.56 | p3nguin | The same thing will happen when you configure it to write to the database. |
| 08:12.26 | Sean-Der | The only issue is that the dest is broken in CDR. So now I can use the CDR function to set a different dest |
| 08:13.33 | p3nguin | I don't see anything wrong with that. Just Set(CDR(dst)=whatever-you-want-it-to-be) in the extension. |
| 08:14.06 | p3nguin | It really should be right, though. What shows up as the destination, and what did you expect to show up? |
| 08:15.16 | Sean-Der | dest = 's' |
| 08:15.32 | p3nguin | So you've called extension s. |
| 08:15.44 | p3nguin | Call a different extension and the destination will be different. |
| 08:16.00 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 08:18.41 | Sean-Der | My boss wants me to make it so when someone calls the 1800 number it says they called the 1800 directly by the inbound and then record the entire call duration |
| 08:18.52 | Sean-Der | So that is the end goal |
| 08:19.06 | p3nguin | Configure things accordingly, and that will happen. |
| 08:19.14 | kaldemar | dst is a read-only field. you can't set it. |
| 08:19.22 | p3nguin | gasp |
| 08:19.36 | p3nguin | That kind of makes things more difficult. |
| 08:20.15 | p3nguin | When I receive a call, it doesn't say it went to extension s because it doesn't go to extension s. Configure your extensions to be correct and the dst should be what you expect. |
| 08:21.01 | Sean-Der | I wish I could just record the length of the channel and then insert it into the table of my choice. |
| 08:21.12 | Sean-Der | p3nguin: I can't change anything that my boss has created |
| 08:21.24 | Sean-Der | even if I am right I will get fired if I challenge him |
| 08:21.25 | p3nguin | Tell him to fix it, then. |
| 08:21.40 | p3nguin | Explain to him that it is wrong, and explain why it needs to be fixed, and then get it fixed. |
| 08:21.50 | Sean-Der | I need my job, its just my job to figure out how to bubble gum everything. |
| 08:22.21 | Sean-Der | Its 3:30 AM I need sleep I have to get up at a decent time tommorow |
| 08:22.40 | kaldemar | to find a new job? |
| 08:23.34 | p3nguin | Does the dst change if you use Goto() or transfer from a phone? |
| 08:24.06 | Sean-Der | p3nguin: I assume it does? I don't know I can check |
| 08:24.21 | Sean-Der | I am going to mess around with CEL tommorow. I need some sleep right now |
| 08:24.25 | p3nguin | I'm trying to think of how that dst would be s if the extension wasn't really s to begin with. |
| 08:24.39 | Sean-Der | Thank you for all your help so far! |
| 08:24.54 | Sean-Der | Its an inbound that goes to a time group (if 1 AM)else |
| 08:25.03 | Sean-Der | that then goes to a ivr |
| 08:25.11 | Sean-Der | which you can then select an extension |
| 08:25.18 | p3nguin | And that's probably where it turns into s. |
| 08:25.36 | p3nguin | Most people use extension s for ivr. |
| 08:26.22 | Sean-Der | p3nguin: The issue is that we have a lot of ivr's on the server. And querying the cdrdb will return a million s's |
| 08:27.12 | Sean-Der | So if I am able to get all these inbounds to insert the billsec with the dst as what inbound route it came through I will be set |
| 08:27.16 | p3nguin | If what is happening is what I think might be happening, I understand. |
| 08:27.42 | Sean-Der | Sorry I can't explain better. I am still learning |
| 08:28.19 | p3nguin | You've at least got some ideas and can do some testing. |
| 08:28.45 | Sean-Der | That is what being a code monkey develops. Desperate ideas for impossible situations |
| 08:29.24 | Sean-Der | My days of debugging ugly Javascript and PHP ripped off random websites by 'consultants' is finally paying off |
| 08:30.39 | Sean-Der | I could store in the userfiled of DST the information I want |
| 08:30.47 | Sean-Der | CDR allows that be writeable |
| 08:30.56 | Sean-Der | and then my report software will run off that |
| 08:32.33 | Sean-Der | Thanks again night! |
| 08:38.51 | *** join/#asterisk s[X] (~mark@ppp118-208-17-182.lns20.bne1.internode.on.net) |
| 08:39.06 | IsUp | http://www.youtube.com/watch?v=kfchvCyHmsc |
| 08:57.31 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:09.21 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:11.49 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:26.31 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 09:42.20 | *** join/#asterisk bloudermilk (~bloudermi@cpe-76-168-253-208.socal.res.rr.com) |
| 09:43.13 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 09:44.23 | verywiseman | how can i make connection btw 2 server , one of them have static ip , and other have not? |
| 09:47.12 | kaldemar | define connection |
| 09:48.59 | verywiseman | kaldemar, how? |
| 09:50.31 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:50.49 | kaldemar | i meant what do you mean by "connection"? |
| 09:53.50 | verywiseman | kaldemar, to route calls between them |
| 09:54.06 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:54.27 | IsUp | verywiseman: use IAX maybe? |
| 09:54.53 | verywiseman | IsUp, i know |
| 09:55.17 | verywiseman | my question is , how can i do that if one server have static real ip , and other not have |
| 09:55.29 | kaldemar | verywiseman: what protocol/technology are you planning to use! |
| 09:55.42 | verywiseman | iax |
| 09:56.38 | IsUp | verywiseman: i think you can use dynamic dns service as my opinion |
| 09:56.43 | kaldemar | make the dynamic one register to the static one. |
| 09:56.56 | kaldemar | no dns needed. |
| 09:57.15 | IsUp | yes, register is better solution :) |
| 09:57.37 | verywiseman | kaldemar, thanks |
| 10:01.00 | verywiseman | kaldemar, if there is 3rd server has not real ip , can it register to the server which have real ip , and make call to the server which has not real ip? |
| 10:04.54 | kaldemar | any server can do anything when properly set up |
| 10:05.59 | kaldemar | use registrations to let the other ends know where the dynamic ones are. that's what registrations are for. |
| 10:07.52 | verywiseman | kaldemar, look, if i have for example 4 server A,B,C and D , server A only has real ip and other has not. so B,C and D will register on A , and the can talk each other , is it true? |
| 10:08.03 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 10:10.46 | kaldemar | verywiseman: through A, yes. directly to each other, no. unless you make them know each other. |
| 10:14.33 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 10:16.48 | verywiseman | kaldemar, would you check this http://fpaste.org/czwE/ please? |
| 10:24.03 | kaldemar | make B, C and D register to A. |
| 10:28.34 | verywiseman | kaldemar, i did that already , did not you see it in http://fpaste.org/czwE/? |
| 10:31.06 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 10:31.55 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 10:32.17 | kaldemar | verywiseman: no you did not. only registrations are from A to B and from C to B. |
| 10:33.01 | *** join/#asterisk kriegerod (~krieger@79.135.222.22) |
| 10:37.00 | *** join/#asterisk Superstar (~Superstar@ca13.v6.us.gnics.net) |
| 10:38.13 | verywiseman | kaldemar, i am sorry for this mistake , pls check this http://fpaste.org/YK3a/ |
| 10:39.30 | Superstar | Does Asterisk set the source IP on outbound traffic to one we set it to bind to or does it follow linux route? |
| 10:41.21 | kaldemar | Superstar: it is set by the OS. |
| 10:41.37 | Superstar | Great |
| 10:49.51 | verywiseman | kaldemar, i am sorry for this mistake , pls check this http://fpaste.org/YK3a/ |
| 10:52.16 | kaldemar | is it not working? |
| 10:55.45 | verywiseman | kaldemar, i gust ask you if that true ? |
| 10:56.00 | verywiseman | i will test it |
| 10:59.21 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 10:59.21 | *** mode/#asterisk [+o file] by ChanServ |
| 10:59.58 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 11:16.05 | *** join/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 11:48.43 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 11:56.01 | verywiseman | kaldemar, Now B,C,D can talk each other ,is it true? |
| 12:15.01 | *** join/#asterisk hehol (~Adium@ip-178-202-243-147.unitymediagroup.de) |
| 12:33.16 | *** join/#asterisk hajekd (~hajekd@2a00:1268:15:1800:2507:a2a5:4e1e:eb0b) |
| 12:33.22 | kaldemar | verywiseman: only through A and only if your dialplan allows them to. |
| 12:35.56 | verywiseman | kaldemar, ok , if in extensions.conf on C server i put this : "exten => 1234,1,Dial(IAX2/serverD/${EXTEN},30,r)", how can C server locate server D? |
| 12:38.07 | *** join/#asterisk TimeRider (~steve@92.40.253.169.threembb.co.uk) |
| 12:42.03 | *** join/#asterisk longshitian (~longshiti@c83-250-126-197.bredband.comhem.se) |
| 12:52.45 | *** part/#asterisk dapsaille (~Dapsaille@95.210.161.54) |
| 12:59.40 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 13:03.09 | *** part/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 13:21.57 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 13:36.26 | *** join/#asterisk fofware (~fabian@host26.186-108-159.telecom.net.ar) |
| 13:45.26 | *** join/#asterisk longshitian (~longshiti@83.250.126.197) |
| 14:20.30 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 14:21.10 | patrickod | is it possible when using asterisk realtime for sip peers and sip users to have register statements kept in the DB ? |
| 14:21.16 | patrickod | or do they have to to be in sip.conf |
| 14:21.32 | *** join/#asterisk devil_evoxxx (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 14:21.38 | devil_evoxxx | hi al :) |
| 14:21.43 | devil_evoxxx | hi all :) |
| 14:25.24 | devil_evoxxx | someone here use OpenSIPS / Kamailio with asterisk as pstn-gw? |
| 14:33.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:56.11 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 15:07.33 | *** join/#asterisk puzzled (~patrick@53533D66.cm-6-4a.dynamic.ziggo.nl) |
| 15:07.58 | *** join/#asterisk psharmor (~quassel@97-118-211-52.hlrn.qwest.net) |
| 15:11.03 | *** join/#asterisk jkroon (~jkroon@dsl-241-227-236.telkomadsl.co.za) |
| 15:23.32 | kaldemar | verywiseman: "THROUGH A". C must dial A and A can dial D. |
| 15:33.42 | *** join/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 15:34.20 | *** part/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 15:34.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 15:53.13 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 15:53.28 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 15:55.09 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 15:56.18 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 16:09.40 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 16:27.15 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 16:29.43 | patrickod | is there a channel variable that contains the SIP username that's making the call ? |
| 16:29.52 | patrickod | or does this have to be regex'd from the channel variable? |
| 16:35.33 | ectospasm | patrickod: there's the SIPPEER function, that will contain what you want (I think, my Asterisk system is down for right now) |
| 16:35.48 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 16:36.06 | patrickod | ectospasm: thanks I'll try that now and see what it contains |
| 16:36.26 | ectospasm | patrickod: core show function SIPPEER, or core show functions |
| 16:37.05 | ectospasm | it'll be something like 1034-00000a3d or something |
| 16:40.30 | *** join/#asterisk davlefou (~david@41.225.244.208) |
| 16:50.33 | leifmadsen | patrickod: and then use CUT() to strip off the unique identifier since you probably don't need it |
| 16:50.53 | patrickod | leifmadsen: I just found CUT in the docs, that's what I'm using now |
| 16:51.01 | leifmadsen | yep, there you go |
| 16:51.21 | leifmadsen | Set(thisPeer=${CUT(SIPPEER,-,1)}) |
| 16:52.00 | ectospasm | well, Set(thisPeer=${CUT(${SIPPEER},-,1)}) |
| 16:52.21 | ectospasm | or am I wrong? |
| 16:58.18 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 17:00.52 | [TK]D-Fender | You are referencing as a variable, nt as a function there |
| 17:02.52 | ectospasm | I thought you still had to dereference the function the same way as a variable. |
| 17:03.13 | ectospasm | ...if you were reading the value of the function. Same as for the CUT function there. |
| 17:03.38 | ectospasm | ...I don't use CUT much, so maybe it doesn't need SIPPEER to be dereferenced. |
| 17:03.54 | WIMPy | CUT is special. |
| 17:04.11 | WIMPy | It doesn't take a value, but the name of a variable. |
| 17:04.20 | WIMPy | See 'core show function CUT'. |
| 17:05.57 | *** join/#asterisk AmirBehzad (~behzad@86.57.4.5) |
| 17:09.56 | patrickod | when using realtime for sippeers and voicemail is it possible to use a mysql view for the voicemail table to give automatic voicemail functionality to every sip users ? |
| 17:14.30 | *** part/#asterisk d00gster (~dt@77.30.126.165) |
| 17:14.53 | *** join/#asterisk d00gster (~dt@77.30.126.165) |
| 17:25.20 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 17:27.09 | patrickod | is there a known issue where macros don't execute the h extension ? |
| 17:48.52 | patrickod | I can't get asterisk to perform any actions after VoiceMail in the dialplan even though the h extension is set |
| 18:01.31 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:02.10 | leifmadsen | ectospasm: you are wrong -- you'd be using the value of the SIPPEER variable as the function/variable name to pass |
| 18:02.20 | leifmadsen | you give CUT() the name of the function or variable without ${ } |
| 18:04.34 | patrickod | leifmadsen: do you know if older versions of Asterisk (such as that in Debian's repos) have problems with the Voicemail app? I keep getting errors that it exited non-zero and any instuctions placed after it in a macro fail to execute |
| 18:04.53 | leifmadsen | no idea |
| 18:05.05 | leifmadsen | I don't use older versions, especially those shipped with debian |
| 18:05.17 | leifmadsen | which I keep seeing as version 1.4.21 or something redonkulous |
| 18:08.20 | patrickod | does the voicemail app have a certain exetension that it uses if the user hangs up? |
| 18:08.50 | patrickod | I'm running Asterisk 1.6.2.9-2+squeeze3 |
| 18:28.07 | *** join/#asterisk TimeRider (~steve@92.40.247.208.threembb.co.uk) |
| 18:28.16 | SeRi | leifmadsen: you avail? |
| 18:35.47 | kaldemar | patrickod: you most likely have the h exten in the wrong context. |
| 18:36.39 | patrickod | kaldemar: it's in the macro itself, should it be outside ? |
| 18:40.58 | kaldemar | patrickod: in the context that the macro is called from. |
| 18:41.23 | patrickod | kaldemar: ok. and I presume if this macro is being called from a macro itself then it has to be in that context ? |
| 18:43.37 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 18:46.22 | kaldemar | patrickod: in the so called current context that executes the first macro |
| 19:06.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:20.35 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il) |
| 19:21.14 | leifmadsen | SeRi: kinda |
| 19:22.30 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 19:31.09 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 19:34.37 | *** join/#asterisk nix8n82 (~hmg@71-32-140-167.chyn.qwest.net) |
| 19:40.12 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:59.37 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 20:00.25 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:04.07 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 20:08.33 | *** part/#asterisk patrickod (~Patrick@lysander.patrickodoherty.com) |
| 20:10.46 | SeRi | leifmadsen: have you ever build astlinux before? |
| 21:32.07 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 21:55.26 | SeRi | bday time.... |
| 22:06.11 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 22:10.56 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 22:30.00 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 22:45.49 | *** join/#asterisk carloimperia (~carloimpe@109.112.37.198) |
| 22:50.13 | *** join/#asterisk helen_ (Vena@unaffiliated/cmi-dos) |
| 22:50.16 | helen_ | Hello! |
| 22:50.28 | ChannelZ | O hell! |
| 22:53.12 | helen_ | If I dial voice mail extension 4242 once then I get through |
| 22:53.22 | helen_ | But if I dial it twice... |
| 22:53.28 | helen_ | Then I get the following |
| 22:53.30 | helen_ | http://pastebin.com/0HYsjNmw |
| 22:53.32 | helen_ | Why? |
| 22:53.34 | helen_ | Also... |
| 22:53.45 | helen_ | I can't phone other users on the network. |
| 22:54.00 | helen_ | I run http://www.optixlayer.com/ |
| 22:54.13 | helen_ | and this problem needs to be fixed urgently! |
| 22:57.31 | ChannelZ | Did you build asterisk yourself? |
| 22:57.38 | helen_ | No |
| 22:57.49 | helen_ | It's a Ubuntu package. |
| 22:58.17 | ChannelZ | hmm. Well you might need to, and/or it might just be a barf with whatever VM they are running. |
| 22:58.37 | helen_ | hmm |
| 22:58.49 | helen_ | ChannelZ: It needs to be fixed anyway. |
| 22:58.56 | ChannelZ | The timer_fd is an alternate timing source for doing audio mixing, though I'm not sure why it needs it for voicemail |
| 22:59.13 | helen_ | So it's recommended that I try building it? |
| 23:00.08 | ChannelZ | actually.. hang on |
| 23:01.08 | helen_ | k |
| 23:01.57 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 23:02.04 | ChannelZ | WTF |
| 23:02.04 | helen_ | Wheee! |
| 23:02.21 | helen_ | ChannelZ: Your connection got reset. :) |
| 23:02.30 | ChannelZ | yeah after you messaged me |
| 23:02.44 | ChannelZ | which I don't know what the hell you're talking about by the way |
| 23:03.05 | helen_ | Ok |
| 23:03.48 | ChannelZ | you can possibly try using pthread timing instead of timerfd |
| 23:04.28 | helen_ | ChannelZ: pthread with a p at the beggining? |
| 23:05.43 | ChannelZ | yeah. I'm not sure how to be honest as I've never had to change it... possibly you 'noload' res_timing_timerfd in modules.conf and it will either use pthread its self or you might have to specify it |
| 23:06.36 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 23:06.38 | ChannelZ | yeah that worked here |
| 23:07.12 | ChannelZ | what kind of channel were you using? |
| 23:08.07 | helen_ | ChannelZ: ? |
| 23:08.17 | helen_ | Ok i'm very confused. |
| 23:08.18 | ChannelZ | SIP, IAX... |
| 23:08.23 | helen_ | SIP |
| 23:08.28 | ChannelZ | hmm |
| 23:08.45 | ChannelZ | what are you confused about |
| 23:08.54 | helen_ | 18:05 < ChannelZ> yeah. I'm not sure how to be honest as I've never had to change it... possibly you 'noload' res_timing_timerfd in modules.conf and it will either use pthread its self or you might have to specify it |
| 23:09.04 | helen_ | I'm very new to this side of asterisk. |
| 23:09.18 | ChannelZ | edit /etc/asterisk/modules.conf and put "noload => res_timing_timerfd.so" in it, like at the bottom |
| 23:09.18 | helen_ | I only know the sip.conf and the extensions.conf |
| 23:09.30 | ChannelZ | then stop asterisk and start it again (reload won't work) |
| 23:09.37 | helen_ | ok |
| 23:09.46 | ChannelZ | then on the console do "module show like pthread" and see if its use count is 1 |
| 23:12.27 | helen_ | Omg I hate the command line. |
| 23:14.40 | helen_ | xpot: Use count 0 |
| 23:14.44 | helen_ | ChannelZ: |
| 23:14.54 | helen_ | xpot: Sorry for the mishighlight |
| 23:15.22 | ChannelZ | what about "module show like timerfd" |
| 23:16.11 | helen_ | ChannelZ: Use count 1 |
| 23:16.55 | ChannelZ | and you stopped/restarted Asterisk? |
| 23:17.22 | helen_ | Yes |
| 23:17.29 | helen_ | kill pid |
| 23:17.32 | helen_ | is what I did |
| 23:17.37 | ChannelZ | you might need to move the noload up higher in modules.conf, maybe something else loaded first that loaded it like meetme or something |
| 23:17.38 | helen_ | because it wouldn't restart |
| 23:17.49 | helen_ | k |
| 23:18.10 | ChannelZ | in the console you can do "core stop gracefully" and then run it again once it's quit |
| 23:18.40 | helen_ | k |
| 23:18.52 | ChannelZ | either that or maybe pthreads ain't gonna work on that thing either |
| 23:20.05 | ChannelZ | although I guess I should have asked awhile ago, what version of asterisk is this? |
| 23:20.21 | ChannelZ | guess it has to be at least 1.6 |
| 23:21.26 | helen_ | Asterisk 1.6.2.5-0ubuntu1.4 |
| 23:22.27 | helen_ | Ok i'm getting Use count 1 now. :) |
| 23:26.19 | ChannelZ | nature calls |
| 23:27.21 | *** join/#asterisk carloimperia (~carloimpe@109.112.37.198) |
| 23:27.50 | helen_ | ChannelZ: Ok voice mail is working now... |
| 23:28.00 | helen_ | there is another problem though. |
| 23:28.07 | helen_ | I can't make calls to other users. |
| 23:28.25 | helen_ | and yes they are on the same network. |
| 23:36.00 | helen_ | ChannelZ: Another thing... |
| 23:36.05 | helen_ | I can call myself. |
| 23:46.27 | *** join/#asterisk srd (hbunting@ec2-50-18-185-63.us-west-1.compute.amazonaws.com) |
| 23:46.54 | srd | how does 1.8 differ from 10.0? |
| 23:48.17 | carrar | it works |
| 23:48.29 | srd | with google voice for incoming calls? |
| 23:48.43 | *** join/#asterisk Sean-Der (~Sean-Der@NW1-DSL-74-215-64-154.fuse.net) |
| 23:49.00 | ChannelZ | helen_ well without seeing your dialplan or anything I can't speculate as to why, or what "can't make calls" even means |
| 23:49.22 | ChannelZ | The console reveals all |
| 23:49.43 | helen_ | Ok i'll pastebin my sip.conf and extensions.conf |
| 23:51.26 | ChannelZ | verbose console output is probably a better place to start |
| 23:51.29 | ChannelZ | core set verbose 3 |
| 23:55.36 | helen_ | http://pastebin.com/3ZTV40GL |
| 23:56.56 | Sean-Der | I am trying to enable CEL. I uncommented enable=yes in cel.conf |
| 23:57.37 | Sean-Der | Since I want it to go to MySQL I am going to uncomment cel_odbc.conf the [first] section |
| 23:57.55 | Sean-Der | Do I have to manually create the table it is going to? If so what is the schema |
| 23:58.16 | carrar | People still use Oracle err I mean MySQL |
| 23:58.19 | p3nguin | There should be a template somewhere that you can use to create your table. |
| 23:58.57 | Sean-Der | http://asteriskfaqs.org/tag/cel |
| 23:59.06 | Sean-Der | This looks promising like the third down? |