| 00:00.19 | DelphiWorld | evening |
| 00:09.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 13:07.47 | *** join/#asterisk infobot (ibot@rikers.org) |
| 13:07.47 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 13:10.33 | [TK]D-Fender | IMS77: * is not a proxy. It takes the media and passes it to the other very separate channel which gets potentially resequenced. |
| 13:11.12 | [TK]D-Fender | IMS77: Jitter / PL may make things seem different coming in vs out. |
| 13:11.21 | IMS77 | A call B via Asterisk. PPS seq=1 between A and Asterisk, then SPS seq=0 between A and Asterisk then PPS=29674 between Asterisk and B then SPS=29675 between Asterisk and B |
| 13:11.21 | IMS77 | You can see if the first part seq numbers are not in order but in the 2nd part they are in the order |
| 13:13.41 | *** join/#asterisk aurelihein (~Aurelien@srvmail.castel.fr) |
| 13:17.41 | IMS77 | Why on the audio channel is it not working in the same way? |
| 13:18.27 | IMS77 | there is no repack, seq number is equal in both channels |
| 13:19.17 | IMS77 | SSRC is equal with audio but not with the video |
| 13:25.12 | *** join/#asterisk okeefematt (~okeefemat@122-59-246-95.jetstream.xtra.co.nz) |
| 13:27.16 | okeefematt | Does anyone know how to limit outbound calls to emergency on an IAX trunk for certain extensions? |
| 13:27.43 | WIMPy | Dialplan |
| 13:27.47 | WIMPy | As always |
| 13:30.11 | IMS77 | When i call an old version of a peer i have no jitter but with a new one I have a lot. Why audio passthrough but not the video? |
| 13:36.01 | [TK]D-Fender | We have no details to go on to make a suggestion... |
| 13:37.19 | IMS77 | what kind of details you need? |
| 13:38.32 | [TK]D-Fender | Seeing the calls for starters.. |
| 13:38.32 | okeefematt | I tried entering the patterns into the extensions.conf under the context named [internal-line], saved the file and was able to still call the number i tried to restrict |
| 13:39.32 | [TK]D-Fender | okeefematt: Then you have done it incorrectly. |
| 13:39.35 | *** join/#asterisk jwr (~quassel@205.196.167.54) |
| 13:40.01 | [TK]D-Fender | okeefematt: That context name doesn't really tell us anything. |
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| 13:42.49 | okeefematt | In the gui, under 'extensions', was a field called 'context' and 'internal-line' was the value |
| 13:44.03 | okeefematt | for that particular extension - I thought that I would have to place a dialplan under [internal-line] in extensions.conf? |
| 13:44.59 | [TK]D-Fender | okeefematt: That context name is not special in any way. You can call them whatever you want. [internal-line] means nothing to us. What matters is the entirety of what you make |
| 13:45.22 | [TK]D-Fender | okeefematt: I could have a single context named [fred] for all it matters to anyone else. |
| 13:46.22 | [TK]D-Fender | okeefematt: And "GUI" is something we don't specifically support here |
| 13:46.41 | okeefematt | the web interface isn't supported? |
| 13:46.47 | [TK]D-Fender | okeefematt: There are NAMY different GUI's for Asterisk .. you should also be a lot more specific about which one you're using... |
| 13:47.16 | [TK]D-Fender | "the web interface" is a VAGUE term when there are a dozen of them |
| 13:47.24 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:48.02 | [TK]D-Fender | MANY* |
| 13:48.12 | [TK]D-Fender | is feeling a little dyslexic this morning... |
| 13:48.23 | okeefematt | okay fair enough, sorry - I've spent a total of 3 hours trying to configure a system for a charity |
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| 13:48.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:51.44 | okeefematt | I believe it's freepbx, there might be more than one version |
| 13:51.59 | [TK]D-Fender | Meaning? |
| 13:52.59 | [TK]D-Fender | If you're using freepbx you should never ever be touching extensions.conf in the first place. |
| 13:53.14 | *** join/#asterisk jwr__ (~quassel@205.196.167.54) |
| 13:53.25 | [TK]D-Fender | That all gets blown away every time you commit any other change |
| 13:53.30 | okeefematt | okay, well that's what I thought - but it didn't really provide me with a 'dialplan' for each extension |
| 13:54.02 | okeefematt | which is why I probably found the extensions.conf didn't work |
| 13:54.12 | okeefematt | with the dial plans |
| 13:55.43 | okeefematt | Are you saying that any changes I make in nano/vim will get destroyed if I update via freepbx? |
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| 14:00.44 | okeefematt | Seems crazy that's the case, as the article on the 'freepbx.org' site 'ow-to-give-a-particular-extension-different-or-restricted-trunk-access' says to alter extensions.conf |
| 14:04.03 | Qwell | okeefematt: link? |
| 14:04.25 | okeefematt | http://www.freepbx.org/support/documentation/howtos/how-to-give-a-particular-extension-different-or-restricted-trunk-access |
| 14:04.53 | *** join/#asterisk tom______ (~tom@wsip-24-120-113-148.lv.lv.cox.net) |
| 14:05.25 | Qwell | and where does it say to change extensions.conf? |
| 14:05.40 | alami | i want to make changes to extensions.conf and dialplan reload with a php script, does AMI, or AGI? help |
| 14:05.55 | okeefematt | 'this one is written for those who have some experience with adding contexts to /etc/asterisk/extensions_custom.conf o' |
| 14:06.46 | Qwell | okeefematt: uh huh, so where does it say to change extensions.conf? |
| 14:07.03 | okeefematt | Ah, I changed extensions_custom.conf |
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| 14:08.45 | okeefematt | so I placed the dialplans in extensions_custom.conf (sorry) |
| 14:09.43 | [TK]D-Fender | okeefematt: What version are you running? |
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| 14:13.24 | okeefematt | I'm not 100% sure, but it could be trixbox 2.6 |
| 14:13.33 | [TK]D-Fender | Ancient garbage. |
| 14:13.46 | [TK]D-Fender | That was discontinued like 4 years ago... |
| 14:13.48 | okeefematt | Yeah, was wondering whether to upgrade |
| 14:15.10 | okeefematt | Is it more difficult to configure? or just buggy? |
| 14:15.40 | [TK]D-Fender | It has fewer options to control the flow |
| 14:15.50 | [TK]D-Fender | you know,.. years and years of improvements and new modules |
| 14:16.01 | [TK]D-Fender | including one for controlling outbound route access |
| 14:16.23 | okeefematt | yeah,.. even though being 100% new to asterisk today, I felt that the options were limited |
| 14:17.15 | okeefematt | Thanks [TK]D-Fender |
| 14:17.29 | [TK]D-Fender | okeefematt: If you want something reasonable for you to do yourself.. you'll need something more basic |
| 14:17.48 | [TK]D-Fender | otherwise you're talking a chunk of manual dialplan. And the approach that one took... is pretty bad |
| 14:18.23 | [TK]D-Fender | you should be able to change the order of your outbound routes. Go test that in the version you have. |
| 14:19.30 | tom______ | I have asterisk set up in a sorta b2bua format. A call comes in, I answer it and play a message. After the message I forward it to an outside did. When that did sends a bye, I want to not proxy the bye. To just kill that call leg and play another message. Can someone point me in the right direction on how to do that? |
| 14:22.38 | okeefematt | [TK]D-Fender: Thanks, yeah I might try upgrading first - then hopefully it'll be a bit simpler to setup the outbound calling rules! |
| 14:22.47 | WIMPy | 'core show application Dial' There are several options to continue. |
| 14:24.07 | [TK]D-Fender | tom______: Asterisk is ONLY a B2BUA. It is never a "proxy", or "router". |
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| 14:25.10 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:27.16 | tom______ | correct but as a b2bua a call has a uas and uac side. I want to stop the uac side from passing the bye to the uas side |
| 14:28.30 | [TK]D-Fender | tom______: It doesn't "pass" it. Your dialplan determines what steps are taken. |
| 14:28.41 | [TK]D-Fender | tom______: read Dial's instructions. |
| 14:30.53 | tom______ | I am using dialstatus to catch the bye and go to a macro but it does not seem to see the bye |
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| 14:31.41 | [TK]D-Fender | tom______: If you leave dial... then the call is ended.... |
| 14:32.02 | [TK]D-Fender | tom______: You should probably SHOW us what you're doing so we can tell you where you went wrong... |
| 14:32.09 | [TK]D-Fender | PASTEBIN is your friend... |
| 14:32.10 | [TK]D-Fender | ~pb |
| 14:32.18 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:33.02 | tom______ | http://pastebin.com/CrbuueBW |
| 14:36.26 | [TK]D-Fender | tom______: Once either side hangs up that call the entire channel will die. |
| 14:36.44 | [TK]D-Fender | tom______: You have not followed Dial's instructions. |
| 14:36.55 | [TK]D-Fender | tom______: "core show application dial" <- read the options again |
| 14:37.21 | tom______ | ok. so the g option |
| 14:37.38 | [TK]D-Fender | yes |
| 14:39.10 | tom______ | thank you for the help |
| 14:39.54 | [TK]D-Fender | You're welcome. |
| 14:44.13 | Flash_G0rd0n | Hello guys,I'm trying to implement RealTime with Asterisk.So far I managed to add users,register them and dial.The problem I'm facing is they are not auto removed from cache when they disconnect although rtautocleaner=yes.Any suggestions or hints ? |
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| 14:49.53 | voip | woot |
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| 16:06.23 | Blue2000k | Morning |
| 16:08.09 | Blue2000k | Where would be a good place to ask questions in regards to asterisk and trying to resolve an issue with our phone system? |
| 16:08.34 | Qwell | ~ask |
| 16:08.35 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 16:11.13 | Blue2000k | Here goes nothing. We have the asterisk phone system and we are using Aastra phones on the system. We use FreePBX Admin, Elastix, and WinSCP to configure our systems. The person who setup the system is no longer with the company. We are able to get the phones to show the correct user at the extension but when an extension calls another extension the caller ID is incorrect. For example... |
| 16:11.15 | Blue2000k | ...Bob at 101 calls Susan at 102, Susan shows the Stacey is calling from 101 and not Bob. Any thoughts or suggestions on where I may be able to find the config file to change that? |
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| 17:13.12 | Phunter | I am having a hard time with queues.conf. http://pastebin.com/bii2j8xq |
| 17:13.36 | Phunter | with the 'n' option, It works well, except the * to exit the queue doesn't. |
| 17:13.51 | Phunter | and the timeout is like 60 seconds. |
| 17:14.53 | Phunter | correction: with 'n' the * to exit the queue works, but the timeout is ignored. Without it, the queue is not exitable. |
| 17:15.16 | Phunter | Plus the periodic-announce-frequency is ignored with 'n' option it seems. |
| 17:15.44 | [TK]D-Fender | Phunter: I don't see any reason * should exit there.... |
| 17:16.03 | Phunter | H option allows * to exit the queue by Caller. |
| 17:16.29 | Phunter | but only when n option is used as well. |
| 17:16.46 | [TK]D-Fender | Phunter: that is to end the call with the MEMBER once established. |
| 17:17.17 | Phunter | Ohhh I seee. |
| 17:17.22 | [TK]D-Fender | Phunter: That is not an 'exit the queue while waiting" |
| 17:17.27 | Phunter | Is there any way to exit the que while waiting? |
| 17:17.31 | Phunter | queue* |
| 17:17.57 | Phunter | instad of 'trapping' the caller? |
| 17:18.13 | [TK]D-Fender | Phunter: "context" |
| 17:18.32 | Phunter | Thats only when the queue itself exits.. |
| 17:18.38 | Phunter | If i am not mistaken |
| 17:18.58 | Phunter | But I still have to find a way to kick them out if they want to. |
| 17:19.00 | [TK]D-Fender | context=debug-out ; Here we go when the caller presses a single digit, while in the queue |
| 17:19.20 | [TK]D-Fender | Your comment was exactly on-spot |
| 17:19.31 | [TK]D-Fender | you want out? THIWS is how you let them out. |
| 17:19.53 | Phunter | Okay so how would a caller exit while in the queue to get to said context? |
| 17:21.15 | Phunter | Set timeout to like a minute, and if they want out press something, or kick them back to the queue? |
| 17:22.26 | Phunter | within that context? |
| 17:23.37 | [TK]D-Fender | enter queue. Press key. Have a match in that context just like it says. Done. |
| 17:25.06 | Phunter | The Queue Application doesn't accept DTMF from what I have tested.. |
| 17:25.36 | Phunter | Not in the sense that Background would. |
| 17:25.42 | Phunter | to my knowledge. |
| 17:26.09 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 17:26.48 | stevePearPear | Hi anyone faces this issue? Im using SIPML5 with Asterisk and Im getting Error 502 when SIPML5 tries to make a connection to Asterisk through wss |
| 17:27.04 | stevePearPear | from Asterisk: == WebSocket connection from '127.0.0.1:33930' for protocol 'sip' accepted using version '13' |
| 17:27.19 | stevePearPear | when i try to connect again, i do not see any response from Asteriskâs console |
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| 17:28.21 | [TK]D-Fender | Phunter: It does. |
| 17:28.39 | [TK]D-Fender | Phunter: And what you showed earlier does not work. |
| 17:28.40 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:30.24 | Phunter | Okay. I changed it to 1 instead of * |
| 17:30.43 | Phunter | Better yet, im gonna move that into the original context. |
| 17:33.12 | cusco | what would cause: astobj2.c: bad magic number for object 0x7fb660282128. Object is likely destroyed. |
| 17:33.16 | cusco | ? |
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| 17:34.37 | Phunter | [TK]D-Fender: http://pastebin.com/wFcNmDHu |
| 17:34.44 | Phunter | not so sure Queue accepts DTMF. |
| 17:34.54 | mjordan | cusco: a bug. What version are you running? |
| 17:35.09 | cusco | 11.10.0 |
| 17:35.22 | [TK]D-Fender | Phunter: Nor should it between those 2 pastebins of yours |
| 17:35.27 | mjordan | hm. Did you get a crash shortly thereafter? |
| 17:35.33 | cusco | mjordan: plenty of these printed and asterisk dies |
| 17:35.36 | cusco | yes |
| 17:35.41 | cusco | last in log is this channel... |
| 17:35.52 | cusco | I can paste the whole channel from full.log |
| 17:35.56 | mjordan | the log isn't useful |
| 17:36.00 | mjordan | a backtrace, OTOH, would be |
| 17:36.18 | Phunter | [TK]D-Fender: http://pastebin.com/HgXYD7hV <-- Try that. |
| 17:36.23 | mjordan | cusco: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 17:36.34 | mjordan | please file an issue once you have a backtrace generated per the instructions on the wiki |
| 17:36.36 | cusco | hm |
| 17:37.25 | [TK]D-Fender | Phunter: Nope, lok at th4e CONTEXT |
| 17:38.51 | Phunter | facepalms. |
| 17:40.18 | cusco | mjordan: without optimization is the 'DONT OPTIMIZE' I assume (to recompile) |
| 17:40.29 | cusco | debug threads as well? |
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| 17:48.28 | mjordan | no |
| 17:48.37 | mjordan | debug threads has nothing to do with crashes, and generally should never be used. |
| 17:49.05 | mjordan | DEBUG_THREADS is (a) a performance killer; (b) has known issues that can cause lock ups, and (c) should only be used when you *know* you have a deadlock and you're trying to get evidence for a bug marshal |
| 17:49.13 | mjordan | otherwise, do NOT ever enable DEBUG_THREADS. |
| 17:49.37 | mjordan | DONT_OPTIMIZE and BETTER_BACKTRACES are what that page calls out, and those are the two options that should be enabled, as they make core dumps useful |
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| 17:58.42 | cusco | ok |
| 17:59.34 | cusco | mjordan: better backtraces really needed? it has some extra dependencies - libbfd |
| 17:59.59 | mjordan | it is extremely desireable :-) If the backtrace you provide with DONT_OPTIMIZE doesn't have sufficient information, we'll ask for BETTER_BACKTRACES as well |
| 18:00.05 | cusco | ok |
| 18:03.06 | cusco | thank you mjordan |
| 18:03.33 | mjordan | np, thanks for reporting it |
| 18:07.24 | cusco | not yet, I'll have to wait for it again |
| 18:07.28 | cusco | probably tomorrow |
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| 18:41.30 | Flash_G0rd0n | Hey guys what can be the problem of not auto-expiring real-time peers when rtautocleaner=yes on asterisk 11.8.1 ? |
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| 18:58.07 | mjordan | hm. And now he left. |
| 18:58.15 | mjordan | I'm wonder what rtautocleaner is... |
| 19:06.15 | file | mjordan, squeaky clean! |
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| 19:38.16 | TazzNZ | morning all |
| 19:38.26 | Chainsaw | Evening Tazz. |
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| 22:55.54 | voip | greetings earthlings |
| 22:58.11 | Lepon | Hey people, since you have such a great habbit of point me in the right direction to solve my issues, wondering if you have any thoughts of my latest problem. |
| 22:59.23 | Lepon | I need to record a message that is being played to a channel with chanspy but I also need to record the convosation in that bridged channel that comes after the chanspy has left. I was just recording the base conversation but that doesn't seem to record anything played by chanspy |
| 22:59.50 | Lepon | Also thank you [TK]D-Fender for your suggestion that worked nicely to get me this far |
| 23:00.33 | Lepon | I looked at recording the message through the chanspy record function but it leaves after it plays the message so I don't get the rest of the convosation that comes after |
| 23:01.29 | Lepon | But I don't want to leave the chanspy on their forever recording because its only an arbitart amount of time I need to record after the message plays which will be triggered by the people in the base channel to stop the recording |
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| 23:15.36 | Lepon | or alternativly, is there a way from a base channel to control or execute anything on the channel that is spying on you? |
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