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| 04:46.41 | [gnubie] | waves |
| 04:47.07 | [gnubie] | hi tzafrir.. |
| 05:06.43 | nix8n82 | What framework for agi, in python or php, is up to date with asterisk 11 and 13? |
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| 07:18.11 | themrrobert | not sure what happened but all of a sudden asterisk stopped putting ice-frag and ice-pwd in the SDP |
| 07:18.33 | themrrobert | i recompiled ensuring that all requried prereqs are installed, but no joy. double checked my settings, all correct |
| 07:19.50 | themrrobert | libuuid, uuid, and the -devel versions, lib-srtp, --with-cryto, --with-ssl, --with-srtp, icesupport=yes everywhere, all clients have dtlssetup=actpass (actpass is showing up in the SDP, just not the ice-xxx ) turnserver + user/pass + stunserver in rtp.conf |
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| 07:21.06 | themrrobert | only thing i did between it working and not was: upgrade glibc (minor version upgrade, from x.x.xx-yy to x.x.xx-yyy (only yy changed) and enable better_backtraces and dont_optimize |
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| 07:26.34 | themrrobert | actually it looks like it's not putting a=setup=actpass, maybe just active |
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| 07:36.39 | themrrobert | i see in the debug that asterisk is also failing to interpret ice-pwd / ice-ufrag which isn't good |
| 07:40.34 | themrrobert | version asterisk 11.17.1 |
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| 07:55.13 | themrrobert | okay I fixed it. I unziped the tarball for a clean build area and reconfigured/compiled and it worked. idk how it broke as it used to be a working build root, but oh well, it's working now that's all i care about! |
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| 08:10.36 | EmleyMoor | Is there any way to ring two DAHDI channels at the same time with the same ring cadence without specifying them individually? |
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| 09:09.43 | WIMPy | EmleyMoor: How do you specify them at all? I can't remember seing such thing. |
| 09:14.29 | proute | Hi everybody, |
| 09:15.06 | proute | I try to use visoconference with Asterisk 1.8.32.3 (The last release). It's work fine with 2 phone in point to point mode. |
| 09:15.41 | proute | Now I want to use visoconf with meetme (conference mode). |
| 09:15.57 | WIMPy | What is that thing? |
| 09:16.16 | proute | Does Asterisk support visioconf in mutlipoint mode (more than 2 users with video)? |
| 09:16.17 | proute | If yes, which module should I use? |
| 09:16.17 | proute | thanks for your help |
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| 09:17.14 | WIMPy | I guess the question should be the other way round: What does that visioconf thing support. I haven't heard of that so far. |
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| 09:24.03 | proute | Hi WIMPy |
| 09:25.10 | proute | I would like use Asterisk to do visioconf with 5 users for example. And I want to see on my videophone, the others users (For example 4 videos on my screen) |
| 09:25.36 | WIMPy | What is that visioconf thing? |
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| 09:27.04 | proute | visoconference=video conference |
| 09:27.38 | proute | I would say video conference (sorry) |
| 09:27.54 | WIMPy | Err. Are you just saying visioconf to refer to the general concept of a video conference? It sounded like some sort of product name to me. |
| 09:28.01 | WIMPy | Ok |
| 09:28.48 | WIMPy | MeetMe cant's do that. It's G.711 only. But MeetMe has been pretty much obsolete for some time. ConfBridge can do it, available since Asterisk 11. |
| 09:29.14 | WIMPy | But it can't do video mixing. It can only switch video stream to whoever is talking at the moment. |
| 09:29.51 | proute | WIMPy, yes I try Confbridge whit Astersk 11. And I can't see all users |
| 09:30.06 | WIMPy | No, only one at a time. |
| 09:31.15 | proute | Is there a module in Asterisk able to overpass this feature? |
| 09:31.34 | proute | to see all users on the same screen? |
| 09:31.35 | WIMPy | Not so far. |
| 09:31.38 | proute | :( |
| 09:31.40 | proute | :'( |
| 09:32.03 | WIMPy | There have been discussions, but I don't think we've heard of anyone actually doing the work. |
| 09:33.10 | proute | Ok |
| 09:36.01 | proute | DO you know if an open source solution exists to do this? |
| 09:38.23 | drazoro | tt |
| 09:38.32 | drazoro | t |
| 09:38.49 | WIMPy | No. I have no idea. |
| 09:39.43 | proute | arf |
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| 10:11.00 | EmleyMoor | WIMPY: DAHDI/xry rings channel x with ring pattern y |
| 10:11.51 | EmleyMoor | I want a way to say DAHDI/(1&2)r3 other than DAHDI/1r3&DAHDI/2r3 |
| 10:14.21 | WIMPy | I asked Dr Google in between and couldn't find that feature. |
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| 10:49.42 | MadHatter42 | anyone ?> |
| 10:50.32 | MadHatter42 | anyone ?> |
| 10:50.40 | MadHatter42 | i was trying to set up a dial plan for chanspy |
| 10:50.44 | MadHatter42 | but i'm a bit unclear |
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| 11:17.05 | nix8n82 | I know it is a matter of opinion. What is a good framework/library for agi and ami in php or python? |
| 11:17.31 | nix8n82 | That wor |
| 11:18.09 | nix8n82 | That is current with asterisk 11 and 13 |
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| 11:18.26 | WIMPy | What do you want it for? |
| 11:21.03 | nix8n82 | Mainly to access database info and update call routes and endpoints to dial |
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| 11:22.33 | WIMPy | And why do you want something in between? |
| 11:22.51 | nix8n82 | I want to be able to failover to a remote server if local goes down or needs maintance |
| 11:23.04 | MadHatter42 | chanspy anyone ? |
| 11:23.56 | nix8n82 | And somewhat be able to dynamicly add multiple uc to a extension number |
| 11:25.08 | nix8n82 | So if a user dials 10 I i |
| 11:25.48 | nix8n82 | It rings dev 1 dev2 and dev3 |
| 11:26.45 | WIMPy | That sounds like somethign you could fetch from any database and don't need to talk to Asterisk about. |
| 11:27.43 | nix8n82 | Also if they take a dev3 off local net and register remote and it acts like it is on the local pbx |
| 11:28.50 | WIMPy | Asterisk doesnt cae where a decive is. |
| 11:29.35 | WIMPy | Unless you need NAT support, but you still don't have to make a difference. |
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| 11:40.25 | nix8n82 | If I talk to a database and asterisk doesn't need to know what would I use for sip signalli |
| 11:40.42 | nix8n82 | Signaling |
| 11:41.14 | nix8n82 | And possible media proxy? |
| 11:41.43 | WIMPy | You would read the database before dialling. |
| 11:42.04 | WIMPy | How did you plan to do it? |
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| 11:42.31 | WIMPy | Dial()ing that is. |
| 11:42.48 | nix8n82 | Through odbc or the internal astdb |
| 11:44.39 | nix8n82 | Makes sense |
| 11:46.43 | nix8n82 | What about just for fun anyone know of agi and ami libraries that have kept up with asterisk 11 and 13? |
| 11:50.22 | nix8n82 | I want to check If the remote has peers registered to it before dialing and include it if it does. And the other way around if the uc is registered to the remote |
| 11:51.55 | WIMPy | That's what Asterisk will do by itself if you enable qualify. |
| 11:52.03 | nix8n82 | Based of uid.ext-number |
| 11:52.21 | WIMPy | What's that? |
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| 11:55.33 | nix8n82 | MacorUID.1000 so multiple device claim to have exten 1000 and build the dial string based on all peers ending in 1000 |
| 11:57.06 | MadHatter42 | i was trying to set up a dial plan for chanspy |
| 11:57.21 | MadHatter42 | to listen a specific extension by dialing for exmaple *111+ext |
| 11:57.27 | MadHatter42 | any ideas ? |
| 11:57.37 | MadHatter42 | i've set it in the from-internal-custom but still its not working |
| 11:58.23 | WIMPy | nix8n82: I don't really understand your concept. Those peers need to be predifined anyway. |
| 11:58.47 | WIMPy | MadHatter42: I guess you should look at ExtenSpy instead. |
| 11:59.16 | MadHatter42 | WIMPy, let me check that |
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| 12:00.36 | MadHatter42 | WIMPy, that looks like exactly what I need |
| 12:00.40 | MadHatter42 | any examples on that ? |
| 12:01.31 | WIMPy | That's what Dr Google is there for. |
| 12:01.51 | nix8n82 | They do and based on if they are on our network they go |
| 12:02.05 | nix8n82 | To the local pbx |
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| 12:02.46 | MadHatter42 | WIMPy, yeah but i'm a having weird results |
| 12:02.53 | MadHatter42 | thats why i asked |
| 12:02.56 | nix8n82 | Or if they are ont lte or another network not our own. They register to a public server |
| 12:03.28 | MadHatter42 | WIMPy, if i wanto do that *111+ext |
| 12:03.37 | nix8n82 | Basicly using dns to determine which server they register to |
| 12:04.21 | WIMPy | nix8n82: And why do you care? What's wrong with just calling both? |
| 12:04.25 | nix8n82 | And far as the user is concerned it is the same thing |
| 12:05.07 | MadHatter42 | WIMPy, http://pastebin.com/dpH9u49q |
| 12:05.10 | nix8n82 | I guess nothing really |
| 12:05.12 | MadHatter42 | am I doing this right ? |
| 12:05.57 | nix8n82 | I just dont want to hard code it in my dial app |
| 12:06.17 | MadHatter42 | #exten => *111xxxxxxx.#,n,ExtSpy(sip/${EXTEN:6},q) |
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| 12:06.27 | MadHatter42 | is it the correct way ? |
| 12:07.00 | nix8n82 | Kind of why I want to see if they are reachable before dial |
| 12:08.24 | WIMPy | nix8n82: Can I safely interpret this as There's no reason, just educational purposes? |
| 12:09.38 | WIMPy | MadHatter42: That's just comments and they don't look like they would so sonething sensible if they weren't. |
| 12:09.59 | nix8n82 | More educational than practical at this point..hopefully practical latte |
| 12:10.04 | nix8n82 | Later on |
| 12:10.14 | WIMPy | The extension is not a pattern and wouldn't make sense as a pettern, either. And the number of cut off digits doesn't match, either. |
| 12:10.15 | MadHatter42 | WIMPy, i set them coments for the moment but if i remove the comments would that work ? |
| 12:10.55 | nix8n82 | Unless there is a better way to do what needs to be done |
| 12:11.01 | themrrobert | MadHatter42 no |
| 12:11.07 | MadHatter42 | any more tangible examples ? |
| 12:11.25 | MadHatter42 | exten => 1234,1,ExtenSpy(399709,q) |
| 12:11.26 | WIMPy | Read the chapter about dialplan basics in the |
| 12:11.31 | WIMPy | ~book |
| 12:11.31 | 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 |
| 12:11.45 | themrrobert | MadHatter42 first of all, it's ExtenSpy, so that's the wrong app. 2nd of all, ChanSpy uses TYPE/Extension , ExtenSpy uses exten@context |
| 12:11.57 | WIMPy | Without those you won't get very far. |
| 12:12.16 | MadHatter42 | for instance if i dial 1234 to hear 399709 would this work exten => 1234,1,ExtenSpy(399709,q) |
| 12:12.16 | MadHatter42 | <PROTECTED> |
| 12:12.20 | themrrobert | also chanspy is chanprefix, so you can rotate multiple different channels with the same prefix MadHatter42 |
| 12:12.21 | WIMPy | Yes, the parameter has wrong syntax as well. |
| 12:12.47 | WIMPy | Far to many errors. |
| 12:12.49 | MadHatter42 | WIMPy, any pdf ? |
| 12:13.03 | nix8n82 | Was the book for me? |
| 12:13.26 | WIMPy | Check the links infobot gave you. The latest edition is not available as pdf for free IIRC. |
| 12:13.37 | MadHatter42 | i'm not asterisk guru WIMPy i'm just stuck with asterisk for the moment |
| 12:13.43 | WIMPy | No, the book is for MadHatter42. |
| 12:13.44 | themrrobert | Google is your friend in that regard MadHatter42 |
| 12:13.47 | MadHatter42 | i would like to learn it |
| 12:14.12 | themrrobert | definitely read that book first, it's where i started |
| 12:14.31 | WIMPy | That's what it's there for. |
| 12:14.40 | themrrobert | ofc i went out and bought it from the store because my new job featured it lol |
| 12:14.45 | themrrobert | well old new job |
| 12:20.48 | nix8n82 | I have been up for way too long. WIMPy, thanks for talking with me I appreciate it. |
| 12:22.38 | nix8n82 | Good night or morning. Im idling |
| 12:23.21 | WIMPy | 2pm here :-) |
| 12:27.51 | themrrobert | u in UK area WIMPy? |
| 12:28.08 | themrrobert | or Germany/France depending on which side of 2pm you're in |
| 12:28.50 | WIMPy | de |
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| 13:19.55 | leffelad | Hi, Anybody had any luck with monitoring tools with asterisk for parsing the logs and SIP messages, everywhere I look online seems to be content that was last edited at the turn of the millenium |
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| 13:29.12 | leffelad | anyone? |
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| 13:33.45 | joako | In my sip.conf I have two identical peers, except for the name, callerid, and secret. I configure these as line1, and line2 to the same phone. When I place a call from line1 it shows the callerid, but when I place a call from line2 it shows unknown. |
| 13:34.33 | joako | In extensions.conf I add: Set(CONNECTEDLINE(name,i)=CID: ${CALLERID(number)}) and then for line1 the phone shows the callerid number, but on line2 it shows the SIP username |
| 13:40.02 | leffelad | joako, nobody seems to be replying today |
| 13:40.47 | joako | leffelad, thats normal. But im an idiot, I had calerid = .... |
| 13:41.37 | leffelad | joako, ah lol, good stuff that its sorted |
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| 13:55.19 | polysics | hello! |
| 13:55.30 | polysics | we have this interesting quirk that maybe someone can help me solve |
| 13:56.07 | polysics | two channels are bridged, with a call coming in from a DID, other leg getting "originate"d out, then they get bridged and start chatting |
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| 13:56.35 | polysics | every 15 minutes, the DID provider sends a reINVITE. That causes the bridge to break. |
| 13:57.08 | polysics | but if I change the dialplan so instead of going through Adhearsion, the call just Dial()s the other peer, the reINVITE still comes but nothing happens |
| 13:57.14 | polysics | does my explanation make sense? :) |
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| 14:10.15 | *** join/#asterisk kharwell (kharwell@nat/digium/x-xsfjkpmmquqxazde) |
| 14:11.31 | polysics | is there anything I could go get that helps figure out what is going on? |
| 14:12.11 | *** join/#asterisk Sudravirodhin (~Sudraviro@162.218.22.124) |
| 14:12.44 | carrar | Sounds like you fixed it already |
| 14:12.58 | polysics | carrar: not sure I understand :) |
| 14:13.27 | carrar | You said you used dial and "nothing happens" aka it hangs up |
| 14:13.44 | polysics | then I mis-expressed myseld |
| 14:13.55 | polysics | I mean "the reinvite comes in but hte bridge does not break" |
| 14:14.02 | carrar | which is what you want |
| 14:14.15 | polysics | yeah, but I need to use originate in the actual application |
| 14:14.44 | carrar | Sounds like you need to be asking the people who wrote Adhearsion |
| 14:15.15 | polysics | which would be about 30% me, by the way |
| 14:15.38 | carrar | Since it works if you use dial |
| 14:15.55 | polysics | you are fundamentally right |
| 14:16.17 | polysics | this is probably something that needs to be fixed either in Ahn or with an SBC |
| 14:17.06 | carrar | What is it you are thinking should happen that isn't happen when you use dial to create a new call out? |
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| 14:19.33 | carrar | Or are you really looking for a SIP PROXY device? |
| 14:19.51 | polysics | I am not sure at this point. Ahn sees the INVITE as a new offer and breaks the bridge |
| 14:20.22 | polysics | Kamailio could certainly just eat that INVITE |
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| 14:21.07 | carrar | Sounds like to you need to find out what your stuff in Adhearsion is doing differently then when you use dial |
| 14:21.57 | carrar | since a re-invite is normal |
| 14:22.08 | carrar | making sure the call is valid |
| 14:22.08 | polysics | yes, it took me a while but I found out that |
| 14:22.32 | polysics | and it's not controlled by directmedia=no or session-timers settings |
| 14:22.37 | polysics | because it is actually neither |
| 14:23.21 | carrar | I've not used Adhearsion, perhaps someone else has |
| 14:24.18 | carrar | Just out of curiosity, what are you using Adhearsion for in your specific case? |
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| 14:28.48 | polysics | it's a complex call center situation where a call comes in and is routed to agents based on time of day, actually being logged in, and a few other factors |
| 14:29.03 | polysics | plus calling them on cell phones out of hours but only for certain callers, etc |
| 14:29.07 | carrar | that sounds like basic asteirsk |
| 14:29.22 | polysics | plus recording calls, logging the calls, POSTing to a web app that pops a screen, etc |
| 14:29.23 | carrar | not something that needs some external complicated software |
| 14:30.02 | carrar | web app and screen pops would be external |
| 14:30.38 | polysics | I agree in principle, but it's how it is. It grew out of the outbound call center software which is way easier to grok in Ruby |
| 14:30.51 | polysics | this got started before ARI, mind you :) |
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| 15:34.23 | Sudravirodhin | Does anyone know how the implementation of Attended Transfers works in terms of channel Tech, etc.? |
| 15:34.32 | Sudravirodhin | That is, the features.conf atxfer. |
| 15:34.50 | [TK]D-Fender | Has nothing to do with tech... |
| 15:35.31 | Sudravirodhin | [TK]D-Fender: I plead ignorance, could you go over what happens during an atxfer? |
| 15:35.46 | [TK]D-Fender | What is there to explain? |
| 15:36.04 | [TK]D-Fender | * is sitting waiting for the DTMF from the channel and it simply redirects to somwhere else in the dialplan. |
| 15:36.39 | Sudravirodhin | [TK]D-Fender: Where in the dialplan? I know it's a core function so seeing what it does in the dialplan would be a great help if it is visible. |
| 15:36.52 | [TK]D-Fender | "does in the dialpln"? |
| 15:37.02 | [TK]D-Fender | It just sends that channel to the extension dialed |
| 15:37.30 | [TK]D-Fender | There is a channel vairable to force the location of where those will land as well. |
| 15:37.52 | Sudravirodhin | TRANSFER_CONTEXT? |
| 15:38.27 | themrrobert | So asterisk segfaulted in the middle of a free(), i'm not sure what the wiki means when it says that means memory corruption. does that mean the memory is bad or that asterisk is corrupting the heap/stack? |
| 15:38.40 | [TK]D-Fender | Yes |
| 15:39.31 | themrrobert | any way to determine which one? It's on AWS so i can't do a real memtest, though i did a userspace memtester and it didn't find any errors |
| 15:39.57 | Sudravirodhin | themmrrobert: I think he was answering my question. |
| 15:40.09 | Sudravirodhin | themrrobert: Sorry for overlapping your question. |
| 15:41.10 | Sudravirodhin | [TK]D-Fender: Say I create a context to route my call where I wish, why would setting a callerid be ignored by the atxfer? I know I've asked variations of this question before but I'm still baffled Asterisk would outright disregard a variable just because of atxfer. |
| 15:41.33 | themrrobert | Sudravirodhin np at all |
| 15:42.16 | [TK]D-Fender | Sudravirodhin, an attended transfer starts as a NEW call by the channel starting it. |
| 15:42.33 | [TK]D-Fender | Sudravirodhin, So if you have a SIP device on your side doing it then that is a new call on their behalf |
| 15:42.40 | [TK]D-Fender | and that is THEIR callerid |
| 15:42.45 | [TK]D-Fender | that is what attended implies |
| 15:42.56 | [TK]D-Fender | You see the transferrer's callerid, not the transferee. |
| 15:43.08 | Sudravirodhin | [TK]D-Fender: Implying no inheritance, then. So I can't change the callerid in any way (forced or not). |
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| 16:34.47 | pa | WIMPy, so now i have the mISDN_dsp module loaded with dmtfthreshold=200 |
| 16:34.50 | pa | it goes better |
| 16:35.06 | pa | but with a few specific people i still have the same problem |
| 16:35.24 | pa | like if they are pressing buttons on their phones while speaking |
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| 17:00.20 | WIMPy | Hi pa. Well, As far as I remember 200 is the maximum accepted value. |
| 17:01.08 | WIMPy | The only thing that would really help would be if someone who understands it would take a closer look at the source code. |
| 17:02.10 | WIMPy | I'm pretty sure that level does not work as intended, but I'm not a methemtician. |
| 17:03.51 | *** join/#asterisk JeffC_NN (32cad19e@gateway/web/freenode/ip.50.202.209.158) |
| 17:06.00 | JeffC_NN | I have voicemail storage in ODBC, but it keeps silently falling back to file storage. odbc show all shows: http://pastebin.com/j9nKyKHz |
| 17:12.47 | JeffC_NN | voicemail.conf is http://pastebin.com/sDKzjWfJ |
| 17:16.43 | JeffC_NN | I have debug/verbose both on 10, nothing at all shows up about odbc or voicemail (besides dialplan and recording) |
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| 17:43.47 | themrrobert | 2nd segfault of the day, this time address out of bounds |
| 17:45.38 | mepholic | themrrobert: is it possible |
| 17:45.42 | mepholic | that you're running selinux |
| 17:45.44 | mepholic | or something? |
| 17:45.59 | themrrobert | definitely no selinux |
| 17:46.36 | themrrobert | It's AWS / amazon linux, so selinux is available but completely deactivated |
| 17:47.16 | themrrobert | actually it may not even be available, the /selinux directory is there but its empty |
| 17:47.28 | themrrobert | and theres no /etc/selinux or /etc/selinux.conf |
| 17:47.34 | mepholic | odd lol |
| 17:48.05 | mepholic | I was just thinking, maybe a kernel security module is denying asterisk access to a certain region of memory |
| 17:48.10 | mepholic | causing it to poop itself |
| 17:48.16 | themrrobert | i have a JIRA up with backtraces https://issues.asterisk.org/jira/browse/ASTERISK-25099 |
| 17:48.32 | themrrobert | earlier it faulted on FREE |
| 17:48.35 | themrrobert | invalid pointer |
| 17:48.55 | themrrobert | definitely stack/heap corruption, i'm thinking buffer overflow somewhere |
| 17:49.04 | jeffspeff | i'm trying to compile asterisk 13.3.2. i have enabled mp3 option in menuselect and i ran ./contrib/scripts/get_mp3_source.sh however, make fails with the error "format_mp3.c:39:24: fatal error: mp3/mpg123.h: No such file or directory" |
| 17:49.32 | file | there's been memory overwrite issues within OpenSSL itself and there also seems to be a problem within the res_rtp_asterisk of it itself, noone has as of yet narrowed it down exactly |
| 17:49.36 | jeffspeff | i can see the file exists in "/root/asterisk-13.3.2/contrib/scripts/addons/mp3/mpg123.h" |
| 17:49.50 | mepholic | jeffspeff: simple |
| 17:49.55 | mepholic | use a better codec than mp3 |
| 17:49.56 | themrrobert | hey , 3rd crash of the day. This time not segfault, but "Aborted" |
| 17:50.19 | jeffspeff | mepholic, i don't plan on using mp3 much, but i want the ability in there in case it is needed. |
| 17:50.33 | mepholic | jeffspeff: asterisk has to re-encode anything that's not using the codec the phones are using |
| 17:50.49 | mepholic | so it's just better to take your source content and re-encode it for asterisk |
| 17:50.50 | jeffspeff | i've never had this issue with pervious versions of asterisk. not sure what i'm missing or what's wrong this time. |
| 17:50.52 | mepholic | to g711 or g722 |
| 17:51.11 | mepholic | will prevent asterisk from doing extra work each time |
| 17:51.24 | mepholic | and you don't have to compile in 'propriatary' bullshit |
| 17:51.34 | WIMPy | jeffspeff: Why don't you use a free codec? |
| 17:51.58 | mepholic | most free codecs sound better than mp3 anyways |
| 17:52.06 | jeffspeff | i don't plan on using mp3 much, but i want the ability in there in case it is needed. |
| 17:52.20 | WIMPy | Yes. I see not point in using mp3 at all. |
| 17:52.23 | mepholic | it won't be if you just pre-encode the files to something that doesn't suck |
| 17:52.27 | themrrobert | this time it crashed in the ssl module |
| 17:52.31 | mepholic | I agree with WIMPy |
| 17:52.52 | mepholic | jeffspeff: if you >need< mp3 support in asterisk, you're doing something wrong |
| 17:53.20 | mepholic | you can achieve the same thing by converting an mp3 to a raw pcm file or, g711 with ffmpeg |
| 17:53.34 | jeffspeff | mepholic, if that's the case then why is it even a supported feature to begin with? |
| 17:53.47 | mepholic | because of people that "need" mp3 support |
| 17:54.05 | mepholic | it's a highly less than ideal feature to use |
| 17:54.15 | WIMPy | Since when do the fact that sometinh is supported mean that it makes sense? |
| 17:54.27 | WIMPy | does |
| 17:54.31 | mepholic | i mean |
| 17:54.36 | mepholic | look at ALL of mongodb |
| 17:54.37 | WIMPy | Ugh bad typing again today :-( |
| 17:54.44 | mepholic | :) |
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| 17:56.13 | themrrobert | actually probably was a invalid pointer passed to openssl that caused the crash, so it's still memory corruption |
| 17:56.25 | mepholic | are you using ECC? |
| 17:56.35 | themrrobert | i'm using AWS. i'm firing up a new instance |
| 17:56.39 | mepholic | oh |
| 17:56.41 | mepholic | DERP |
| 17:56.42 | themrrobert | and starting from scratch |
| 17:56.48 | mepholic | third time you've said that |
| 17:56.50 | mepholic | lool |
| 17:57.12 | themrrobert | userspace memtester found no issues, so not too hopeful of a new instance solving anything, but worth a shot |
| 17:57.20 | themrrobert | its definitely not the same as a kernel space memtest |
| 17:58.29 | jeffspeff | ~book |
| 17:58.29 | 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 |
| 17:59.00 | jeffspeff | just checking to see if they'd released a newer book yet. |
| 17:59.56 | WIMPy | As we heard that's very unlikely to happen. |
| 18:01.46 | JeffC_NN | (forgive the copy/paste, but I got no response earlier) I have voicemail storage in ODBC, but it keeps silently falling back to file storage. odbc show all shows: http://pastebin.com/j9nKyKHz voicemail.conf is http://pastebin.com/sDKzjWfJ I have debug/verbose both on 10, nothing at all shows up about odbc or voicemail (besides dialplan and recording) |
| 18:02.43 | JeffC_NN | version is: Asterisk 13.0.1 built by root @ sip on a x86_64 running Linux on 2015-04-11 00:26:17 UTC |
| 18:04.30 | JeffC_NN | Looking for any ideas regarding how I can test ODBC storage, and where to look for logs/errors, etc |
| 18:06.45 | jeffspeff | WIMPy, why's that unlikey? |
| 18:07.32 | WIMPy | Because writing books aparrently doesn't pay the rent. |
| 18:08.44 | jeffspeff | lol, true |
| 18:08.46 | themrrobert | i look at nearly a dozen backtraces, and had tunnel vision. couldn't see past the first items... digium guy takes 1 look at it and instantly knows where it is. Having seen what he said, it's not obvious.. |
| 18:09.07 | file | there's like 5 other issues open for the same thing |
| 18:09.28 | themrrobert | i gotta get back into c programming, never really got that far into it, spent more time on assembly than c. |
| 18:09.51 | themrrobert | @file I swear i checked first, I didn't know what I was looking for. I have a much better idea what to look for now |
| 18:11.12 | file | because it's memory corruption they also occur in different ways, but it's all memory corruption |
| 18:16.29 | file | newtonr, we really should have one ticket for DTLS related memory corruption/crashes |
| 18:18.39 | newtonr | file, i can round them up and link them to an issue |
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| 18:18.52 | newtonr | i thought we had one actually |
| 18:19.04 | file | I don't think so |
| 18:19.10 | newtonr | i'm probably thinking of an older issue |
| 18:32.27 | pa | WIMPy, oh i see. Well thanks, i guess this is the best result then :) |
| 18:32.38 | pa | but does it happen to you too? as far as i understood you also use mISDN |
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| 18:34.49 | WIMPy | pa: Yes it does. But 200 seems to work ok for me. |
| 18:35.32 | WIMPy | But it certainly could go higher als real DTMF is still detected reliably even at low volumes. |
| 18:36.06 | jepperl | Hey there :) i have this issue where, when i dial into a parking space (say 2025) and the ParkedCall() application runs to unpark the parked channel, a weird scratching noise is played to the parked channel. Have anyone experienced something like that? There really isnt anything special going on, im just unparking a guy, and he hears a scratching n |
| 18:36.06 | jepperl | oise.. |
| 18:36.15 | pa | WIMPy, well i think the situation improved now, but there's one specific person's cellphone that still triggers it as much as before |
| 18:36.20 | pa | no idea what phone that is |
| 18:36.49 | pa | i think with everybody else it's pretty much gone |
| 18:37.11 | pa | i could try 300 |
| 18:37.14 | pa | might help |
| 18:37.28 | pa | so the risk is that i could be unable to dial buttons once in call? |
| 18:38.01 | WIMPy | pa: Well, as I said, I think it's a bug in the DTMF detection regarding minimum levels. But That's for someone who understands that part. |
| 18:38.32 | WIMPy | Yes, but I think we're far away from that risk even at the highest possible setting. |
| 18:38.39 | pa | oh i see |
| 18:38.44 | pa | so i could try to increase |
| 18:38.53 | pa | and then maybe stop when i can't dial anymore |
| 18:39.20 | WIMPy | I'm not sure. 200 might be the maximum already. |
| 18:39.25 | pa | ah |
| 18:39.33 | pa | so i would have to at least rebuild the module |
| 18:40.00 | pa | well might be not a big deal. I use it from the svn anyway (the kernel one can't be patched for hfc-s) |
| 18:40.22 | marceloamorim | guys, I had this warning "Can't send 10 type frames with SIP write" when I call from CISCO 7960. Do you know why? |
| 18:42.03 | WIMPy | pa: Just looked and it takes values from 20 to 500. |
| 18:43.42 | WIMPy | Actually I used 400. |
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| 18:45.49 | pa | ah i see |
| 18:45.54 | pa | thanks :) |
| 18:45.57 | pa | i'll try to increase |
| 18:46.08 | WIMPy | So you just need to reload the module. |
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| 19:43.28 | billzeta | I'm having problems using PJSIP -- I routinely get the following error: res_pjsip_registrar.c:505 rx_task: Unable to bind contact 'sip:nicole@[public_ip_address]:51285;rinstance=1cb863ba21faaa6f' to AOR 'nicole'. I can call out fine with the extension and it shows registered in the SIP client, but asterisk shows the endpoint as unavailable and is unable to recieve inbound calls. Everything |
| 19:43.28 | billzeta | works fine when using chan_sip, just not with PJSIP. |
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| 19:51.43 | talntid | anyone know of a DID provider who has actual 1-800-#'s? I need a 1-800-# for a DID |
| 19:51.47 | talntid | not 855, not 888 |
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| 22:48.34 | Kobaz | is there a highres channel start time? ie: like CDR(start) |
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| 22:57.40 | Kobaz | oh sweet |
| 22:57.42 | Kobaz | there's a u option |
| 22:57.48 | Kobaz | for unparsed.. so the raw time value |
| 23:05.15 | Katty | infobot: seen jforde |
| 23:05.17 | infobot | Katty: i haven't seen 'jforde' |
| 23:05.27 | Katty | infobot: seend jaytee |
| 23:05.33 | Katty | infobot: seen jaytee |
| 23:05.33 | infobot | Katty: i haven't seen 'jaytee' |
| 23:05.40 | Katty | infobot: useless! |
| 23:05.40 | infobot | ACTION starts crying and hides from katty in the darkest corner of the room. :( |
| 23:05.46 | Katty | infobot: good bot |
| 23:05.46 | infobot | aw, gee, Katty |
| 23:05.49 | Katty | infobot: botsnack |
| 23:05.49 | infobot | aw, gee, Katty |
| 23:10.25 | Kobaz | it's katty |
| 23:10.33 | Katty | shh, don't tell :> |
| 23:10.36 | Katty | hugs Kobaz |
| 23:11.06 | Kobaz | yay |
| 23:11.11 | *** join/#asterisk areski (~areski@80.174.128.14.dyn.user.ono.com) |
| 23:11.29 | Kobaz | This warped Beatles LP sounds pretty good actually |
| 23:14.30 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 23:14.34 | Kobaz | need to get some glass pieces to flatten it |
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| 23:21.06 | *** 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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| 23:59.36 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |