| 00:00.09 | wedhorn | or actually, it seems to go passed that anyway. |
| 00:00.19 | wedhorn | apart from the log clutter, does it work? |
| 00:00.34 | snuff-home | yer no issues with it 'working' just spam on my screen |
| 00:01.57 | wedhorn | okay, suggest change type7937 to supported, but no button_template in get_button_template and |
| 00:01.58 | snuff-home | mm i see how now.. not really fast access to speed dials.. still have to go through a menu |
| 00:02.23 | wedhorn | change the log warning to a debug template message |
| 00:02.50 | wedhorn | yeah, but I'm guessing they are using the button messages rather than softkeys |
| 00:16.02 | snuff-home | mm where did my file upload go on jira.. :( |
| 00:16.12 | snuff-home | found |
| 00:26.32 | wedhorn | snuff-home: suggest a comment in 7937 that you want an empty template |
| 00:27.39 | wedhorn | also, we may want to change handle_message(SPEED_DIAL_STAT_REQ_MESSAGE) to send back something empty (but we should probably have a look at a ccm pcap first) |
| 00:28.43 | snuff-home | mm.. sounds reasonable.. not lookin forward to turning on pizza box again.. stupid noisy 1 ru |
| 00:30.02 | wedhorn | hehe, and while you've got it fired up, you may want to try some of those speedials, especially empty ones (or even see if they appear) |
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| 01:39.14 | coreyf1513 | anyone have a minute to look at a tiny patch to a conditional in rtp_engine.c / local_bridge_loop? http://pastebin.com/RSeTeMZP - I don't fully understand the rtp engine yet so I'm not sure if this would address a bug or create one. |
| 01:44.08 | opticron | yeah...use of binary OR there doesn't really make much sense |
| 01:45.09 | opticron | also, I might expect that the first set of logical ORed conditionals to be enclosed in an extra set of parens, but I'd have to look at more context than is available in the patch |
| 01:45.23 | opticron | are you experiencing a bug that this fixes? |
| 01:45.24 | coreyf1513 | yeah I don't fully understand this source file, I was just reading it and eclipse suggested adding parentheses.. then i saw the binary or... the patch makes that conditional the same as remote_bridge_loop |
| 01:45.33 | opticron | ok |
| 01:45.46 | opticron | interesting |
| 01:45.48 | coreyf1513 | no... IDE made me notice the code, it just looks wrong |
| 01:46.26 | coreyf1513 | if i experienced a specific issue I would know how to test my patch and I would have posted to jira in that case :) |
| 01:46.35 | opticron | I may have a look at svn blame tomorrow to see how that code got to be in the other function |
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| 01:47.23 | opticron | and if it was originally like local_bridge_loop |
| 02:04.46 | coreyf1513 | opticron: I just went to the oldest revision of rtp_engine.c I could find (/team/file/rtp_engine-mark2@183101), the conditional in remote_bridge_loop and local_bridge_loop were the same as they are now. I don't know where the code came from before that |
| 02:08.15 | opticron | it was probably main/rtp.c |
| 02:08.43 | opticron | asterisk's RTP engine wasn't always modular |
| 02:34.31 | coreyf1513 | the difference in behavior was commited to trunk r41235 |
| 02:35.11 | snuff-home | mm that's old :) |
| 03:13.42 | coreyf1513 | so i still don't know if i should post this to jira or not.. I still don't really understand this part of the code |
| 03:20.10 | wedhorn | coreyf1513: while your change seems more sensible, won't they both arrive at the same result... |
| 03:21.56 | wedhorn | I mean the or, not the brackets |
| 03:23.27 | coreyf1513 | let me do a quick test of something with that |
| 03:25.58 | wedhorn | actually, given that | has precedence over &&, they should be equivalent |
| 03:27.21 | coreyf1513 | so my patch doesn't change the result, it only makes it clearer? |
| 03:29.35 | wedhorn | yep |
| 03:31.26 | coreyf1513 | thank you for re-educating me on order of operations |
| 03:33.39 | wedhorn | no worries |
| 03:34.07 | wedhorn | snuff sent me the link only a couple weeks ago ;) |
| 03:34.20 | snuff-home | was almost bout to ask that ;) |
| 03:35.01 | coreyf1513 | I just didn't realize that any OR would beat any AND.. |
| 03:35.32 | snuff-home | remember.. its a bitwise op.. not logical test |
| 03:37.11 | snuff-home | bitwise beats logical :) |
| 03:37.39 | wedhorn | okay snuff, so why does == beat & :o |
| 03:38.19 | wedhorn | bitwise seems to be smack in the middle of the logical stuff ... |
| 03:38.21 | snuff-home | cause.. its like the rule.. 'i before e.. except after c'.. something always has to be different :) |
| 03:38.57 | wedhorn | lol, okay, will pay that one |
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| 06:18.34 | snuff-home | mm.. looks like for rpid.. need to really implement my own parser within pjsip to stop it being hackish.. |
| 06:40.59 | snuff-home | basic clid is easy.. might need to rework the outbound from msg.. make it more 'pjsip' than our own strcat |
| 06:41.44 | snuff-home | though saying that i still haven't completed anything.. writing 1 compiled piece.. but i'll get there |
| 06:45.09 | snuff-home | but my head is putting things *slowly* together |
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| 06:53.25 | schmidts | good morning |
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| 07:42.14 | oej | Morning folks |
| 07:43.31 | schmidts | good morning olle |
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| 10:51.29 | schmidts | i have a question about rtp debug. I am still working on my problem of one way audio after Hold and i can see only rtp packets sent from asterisk to both call legs, but atleast the incoming call leg still works |
| 10:52.10 | schmidts | so what i mean is, that asterisk should GOT a rtp packet from the gateway (incoming call) and then sent this to the sip client on the other side, but i didnt see this one in the rtp debug |
| 10:52.20 | schmidts | but i can hear the caller so i know its working ;) |
| 10:57.31 | schmidts | and why do i get these output when a call is on hold: [Mar 11 11:53:32] DEBUG[15846][C-00000003]: res_rtp_asterisk.c:3406 bridge_p2p_rtp_write: Remote address is null, most likely RTP has been stopped |
| 10:57.31 | schmidts | ? |
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| 12:22.58 | schmidts | when does asterisk change to P2P bridging instead of normal bridging? |
| 12:25.30 | schmidts | i am talking about something like this: http://pastebin.com/SFUhyXwb |
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| 12:25.55 | schmidts | asterisk changes to p2p bridging after the call is taken back from hold |
| 12:26.04 | file | when nothing in the core needs the media and when the codecs allow it |
| 12:27.21 | schmidts | thats bad, cause there is some kind of codec missmatch happening in this situation which make this call one way audio after hold |
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| 12:38.43 | schmidts | could someone please take a short look at this: https://issues.asterisk.org/jira/secure/attachment/46722/Holdbug-codecs.txt |
| 12:38.48 | schmidts | i have no idea why this happens |
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| 14:30.45 | file | tacos! |
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| 14:34.39 | file | caramelized onions for all! |
| 14:40.15 | jsmith | Yummy... they'll go well with my cinnamon-infused hot chocolate |
| 14:40.35 | jsmith | It's not everyday you get onions and cinnamon together :-) |
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| 14:58.47 | leifmadsen | file: I want tacos |
| 15:00.10 | file | yay tacos |
| 15:17.37 | putnopvut | I keep hearing about these so-called "tacos" but I've yet to actually see one. I think you're just making them up. |
| 15:18.10 | file | no comment. |
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| 17:31.55 | *** join/#asterisk-dev drdru (~drdru@76.77.182.56) |
| 17:32.35 | drdru | is there any way besides ChanSpy to inject audio into a channel? |
| 17:32.47 | drdru | I'm planning to do some custom module development, but I'd like to get as close as possible before I start |
| 17:32.54 | drdru | I want to capture audio from each side of the call separately (so I'm using Monitor) |
| 17:33.02 | drdru | and I want to inject audio into each side of the call independently, but I don't want to have an extra extension involved, so I don't want to use ChanSpy |
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| 17:47.12 | file | there's an internal API for it, audiohooks |
| 17:47.31 | drdru | ok |
| 17:47.49 | drdru | I basically want to process raw PCM from each side of the call separately |
| 17:48.00 | drdru | and be able to inject raw PCM into each side separately |
| 17:49.08 | file | audiohooks does signed linear, but it effectively allows you to do that |
| 17:51.12 | drdru | ok awesome |
| 17:51.54 | drdru | assuming I start with app_skel and write a custom module that uses audiohooks to read pcm and write pcm - how should I plumb asterisk to use my module from the dialplan? |
| 17:52.22 | drdru | right now I do this - Monitor() and then Dial() the desired extension upon receiving a call |
| 17:52.36 | drdru | so would I just replace Monitor() with MyApp() ? |
| 17:52.45 | file | you write a dialplan application or function which when enacted sets audiohooks up and then off it goes |
| 17:52.46 | file | yes |
| 17:52.56 | drdru | ok awesome |
| 17:53.02 | drdru | thanks! |
| 17:53.20 | drdru | I want to use dsp for silence detection |
| 17:53.27 | drdru | so I can tell when someone is talking and when they finish |
| 17:53.36 | drdru | so my module knows when to record and when to stop |
| 17:54.07 | drdru | outside the asterisk world, VAD is what I'm describing |
| 17:54.13 | drdru | but Asterisk VAD seems to be different |
| 17:54.37 | file | Asterisk doesn't support VAD. |
| 17:55.06 | drdru | ok |
| 17:55.19 | drdru | my VAD algorithm works with raw pcm buffers |
| 18:17.34 | mjordan | putnopvut: in the asterisk test suite, didn't you add a parameter to the SIPp test YAML that let's you specify a common scenario outside of that test's directory structure? |
| 18:21.48 | putnopvut | I don't believe so. I did something like that for configuration, but not for SIPp scenarios. |
| 18:22.09 | putnopvut | I made it so that you could put common config files in one location. That's what's done for the bridging test case. |
| 18:23.32 | mjordan | nuts |
| 18:24.23 | file | pistachios? |
| 18:24.33 | mjordan | are delicious |
| 18:24.42 | putnopvut | indeed |
| 18:32.40 | jsmith | Aye... the pepper flavored ones at Costco are super-addictive |
| 18:36.41 | file | I'm trying a "healthy snack" monthly box thingy this month, in past times they've included pistachios with curry and stuff |
| 18:37.13 | file | also ugh I'm in chan_sip.c reading code and it makes me want to cry |
| 18:37.47 | Marquis42 | That is the appropriate reaction to reading code in there. |
| 18:48.20 | file | let's kill it with fire!!! |
| 18:48.53 | leifmadsen | aren't you killing it with another channel driver? |
| 18:49.00 | leifmadsen | channel driver fight to the death! |
| 18:49.04 | Qwell | leedm777: Not anymore. mjordan deleted it last night. |
| 18:49.06 | Qwell | leifmadsen too |
| 18:49.10 | Qwell | err, this morning |
| 18:49.21 | leifmadsen | Qwell: you do not parse |
| 18:49.28 | file | he removed the testsuite branch momentarily |
| 18:49.33 | Qwell | oh, was it testsuite? |
| 18:49.35 | file | yes |
| 18:49.41 | leifmadsen | and only momentarily :) |
| 18:49.43 | Qwell | I just saw the subject of the email |
| 18:49.56 | leifmadsen | I was under the impression he was deleting cdr.c though |
| 18:50.15 | Qwell | leifmadsen: damn near |
| 18:51.35 | mjordan | har har |
| 18:51.51 | file | mjordan is the one man CDR TASK FORCE |
| 18:51.56 | mjordan | of doom |
| 18:51.59 | mjordan | don't forget that part. |
| 18:52.01 | file | buy him drinks ... many drinks ... |
| 18:52.21 | mjordan | Qwell: if we just delete everything, we can close a lot of bug reports |
| 18:52.23 | Qwell | wonders if mjordan knows of the vldtmf taskforce |
| 18:52.29 | mjordan | does |
| 18:52.35 | mjordan | classic picture. |
| 18:52.55 | leifmadsen | my best photoshop creation ever |
| 18:53.03 | file | someone should alter an mjordan picture to accomplish the same! |
| 18:53.08 | mjordan | hdies |
| 18:53.11 | mjordan | hides even |
| 18:53.18 | Qwell | mjordan needs to grow an epic 'stache first. |
| 18:53.27 | Qwell | OMG, I saw the most amazing mustache ever this weekend. |
| 18:53.35 | mjordan | I don't grow good facial hair |
| 18:54.24 | mjordan | Qwell: civil war style? |
| 18:54.42 | Qwell | mjordan: umm, yes, actually. Seen the guy before? |
| 18:55.00 | file | mjordan, marker. |
| 18:55.13 | putnopvut | haha, google image search for "dtmf task force" brings up that picture as the first result |
| 18:55.35 | mjordan | that's awesome |
| 18:55.50 | putnopvut | (at least for me it did) |
| 18:56.01 | putnopvut | google results can vary from person to person, from what I understand. |
| 18:56.11 | leifmadsen | putnopvut: same here |
| 18:57.05 | leifmadsen | I love how it is part of a list of dtmf related "most interesting" tags |
| 18:57.10 | leifmadsen | http://www.flickriver.com/photos/tags/dtmf/interesting/ |
| 18:58.17 | putnopvut | lol that picture sticks out like a sore thumb in that gallery |
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| 19:05.06 | maxhbp204 | Hi I have just recently installed asterisk 10.12.1 and also purchased fax for asterisk licence too and i am trying to install, i have done registeration as well and benchmark one too, and finally i have copied so file to required directory, but I am getting following errorloader.c:481 load_dynamic_module: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so:... |
| 19:05.08 | maxhbp204 | ...undefined symbol: ast_fax_tech_register |
| 19:05.12 | maxhbp204 | can anyone help me for this? |
| 19:05.52 | maxhbp204 | i have 64 centos 6.3 and also installed spandsp and working with default app_fax |
| 19:05.57 | Qwell | This channel is for help with Asterisk code. Also, you need to contact Digium technical support for this type of issue. |
| 19:06.10 | Qwell | However, to answer your question, it's because you don't have res_fax loaded. |
| 19:06.29 | maxhbp204 | ok thanks let me check that way |
| 19:06.32 | maxhbp204 | thanks you so much |
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| 19:30.00 | Qwell | leedm777: In your configure.ac change, it should probably say "JSON support not found" |
| 19:30.09 | Qwell | </pedantry> |
| 19:30.13 | file | JaSON support not found |
| 19:30.51 | leedm777 | Qwell: thx |
| 19:40.19 | snuff-home | file.. might bug u later bout pjsip stuff ;) |
| 19:40.28 | file | k |
| 19:40.35 | snuff-home | heads off to train |
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