| 01:04.53 | *** join/#asterisk-dev shido6 (i=shido6@74-130-126-198.dhcp.insightbb.com) |
| 01:16.47 | *** join/#asterisk-dev dlynes_home (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 01:38.27 | *** join/#asterisk-dev ksteward (n=ksteward@71.174.94.169) |
| 02:06.25 | *** join/#asterisk-dev DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 02:13.51 | DarkRift | I've seen once a websiet that had the asterisk soruce code online with the explanation of all it's functions and in what file they are, anyone know what I'm talking about ? I can't seem to find it anymore |
| 02:20.13 | snuff-away | yes |
| 02:20.19 | snuff-away | would be the 'doxygen' |
| 02:20.57 | snuff-away | DarkRift: http://www.asterisk.org/doxygen/trunk/ |
| 02:23.13 | DarkRift | Yeah, that's it thanks a bunch |
| 02:37.35 | *** join/#asterisk-dev ManxPower (n=manxpowe@36.sub-70-197-245.myvzw.com) |
| 03:01.56 | *** join/#asterisk-dev moy (n=moyhu@189.169.50.222) |
| 04:06.17 | *** part/#asterisk-dev ksteward (n=ksteward@71.174.94.169) |
| 04:30.29 | moy | somebody still around? |
| 04:48.02 | tzafrir_home | maybe |
| 05:12.15 | moy | tzafrir_home: just wanted to confirm if Action: Redirect is broken on trunk ... or may be I screwed up something |
| 05:13.10 | moy | I was testing http://bugs.digium.com/view.php?id=11282 on ast_do_masquerade() conditions to see how it behaves, but Redirect is not working even when not in that new app |
| 05:17.03 | tzafrir_home | sorry, can't really confirm or deny... |
| 05:32.17 | moy | np |
| 05:33.09 | *** join/#asterisk-dev DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 05:36.08 | jameswf-home | jbot: wow |
| 05:36.09 | jbot | I have no life | Lets go raid! |
| 06:16.16 | *** join/#asterisk-dev Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 06:21.35 | Zuchmir | there are a few features i'd like added to asterisk, where's the best place where i can get someone to develop it at a descent cost? |
| 06:22.13 | Corydon76-dig | Two places... the asterisk-biz list or digium custom development |
| 06:28.14 | Corydon76-dig | Or, you tell us what you're interested in, and if it's something that interests me, I'll develop it for free |
| 06:28.35 | Zuchmir | Corydon76-dig: i should subscribe, post a message, and wait for response? is that what you mean? |
| 06:28.59 | Corydon76-dig | Zuchmir: yes |
| 06:29.22 | Corydon76-dig | Oh, and there's a bounty page on the Wiki |
| 06:29.48 | Corydon76-dig | What is it that you'd like to see added? |
| 06:30.27 | Zuchmir | Corydon76-dig: I need ControlPlayback() to have a speedup/slow down option |
| 06:30.50 | Zuchmir | similar to what audacity has |
| 06:31.53 | Corydon76-dig | So ff/rew isn't what you need? |
| 06:32.20 | Zuchmir | i have some lectures which are 30min+ and some of the lectureres tend to speak slow, giving the caller the ability to speed up the sound would be great |
| 06:32.31 | Zuchmir | i need ff/rewind as well |
| 06:32.41 | Corydon76-dig | Do you want speedup that affects pitch? |
| 06:32.53 | Corydon76-dig | or speedup that doesn't affect pitch? |
| 06:33.18 | Zuchmir | at the moment pitch doesn't bother me (I have to at least matchg an existing PBX system which has speedup/slowdown) |
| 06:34.08 | Zuchmir | i know that pitch can also be done, but at the moment it's not as important |
| 06:34.34 | Corydon76-dig | Well, the natural method of speedup/slowdown affects pitch |
| 06:35.07 | Corydon76-dig | Not affecting pitch is more difficult |
| 06:35.11 | Zuchmir | if i was more comfortable with the asterisk src code, i probably would do it myself, but i'm just learning now |
| 06:35.38 | Zuchmir | i understand that, that's why i say i don't care, so long as the speech moves faster |
| 06:36.37 | Zuchmir | the basic concept is read 100 bytes resample it to 80 bytes of sound |
| 06:38.04 | Zuchmir | BTW audacity (which is open source) has both features, but i only need slow/fast |
| 06:41.32 | Corydon76-dig | What's interesting is that I've done some work with a library that does exactly that, but I haven't gotten the audio quality right yet |
| 06:43.30 | Zuchmir | ... sox also does this knid of resampling, it's just that i'm not comfortable "incorporating it" into asterisk |
| 06:44.14 | Corydon76-dig | Yeah, I know what you mean |
| 06:44.32 | Zuchmir | iwould guess it should take no more than 3 hoursto fully implement (for someone confident with the asterisk source code) |
| 06:44.36 | *** join/#asterisk-dev steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:44.39 | Corydon76-dig | The current methods that ControlPlayback uses are all core functionality... |
| 06:44.54 | Corydon76-dig | but resampling currently isn't. |
| 06:45.43 | Zuchmir | i contacted "digium custom development" they want $8500+ that's outragous for this knida feature |
| 06:45.53 | Corydon76-dig | Audio filters in general aren't (yet) |
| 06:46.29 | snuff-away | mm isn't there some type of audio filter app out there for *.. remember seeing somethin on voip-info |
| 06:47.17 | Zuchmir | yeah, but i would rewrite "ast_control_streamfile" to deal with resample |
| 06:47.21 | Corydon76-dig | There is one for voice distortion, I think |
| 06:47.39 | snuff-away | would be what i'm thinkin of Corydon76-dig ;) |
| 06:47.52 | Zuchmir | even if i can take every 4th sample and drop it, it would also do wonders |
| 06:48.30 | Zuchmir | ie instead of 8000 samples per second you would get 6000 |
| 06:48.40 | Corydon76-dig | To do it right, I'm not sure that the pricetag custom dev put on it is all that wrong |
| 06:49.29 | Corydon76-dig | Dropping every 4th sample leads to audio artifacts |
| 06:50.04 | Corydon76-dig | When I dropped every other sample, the audio quality was signficantly lower |
| 06:50.11 | Zuchmir | ok, but as i said, using ast_control_streamfile() as a starting point, adding the functionality from sox for resampling, shouldn't take more thatn a few hours |
| 06:50.32 | Corydon76-dig | Yeah, but trying to use sox doesn't scale |
| 06:51.06 | Corydon76-dig | It's gotta be native |
| 06:51.18 | Zuchmir | i don't mean system(sox ...), i mean pulll in the source code from sox (or audacity) |
| 06:51.37 | Corydon76-dig | Can't do that without a licensing issue |
| 06:52.15 | Corydon76-dig | unless the source is LGPL |
| 06:53.09 | Zuchmir | under GPL you can't copy parts from one prog and paste into another prog? |
| 06:53.37 | Corydon76-dig | Asterisk has to be dual-licensable |
| 06:54.44 | Zuchmir | SoX (Sound eXchange), under LGPL at http://sox.sourceforge.net/ |
| 06:54.44 | Zuchmir | license |
| 06:55.13 | Zuchmir | ...according to: http://devel.zs4.org/tabencode_osx.html and http://www.nabble.com/SoX-14.0.0.-Released-t4423182.html |
| 06:55.54 | Corydon76-dig | Yeah, the only problem with their API is that it's all file-based |
| 06:56.08 | Corydon76-dig | You can't re-render audio data in memory on the fly |
| 06:56.38 | Corydon76-dig | So if you're listening to something and you speed up/slow down the recording, you have to completely re-render the whole file |
| 06:57.15 | Corydon76-dig | and then recalculate position, based upon where in the file you were vs the same timestamp in the new file |
| 06:57.47 | Zuchmir | can't u simply use fdopen() to get a file handle and feed your stuff through that? |
| 06:58.16 | Corydon76-dig | IIRC, no |
| 06:58.57 | Corydon76-dig | It's more insidious... input filename, output filename |
| 07:05.00 | *** join/#asterisk-dev mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 07:05.00 | *** mode/#asterisk-dev [+o mog] by ChanServ |
| 07:05.51 | Zuchmir | sox_gsmread() is probably all that's needed |
| 07:06.05 | Zuchmir | static sox_size_t sox_gsmread(sox_format_t * ft, sox_ssample_t *buf, sox_size_t samp) |
| 07:06.28 | Corydon76-dig | I don't see that in st.h |
| 07:06.53 | Zuchmir | i haven'yt played with the sox code eaither, but my guess would be that's the function needed |
| 07:07.04 | Zuchmir | sox-1.14.0/src/gsm.c |
| 07:07.35 | Corydon76-dig | What if the file format isn't gsm? |
| 07:09.09 | Corydon76-dig | Well, I'm heading to bed |
| 07:13.06 | Zuchmir | sox_read() |
| 07:13.21 | Zuchmir | ... is the generic form |
| 07:13.32 | Zuchmir | ok, thanks, good night! |
| 07:24.44 | *** join/#asterisk-dev IgorG (n=FeedomPa@195.162.32.126) |
| 07:45.58 | *** join/#asterisk-dev tzafrir (n=tzafrir@62.90.10.53) |
| 07:45.58 | *** mode/#asterisk-dev [+o tzafrir] by ChanServ |
| 07:53.15 | *** join/#asterisk-dev pnlarsson (n=pnlarsso@c83-248-96-159.bredband.comhem.se) |
| 08:17.48 | *** join/#asterisk-dev qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 09:28.38 | *** join/#asterisk-dev sergeus (n=asterisk@77.244.64.99) |
| 09:34.16 | sergeus | putnopvut, ping |
| 10:58.31 | *** join/#asterisk-dev Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 11:26.57 | *** join/#asterisk-dev _ys (n=yuri@80.70.236.69) |
| 11:28.23 | _ys | hi |
| 11:28.52 | _ys | r89392 break compilation on FreeBSD: usr/include/malloc.h:3:2: #error "<malloc.h> has been replaced by <stdlib.h>" |
| 13:03.53 | *** join/#asterisk-dev mashroom_ (n=tasty@89-139-158-24.bb.netvision.net.il) |
| 13:59.09 | *** join/#asterisk-dev _ys (n=yuri@80.70.236.69) |
| 14:20.27 | *** join/#asterisk-dev DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 14:23.06 | *** join/#asterisk-dev puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:23.33 | puzzled | hi |
| 16:04.20 | *** join/#asterisk-dev DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 16:04.36 | *** join/#asterisk-dev Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 16:05.12 | Zuchmir | which is the api/function that actually deals with the keys pressed? |
| 16:05.36 | Corydon76-dig | Zuchmir: ast_read |
| 16:05.54 | Corydon76-dig | Keys pressed are sent as DTMF events |
| 16:07.03 | Corydon76-dig | In 1.4, DTMF are actually 2 events, a start and a stop |
| 16:11.24 | *** join/#asterisk-dev oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:11.24 | *** mode/#asterisk-dev [+o oej] by ChanServ |
| 16:24.21 | Zuchmir | Corydon76-dig: thanks |
| 16:26.39 | Zuchmir | i guess i have a fair bit of studying to understand the full call structure of * ... is there a doc somewhere that gives the overview of the structure? |
| 16:28.22 | Corydon76-dig | Newp |
| 16:34.45 | oej | http://www.asterisk.org/doxygen is a good start |
| 16:38.13 | *** join/#asterisk-dev _ys (n=yuri@80.70.236.69) |
| 16:39.19 | _ys | rh |
| 16:39.20 | _ys | eh |
| 16:54.29 | *** join/#asterisk-dev ManxPower (n=manxpowe@151.sub-70-196-184.myvzw.com) |
| 17:06.46 | *** join/#asterisk-dev Juggie (n=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 17:26.21 | _ys | FreeBSD trunk compilation is still broken |
| 17:26.28 | file | I know. |
| 17:26.41 | *** join/#asterisk-dev oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:26.41 | *** mode/#asterisk-dev [+o oej] by ChanServ |
| 17:38.30 | puzzled | with my limited C skills I'm trying to port a 1.2 app to 1.4. I get this error: error: 'struct ast_channel' has no member named 'type' Anyone have a hint where I should look to solve this? |
| 17:39.09 | Corydon76-dig | chan->type |
| 17:39.17 | puzzled | ta |
| 17:49.21 | file | all better. |
| 17:49.45 | _ys | thanks file :) |
| 17:51.18 | *** join/#asterisk-dev fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:53.55 | *** join/#asterisk-dev mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 17:53.55 | *** mode/#asterisk-dev [+o mog] by ChanServ |
| 18:56.45 | *** join/#asterisk-dev noway909 (n=vvv@scandic887.host.songnetworks.se) |
| 18:59.02 | noway909 | On doing sip show peers, i get a list of some phones and there IPs |
| 19:00.04 | noway909 | i want to know what is the criteria behind on adding these phones to the list. e.g. i have a wifi phone that is listed in registry but sometimes it appears in the sip show peers and then i can call it but other times it is not listed under sip show peers and i cannot call it but yet i can ping it at that point in time. |
| 20:44.21 | *** join/#asterisk-dev blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:44.21 | *** mode/#asterisk-dev [+o blitzrage] by ChanServ |
| 21:00.38 | snuff-work | mmm.. got interesting problem |
| 21:01.36 | snuff-work | I updated our prod * server.. and some sips have static ip's defined in realtime |
| 21:01.47 | snuff-work | to 1.4.14 |
| 21:03.57 | snuff-work | when i load peer via sip show peer 3000 load |
| 21:04.05 | snuff-work | the information hadn't changed |
| 21:05.12 | snuff-work | aka the info before was 10.0.2.1.. updated in db to 10.0.3.1 |
| 21:05.29 | snuff-work | sip show peer 3000 load shows -> 10.0.2.1 |
| 21:06.50 | snuff-work | yet if the sip peer calls its correctly identifed as being 3000.. |
| 21:14.39 | snuff-work | any ideas ppl? |
| 21:16.12 | *** join/#asterisk-dev fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:19.58 | snuff-work | even tried restarting * to change the IP |
| 21:22.43 | *** join/#asterisk-dev Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 21:37.54 | *** join/#asterisk-dev ta^3 (n=tacvbo@201.128.232.238) |
| 22:24.42 | snuff-work | nm.. found my issue :( |
| 22:28.25 | *** join/#asterisk-dev grimsy (n=chatzill@60-241-200-26.static.tpgi.com.au) |
| 22:44.48 | *** join/#asterisk-dev anonymouz666 (n=anonymou@200.218.193.6) |
| 22:45.11 | *** join/#asterisk-dev bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 22:59.10 | *** join/#asterisk-dev eliel (n=eliel@151-202-114-200.fibertel.com.ar) |
| 23:55.19 | *** join/#asterisk-dev ManxPower (n=manxpowe@209.16.72.135) |