Re: [asterisk-dev] Re: Help with 240 samples on frames readfromchan_iax
On Mon, Nov 06, 2006 at 06:59:44PM -0600, Moises Silva wrote: MS Hum, it seems to work with slinear, ulaw, alaw and gsm, but not with MS iLBC, it complains about ilbc frame not being multiple of 50, so i MS tried to make it happy, but still misses some translations from MS slinear to iLBC and the sound is choppy. What makes iLBC so special MS any way? Are you get app_conference work with 1.4? -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ signature.asc Description: Digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk Usersconf
On Sat, Aug 05, 2006 at 06:45:32PM -0500, Mark Spencer wrote: MS http://svn.digium.com/svn/asterisk/team/markster/usersconf This feature make multi-protocol user access control more easy. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Conference Call Re-scheduled
On Fri, May 26, 2006 at 03:11:02PM -0400, Russell Bryant wrote: RB The weekly developer conference call has been rescheduled to Wednesday, RB May 31st, at 10 AM CDT (GMT -5), instead of Tuesday. RB Please note that May 31st is the last day that new features can be RB merged into the trunk to be present for the upcoming 1.4 release. We RB will likely be talking about any last minute additions that need to be RB made that day. What about issue #6725? -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ signature.asc Description: Digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Conference Call Re-scheduled
On Sat, May 27, 2006 at 10:10:18AM -0500, Tilghman Lesher wrote: What about issue #6725? TL Given that your patches are for oej's branch, specifically, you TL need to talk to oej. This patch for trunk. oej help me with just copy'n'paste my patches to team/oej/codecnegotiation. Sometimes I create diff with team/oej/codecnegotiation and trunk, and upload it to #6725. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] monitoring a call and media path
On Mon, Dec 05, 2005 at 07:37:55PM -0600, Kevin P. Fleming wrote: I was thinking of hacking things a bit to allow my asterisk to stay out of the media path in the above case, but figured it couldn't hurt to post a quick sanity check here. Anyone see any problems? KPF This is certainly possible, but Asterisk currently assumes that if it is KPF not in the media path, it also won't be able to receive DTMF frames. KPF However, if you are using SIP INFO for DTMF signaling, then it should KPF 'just work', since when Asterisk sees the appropriate DTMF frames it KPF will cause the bridge to 'break' and bring the media path back. With last changes to IAX2, can this be fixed for SIP? -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Corydon76
On Tue, May 02, 2006 at 04:15:04PM -0500, Tilghman Lesher wrote: TL No, the only things that result in bad karma are: 1) opening bugs that TL are duplicates of existing bugs, There is one bug, that _I_ think is not duplicate, but _you_ think are duplicate. If there is one bug report with patch that fix it. And there is another enchantment, that bigger then first patch, and fix first bug also, but has different purpose, is this duplicate bug? I think, that bugs with different enchantment patches from different authors is different bugs. You think that it's duplicates. Who are right? I'm not find answer in bug guidlines. TL 2) opening bugs as MAJOR when they TL are not in the 1.2 series, 3) opening bugs as CRASH when they do not TL manifest themselves as a crash. These are all contrary to the bug TL guidelines, which are freely available on the bugtracker. There are TL other reasons as well, but they are all for actions that are agreed upon TL beforehand. I have no power at this time to create new reasons for TL negative karma on the bugtracker. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Corydon76 Issue Deleted: 0006925 ..... ( Alex ) Corydon76 Issue Deleted: 0006920 ( Alex Smirnov )
On Wed, May 03, 2006 at 01:34:49PM +0400, Alex Smirnov ( DigitalXXI ) wrote: ASD 1) Absolutely agree, it's not a asterisk task to code audio files in ASD different codecs, At now Astersisk _do_ this. ASD easy to convert format is more practical , than ASD amount of different codded files. Only if you have supercomputer with one client :) I doesn't know any codec that can be easy converted to G.729, G.723, speex, iLBC and GSM. Do you know that you need to avoid all codec conversion in high-volume installations, if you can? And if you can have voice prompt in all supported formats, this save your money? ASD 2) What is REAL need to support such codding in * , when u can use ASD any software to convert it to whatever u like ? ASD ( something like split rule :) ) I not understand your idea. ASD 3) and just curios - any ex. for lib usability , except commercial ASD software ? With commercial software no lib usability, because this lib would be GPL'ed for all but Digium clients, that buy license. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Corydon76 Issue Deleted: 0006925, 04-28-06 17:49 Corydon76 Issue Deleted: 0006920
On Sun, Apr 30, 2006 at 04:28:41PM -0500, Tilghman Lesher wrote: TL In the future, to avoid these types of problems, I recommend that if TL one of your bugs is closed, that you seek out either the person who TL closed them or another bug marshal, either via email or via IRC, to TL discuss why you think a bug should be reconsidered. The benefit of TL IRC is that the discussion can take place with multiple developers at TL the same time, so if your patch is still rejected, you can find out TL quickly and move on. It needs that bug marshals email would added to bug tracker docs. At now I hasn't emails all bug marshals. TL In reality, what happened was I applied a fix in SVN for the problem TL that was reported, but the fix was different from the patch that Denis TL posted. Denis subsequently informed me that the extra code in the TL patch that I applied was unnecessary, and I corrected the problem. TL Other people reading this thread might conclude from Denis' post that TL I obstinately refused to read the manpage for fseek(), and that is TL simply not true. Ok. Problem was because I see one, you see another situation. I post bug. Bug was right, patch was right. You say that you apply another patch. Ok. I see in svn-commit that your patch worse and reopen bug. You _silently_ close. I reopen with text from manpage. You _silentlty_ close. What I can think? Only that you doesn't want to read my comments. I post message to asterisk-dev, because I doesn't like bloating code, and think that it's bad and you are doesn't right here. After that I see in svn-commit that you fix it applying my patch. All bug marshals can apply or reject patch, it's ok. But as contributor I need to _know_ was patch applied? Was not? Why it not applied? It's my bug, or bug marshal's? We need policy that can solve this conflicts. If you say nothing and close bug it's conflict situation. If you say you are lamer, and doesn't know ... it _can_ be conflict or no, dependend by contributor. I would not complain for this answer, I need information, not conflicts. If you say I doesn't want apply this patch, because it's ok for all. At now we have first variant, that for me, and some others, looks bad. TL The problem many patches experience is that they are fairly complex, I know it, and break all my internal patches to small independent parts. TL and therefore, even those with commit access do not feel comfortable TL committing them, especially if they do not have time to exhaustively TL test the patch. What we need (and what you need to solicit, if you TL are dedicated to having the patch committed) is to find more people TL who are willing to apply and test the patch. In some ways, oej's TL branch has gone a long way towards making patches easy to test, and TL this is probably the way forward. In order to get your patch into TL oej's branch, though, you need to contact him directly. I contact with oej directly, and he help me. But oej hasn't enought time for all this work. Why Corydon76 can't do his work without conflicting with other developers? Why he close bugs without comment? TL Generally, we close bugs without a comment after first asking, Hey, TL we pointed out a problem, are you going to fix it? and then waiting TL 3 days for a response. In fact, for most bugs which are closed this TL way, more than 30 days pass without comment, and we have to conclude TL that the reporter has lost interest. It's not closing without a comment. Closing without a comment was when I have conflict about fseek. I think that today logging API in asterisk is a crap. Casper try to fix it. Instead of helping him for choose right way, his bugs was closed. Why? If you give me specific bug numbers I can try to find out. TL Bug 6889. What we have here is a replacement of 2 lines of code with TL 1 line of code. It's not complex, and it's not easy to make a TL mistake coding this. All this substitution does is to create wholly TL another macro that must be learned in order to code to the coding TL guidelines. We have other macros, for fairly complex operations, such TL as linked list maintenance, in which it is fairly easy to accidently TL code something wrong. Macros are for making coding easier, and I TL don't believe that these macros make coding any easier. This is why I TL rejected his patch. TL In this case, Casper then made a plea for his patch, on this list, and TL Kevin intervened. Please note that the bug remains open to this day. Because I start to support this patch. Without patch like this I and Casper need to know -- is this interesting? And _why_ it's not commited? What doesn't right with this patch, and how can it be rewrited more clean? I has time for this work. But I hasn't time for flame. I need two things -- strict policy, and info about what patch would be commited, if some issues would fixes, and what patch would not be commited, and I doesn't need to spend my time for supporting this
Re: [asterisk-dev] Corydon76 Issue Deleted: 0006925, 04-28-06 17:49 Corydon76 Issue Deleted: 0006920
On Mon, May 01, 2006 at 07:11:47AM -0500, Kevin P. Fleming wrote: I need two things -- strict policy, and info about what patch would be commited, if some issues would fixes, and what patch would not be commited, and I doesn't need to spend my time for supporting this patches. KPF There is no answer to that question before the patch exists. We can KPF certainly tell you whether your idea seems like a good one and is likely KPF to be something we would accept, but until the actual patch appears KPF nobody can make any decisions about merging. I think that discussion about idea need to be before writing patch :) Ok, I try to write this patch. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Corydon76 Issue Deleted: 0006925, 04-28-06 17:49 Corydon76 Issue Deleted: 0006920
On Mon, May 01, 2006 at 06:48:11AM -0500, Kevin P. Fleming wrote: This moving helps to minimize duplicate code and improve API stability without affecting development speed. KPF This is incorrect. As soon as 'struct ast_channel' is part of the KPF library's API (as it would have to be), then we cannot make changes to KPF the channel structure without having to increase library versions and KPF modifying all the modules to require that version. This will take KPF significant amounts of time. Some modules doesn't need to work with this struct directly. KPF I don't see any value in having modules that are usable across Asterisk KPF versions in binary form; this is open source software, and nearly all KPF modules available for it should only need source-level compatibility, KPF not binary compatibility. It need for linux distributions, for updating third part modules independent with asterisk. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] AST_FORMAT_AMR anyone?
On Wed, Apr 26, 2006 at 01:23:28PM -0500, Kevin P. Fleming wrote: KPF There are tons of codecs that we could add this 'token' support for, and KPF the current methods of defining them and selecting them have finite KPF limits as to the number of formats we can support. KPF I don't think it is likely that we will just start adding 'format types' KPF for formats that will never be in the tree. Asterisk can passthrough some codecs, like G.722 in test-this-branch. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] AST_FORMAT_AMR anyone?
On Wed, Apr 26, 2006 at 04:49:58PM +0100, Tim Panton wrote: TP Any chance of getting a constant allocated in include/asterisk/ TP frame.h for AMR (wide and narrow?) It would be very nice, if AST_FORMAT_G722 constant also would be added to frame.h -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] iLBC packet loss concealment (was: code-cleanup concerns)
On Mon, Apr 17, 2006 at 11:19:19PM +0800, Steve Underwood wrote: Transcoding between variants can appear, if we have recordings (for voice prompts) in ultra wideband version (32k), and would play to phones, that support only low-rate codec version (16k). For speex this transcoding cheaper, then uwb-sln-wb transcoding. SU True. That is a valid situation. However, the computational cost is not SU the major issue. The quality of the converted audio will be far superior SU in a case like the one you mention if its done directly. Decoding and SU recoding low bit rate codecs is really bad news for quality. In the SU Speex case this is avoided completely when a direct rate conversion occurs. Yes. Why transcode_via_sln was added, and used by default? -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] iLBC packet loss concealment
On Mon, Apr 17, 2006 at 10:46:15AM -0500, Kevin P. Fleming wrote: KPF However... this option will cause any 'direct conversion' that is KPF possible to be avoided. There are not any current 'direct conversions' KPF available except for alaw - ulaw, though. I will think today about KPF changing this option to only take effect when the translation path KPF already goes via SLIN, so that 'direct conversions' that may be KPF available are not skipped. If it be done, that option would not needed, and it be very nice thing. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] iLBC packet loss concealment
On Mon, Apr 17, 2006 at 11:12:34AM -0500, Kevin P. Fleming wrote: If it be done, that option would not needed, and it be very nice thing. KPF Well, it will be nice if and when any direct conversions become KPF available. Right now the only one available in the Asterisk standard KPF code base is alaw/ulaw, and going via SLIN doesn't make a huge KPF difference there. I think about adding WB/UWB speex versions support. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] func_module update
Update to func_module.c in attached.
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--- ./func_module.c.orig2006-04-17 22:15:07 +0400
+++ ./func_module.c 2006-04-17 22:39:52 +0400
@@ -64,29 +64,25 @@
Returns \1\ if module exists in memory, otherwise \0\.\n,
};
-static char *tdesc = Checks if Asterisk module is loaded in memory;
-int unload_module(void)
+static int unload_module(void *mod)
{
return ast_custom_function_unregister(ifmodule_function);
}
-int load_module(void)
+static int load_module(void *mod)
{
return ast_custom_function_register(ifmodule_function);
}
-const char *description(void)
-{
- return tdesc;
-}
-
-int usecount(void)
+static const char *description(void)
{
- return 0;
+ return Checks if Asterisk module is loaded in memory;
}
-const char *key()
+static const char *key(void)
{
return ASTERISK_GPL_KEY;
}
+
+STD_MOD(MOD_1 | NO_USECOUNT, NULL, NULL, NULL);
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Re: [asterisk-dev] code-cleanup concerns
On Fri, Apr 14, 2006 at 04:28:23PM -0700, Brian Degenhardt wrote:
BD I'm a bit uncomfortable about some of these recent commits to trunk.
BD While I'm all in favor for code cleanup, it should be taken in account
BD that changing things arbitrarily will break many patches sitting in
BD mantis.
BD Unless somebody can explain why for(;;) is inferior to do{}while(1), or
BD why removing curly braces on one-line if statements is worth the trouble
BD of patch maintainers everywhere, I think this just does more harm than good.
I support _many_ patches from mantis local. svn is very powerfull tool,
that can give you ability to update this patches with minimal time
requirements.
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Re: [asterisk-dev] Asterisk servers as UDP amplifier
On Sat, Apr 15, 2006 at 10:45:30PM +1000, Edwin Groothuis wrote: EG Doing this, my smallest packet was 85 bytes, giving me an EG answer of 289 bytes and thus an amplification of less than 4. Can you post your patch to bugtracker? -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Postgres Realtime driver
On Tue, Apr 04, 2006 at 05:54:10PM +0200, Olle E Johansson wrote: OEJ - What problem does it solve that can not be handled by using ODBC to OEJ postgreSQL? Speed. Stability. OEJ - This driver is based on the mysql realtime driver, which is known OEJ to cause a lot of OEJ problems. Will this happen here too? Any testing experience? OEJ - Are there enough developers out there to maintain this driver? We OEJ don't want OEJ to get stuck with code that is not maintained... I'm not test this code and wait for end work for it. If it is needed -- I can test it, and start to code review. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Test my cool new sexy test-branch! * Extra Asterisk-bonus awards this week
On Thu, Feb 23, 2006 at 10:17:11AM +0100, Olle E Johansson wrote: #5090 OEJ That patch does not apply cleanly to trunk, or at least not to OEJ this version of trunk... I attached patch, updated to trunk revision 10890. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ asterisk.t38.pass.patch.gz Description: GNU Zip compressed data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Best sound format / sample rate for french translated Asterisk sounds?
On Thu, Feb 23, 2006 at 12:37:31PM -0500, Rusty Dekema wrote: RD The best-quality codecs that Asterisk currently supports are ulaw and RD alaw. Using one of those will result in the best sound quality. _best_ quality would be with slinear. -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] get register info from peer section in sip.conf ?
On Wed, Nov 23, 2005 at 02:08:51PM +0100, Olle E. Johansson wrote: OEJ I had a patch that did the opposite, and that's where we are going. OEJ I want to have register=yes within a peer section and remove the OEJ register= statement from the [general] section. Is it patch merged to test-this-branch? -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] ENUM changes: discussion
On Wed, Feb 15, 2006 at 11:57:11AM +0100, Klaus Darilion wrote: KD Just because of having e164.arpa the default one does not restrict the KD usage to other trees. Any tree is supported. But if there is a default KD tree, which one should it be? e164.org? e164.info?... IMO, the default KD tree should be the standard one. I think, that if it is _one_ variable, it can be in asterisk.conf -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] Why asterisk _binary_ links with ssl?
Why asterisk binary links with ssl, where only res_crypto must be linked with it? I create RPM-distribution for Asterisk and don't wont that asterisk package requires openssl, but crypto can be used when subpackage with res_crypto installed. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
