[asterisk-commits] juggie: branch group/NoLossCDR-Redux2 r113925 - /team/group/NoLossCDR-Redux2/...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Apr 9 15:23:38 CDT 2008 

Author: juggie
Date: Wed Apr  9 15:23:38 2008
New Revision: 113925

URL: http://svn.digium.com/view/asterisk?view=rev&rev=113925
Log:
Merged revisions 110020,110023,110036,110084,110087,110132,110161,110164,110211,110237,110268,110270,110272,110303,110337,110339,110396,110444 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

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r110020 | file | 2008-03-19 14:25:33 -0400 (Wed, 19 Mar 2008) | 14 lines

Merged revisions 110019 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines

Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue #11429)
Reported by: sperreault
Patches:
      11429-frametype.diff uploaded by qwell (license 4)

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r110023 | russell | 2008-03-19 14:57:16 -0400 (Wed, 19 Mar 2008) | 2 lines

remove svnmerge-blocked property that is not supposed to be here

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r110036 | file | 2008-03-19 15:13:39 -0400 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110035 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines

Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue #11663)
Reported by: junky

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r110084 | mmichelson | 2008-03-19 16:34:13 -0400 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110083 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines

Add a missing unlock in the case that memory allocation fails in app_chanspy.
Thanks to Russell for confirming that this was an issue.


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r110087 | jpeeler | 2008-03-19 17:05:24 -0400 (Wed, 19 Mar 2008) | 2 lines

This change adds DNS manager support for registrations not referencing a peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries.

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r110132 | qwell | 2008-03-19 17:56:15 -0400 (Wed, 19 Mar 2008) | 1 line

Rename very poorly named function to reflect what it actually does.  This was causing quite a bit of confusion for me...
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r110161 | qwell | 2008-03-19 18:25:34 -0400 (Wed, 19 Mar 2008) | 5 lines

Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading.

Early cleanup for issue #11968

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r110164 | russell | 2008-03-19 18:58:33 -0400 (Wed, 19 Mar 2008) | 13 lines

Merged revisions 110163 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines

Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected.

(closes issue #11990, reported by anakaoka, patched by me)

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r110211 | tilghman | 2008-03-19 23:14:59 -0400 (Wed, 19 Mar 2008) | 2 lines

Fix recent trunk breakage

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r110237 | tilghman | 2008-03-20 01:06:12 -0400 (Thu, 20 Mar 2008) | 5 lines

Upgrade the sounds version; add several directory enhancements:
	1) Number of digits to enter can now be configured
	2) The digits can now match on both first AND last name, instead of only one or the other
(Closes issue #7151)

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r110268 | russell | 2008-03-20 13:41:22 -0400 (Thu, 20 Mar 2008) | 27 lines

Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.

(issue #12164, reported by milazzo and jsmith, patches by me)

res/res_musiconhold.c:
 - I moved a single line so that the sample queue update happened before
   ast_write().  The reason that this was a bug is that the G.722 frame
   originally says it has 320 samples in it (which is correct).  However,
   when the frame is written to a channel that uses RTP, main/rtp.c modifies
   the frame to cut the number of samples in half before it sends it on
   the wire.  This is to account for the stupid incorrect G.722 spec that
   makes it so we have to lie about the number of samples with RTP.  I should
   probably go and re-work the RTP code so it doesn't modify the frame so
   that a bug like this won't happen in the future.  However, this change to
   MOH is harmless.

main/channel.c:
 - I made two fixes in regards to generator timing.  Generators use samples
   for timing.  However, this code assumed 8 kHz samples.  In one case, it was
   a hard coded 160 samples, that is now written as the sample rate / 50.  The
   other place was dealing with timing a generator based on frames coming from
   the other direction.  However, that would have only worked if the sample
   rates for the formats in both directions were the same.  The code now takes
   into account that the sample rates may differ, and scales the generator
   samples accordingly.

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r110270 | russell | 2008-03-20 13:45:29 -0400 (Thu, 20 Mar 2008) | 2 lines

Remove astobj.h from some places where it wasn't needed

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r110272 | mmichelson | 2008-03-20 14:01:36 -0400 (Thu, 20 Mar 2008) | 3 lines

Add missing unlock


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r110303 | russell | 2008-03-20 16:08:26 -0400 (Thu, 20 Mar 2008) | 8 lines

Fix a bug when using zaptel timing for playing back files that have a sample rate
other than 8 kHz.  The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame.  However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.

(another part of issue #12164, reported by milazzo and jsmith, patch by me)

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r110337 | russell | 2008-03-20 17:55:50 -0400 (Thu, 20 Mar 2008) | 22 lines

Merged revisions 110336 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines

Merged revisions 110335 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines

Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix.  The dnsmgr is not appropriate here.  The dnsmgr takes a pointer to an address
structure that a background thread continuously updates.  However, in these cases,
a stack variable was passed.  That means that the dnsmgr thread would be continuously
writing to bogus memory.

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r110339 | russell | 2008-03-20 18:02:20 -0400 (Thu, 20 Mar 2008) | 3 lines

Use the correct buffer for g722tolin16_sample.  This shouldn't have caused any
problems, but Qwell noticed the typo here.

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r110396 | russell | 2008-03-20 19:14:13 -0400 (Thu, 20 Mar 2008) | 17 lines

Merged revisions 110395 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue #12266, reported by dimas)

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r110444 | tilghman | 2008-03-20 21:44:38 -0400 (Thu, 20 Mar 2008) | 2 lines

Add note of the added Directory options, from commit 110237 (closes issue #7151)

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Modified:
    team/group/NoLossCDR-Redux2/channels/chan_h323.c
    team/group/NoLossCDR-Redux2/channels/chan_usbradio.c
    team/group/NoLossCDR-Redux2/channels/misdn_config.c

Modified: team/group/NoLossCDR-Redux2/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/team/group/NoLossCDR-Redux2/channels/chan_h323.c?view=diff&rev=113925&r1=113924&r2=113925
==============================================================================
--- team/group/NoLossCDR-Redux2/channels/chan_h323.c (original)
+++ team/group/NoLossCDR-Redux2/channels/chan_h323.c Wed Apr  9 15:23:38 2008
@@ -254,10 +254,7 @@
 	.write = oh323_write,
 	.indicate = oh323_indicate,
 	.fixup = oh323_fixup,
-	/* disable, for now */
-#if 0
 	.bridge = ast_rtp_bridge,
-#endif
 };
 
 static const char* redirectingreason2str(int redirectingreason)
@@ -3092,10 +3089,12 @@
 		return AST_RTP_GET_FAILED;
 
 	ast_mutex_lock(&pvt->lock);
-	if (pvt->rtp && pvt->options.bridge) {
-		*rtp = pvt->rtp;
+	*rtp = pvt->rtp;
+#if 0
+	if (pvt->options.bridge) {
 		res = AST_RTP_TRY_NATIVE;
 	}
+#endif
 	ast_mutex_unlock(&pvt->lock);
 
 	return res;

Modified: team/group/NoLossCDR-Redux2/channels/chan_usbradio.c
URL: http://svn.digium.com/view/asterisk/team/group/NoLossCDR-Redux2/channels/chan_usbradio.c?view=diff&rev=113925&r1=113924&r2=113925
==============================================================================
--- team/group/NoLossCDR-Redux2/channels/chan_usbradio.c (original)
+++ team/group/NoLossCDR-Redux2/channels/chan_usbradio.c Wed Apr  9 15:23:38 2008
@@ -2282,7 +2282,7 @@
 	o->lastopen = ast_tvnow();	/* don't leave it 0 or tvdiff may wrap */
 	o->dsp = ast_dsp_new();
 	if (o->dsp) {
-		ast_dsp_set_features(o->dsp, DSP_FEATURE_DTMF_DETECT);
+		ast_dsp_set_features(o->dsp, DSP_FEATURE_DIGIT_DETECT);
 		ast_dsp_set_digitmode(o->dsp, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_RELAXDTMF);
 	}
 

Modified: team/group/NoLossCDR-Redux2/channels/misdn_config.c
URL: http://svn.digium.com/view/asterisk/team/group/NoLossCDR-Redux2/channels/misdn_config.c?view=diff&rev=113925&r1=113924&r2=113925
==============================================================================
--- team/group/NoLossCDR-Redux2/channels/misdn_config.c (original)
+++ team/group/NoLossCDR-Redux2/channels/misdn_config.c Wed Apr  9 15:23:38 2008
@@ -38,9 +38,6 @@
 #include "asterisk/pbx.h"
 #include "asterisk/strings.h"
 #include "asterisk/utils.h"
-
-#define AST_LOAD_CFG ast_config_load
-#define AST_DESTROY_CFG ast_config_destroy
 
 #define NO_DEFAULT "<>"
 #define NONE 0
@@ -1095,7 +1092,7 @@
 	struct ast_variable *v;
 	struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
 
-	if (!(cfg = AST_LOAD_CFG(config, config_flags))) {
+	if (!(cfg = ast_config_load2(config, "chan_misdn", config_flags))) {
 		ast_log(LOG_WARNING, "missing file: misdn.conf\n");
 		return -1;
 	} else if (cfg == CONFIG_STATUS_FILEUNCHANGED)
@@ -1149,7 +1146,7 @@
 	_fill_defaults();
 
 	misdn_cfg_unlock();
-	AST_DESTROY_CFG(cfg);
+	ast_config_destroy(cfg);
 
 	return 0;
 }

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