Commit Graph

10874 Commits

Author SHA1 Message Date
Jason Parker
9ad1e5d09c Fix a crash when checking version attribute in an incoming XML caps element.
Issue 9667, patch by phsultan.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@63099 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 20:03:49 +00:00
Pari Nannapaneni
0b01c54b90 explanation for httptimeout in manager.conf
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2007-05-04 16:45:29 +00:00
Joshua Colp
95de3fbf0c Merged revisions 62987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines

When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)

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2007-05-03 16:44:00 +00:00
Kevin P. Fleming
9c638b6507 improve loader a bit, by avoiding trying to initialize embedded modules twice and avoiding trying to load modules from disk when they have been loaded already during the 'preload' pass (reported by blitzrage on IRC, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 16:38:56 +00:00
Russell Bryant
796acdbbdb Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).
This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end.  This is fixed,
along with a couple other little improvements.

* When chan_zap is in the middle of playing a digit to a channel, it feeds
  back null frames, not voice frames.  So, I have modified ast_read to check
  the timing on emulated DTMF when it receives null frames, in addition to
  where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits.  If there was
  no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
  frames that pass through, just use time values.  Now there is no code in this
  section that assumes 8kHz audio.


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2007-05-03 15:23:13 +00:00
Steve Murphy
9056b0fa09 updated the ael regressions to match what's in trunk
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2007-05-03 14:41:53 +00:00
Christian Richter
460e677ea6 Merged revisions 61357,61770,62885 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line

some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line

added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line

fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension.  added encoding of keypad.
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2007-05-03 14:36:32 +00:00
Steve Murphy
fd54b47404 These mods fix bug 9623, where an '@' in the eswitch contents causes a syntax error. I also updated the regressions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 13:54:56 +00:00
Kevin P. Fleming
b488e05054 Merged revisions 62841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 May 2007) | 2 lines

doh... initializing the pointer variable will work just a bit better

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2007-05-03 00:23:37 +00:00
Kevin P. Fleming
ae0f7a3247 Merged revisions 62796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 May 2007) | 7 lines

increase reliability and efficiency of static Realtime config loading via ODBC:
	don't request fields we aren't going to use
	don't request sorting on fields that are pointless to sort on
	explicitly request the fields we want, because we can't expect the database to always return them in the order they were created
	
(reported by blitzrage in person (!), patch by me)

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2007-05-03 00:02:57 +00:00
Kevin P. Fleming
51d6dd84e6 improve static Realtime config loading from PostgreSQL:
don't request sorting on fields that are pointless to sort on
	use ast_build_string() instead of snprintf()
	don't request the list of fieldnames that resulted from the query when we both knew what they were before we ran the query _AND_ we aren't going to do anything with them anyway

(patch by me, inspired by blitzrage's bug report about res_config_odbc)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 23:57:23 +00:00
Russell Bryant
ff3e3ba129 Merge changes from team/russell/inband_dtmf ...
Fix some issues related to generating inband DTMF.  There are two changes here:

1)   The list of DTMF tones in the senddigit_begin() function explicitly
   specified 100ms of the tone followed by 100ms of silence.  This really
   broke things with the way that Asterisk now wants complete control
   over when the digit begins and ends.  So, regardless of what Asterisk
   really wanted to do, this was going to play out the tone at the length it
   wanted to.  This caused various problems like DTMF translation to inband to
   be extremely unreliable.
     The list of tones has been changed so that the correct DTMF tone is played
   indefinitely until Asterisk tells it to stop.

2) ast_write() had to be modified to let a DTMF_END frame get processed even
   when a generator is present.  This is how the tone will finally get stopped.

(issues #8944, #9250, #9348, maybe others.  Thanks to mdu113 from #8944 for
 the testing and feedback!)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 22:59:09 +00:00
Russell Bryant
ae7a38d87b Backport the change that only went in to trunk that fixes the command manager
action over http.  (reported internally by pari and bkruse)


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2007-05-02 20:55:00 +00:00
Steve Murphy
d2c7601c21 Merged revisions 62737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May 2007) | 1 line

Some tweaks to satisfy CDR bug 8796, where being in 'h' extension louses up the dst field
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2007-05-02 20:46:07 +00:00
Tilghman Lesher
8bcfcfca88 Merged revisions 62691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines

Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.

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2007-05-02 17:43:48 +00:00
Steve Murphy
55f4eb3e3d a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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2007-05-02 17:10:50 +00:00
Olle Johansson
d5fda03428 Don't unlock a channel that we already know does not exist (propably isue 8228)
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2007-05-02 06:15:43 +00:00
Russell Bryant
6e13855338 Merged revisions 62547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines

Remove an unnecessary check that makes it so if you hang up after doing an
attended transfer before the target extension answers the channel, the transfer
is not successful.  (issue #9338, patch by svanlund)

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2007-05-01 21:57:10 +00:00
Tilghman Lesher
e5f56f776d Bug 9590 - Memory leaks around find_user() (found by rayjay, different fixes by me)
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2007-05-01 21:34:43 +00:00
Russell Bryant
58352f5d46 Merged revisions 62496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines

Add indications.conf information for the Philippines.
(issue #9525, reported and patched by loloski)

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2007-05-01 16:26:48 +00:00
Russell Bryant
3595c5fed1 Merged revisions 62417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines

This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588, reported and patched by softins)

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2007-04-30 15:58:28 +00:00
Russell Bryant
63a37f4755 When serving dynamic content, include a Cache-Control header to instruct the
browsers to not store the resulting content.  
(issue #9621, reported by Pari, patch by me)


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2007-04-30 15:25:31 +00:00
Jason Parker
16405bbca9 Remove unused (and potentially confusing) jitterbuffer options from sample config.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@62371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30 14:52:31 +00:00
Joshua Colp
5c35c59bb4 Merged revisions 62368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 lines

Update copyright notice. It's now the year 2007!

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2007-04-30 14:36:11 +00:00
Russell Bryant
8936804377 Fix a bug that made the "language" setting in zapata.conf not
functional.  (issue #9626, reported and fixed by sergee)


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2007-04-29 05:50:37 +00:00
Russell Bryant
698a82996e Note that the "talker optimization" option will be enabled by default in 1.6
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2007-04-28 21:56:20 +00:00
Russell Bryant
6b76a95f09 Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks.  This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel.  However, this is the
wrong thing to do.  It should *always* return 0, instead.  When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)


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2007-04-27 21:10:51 +00:00
Jason Parker
45be059315 Merged revisions 62173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 lines

This transcoder message needn't be a NOTICE.
I've seen it cause confusion more than a few times.

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2007-04-27 16:17:46 +00:00
Russell Bryant
a53ebcf546 If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a 
channel.  Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)


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2007-04-27 16:14:11 +00:00
Olle Johansson
34f9e0c4dd Merged revisions 62126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines

Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.


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2007-04-27 14:04:07 +00:00
Olle Johansson
b0121ca8a3 Issue #9608 - fix some annoying DEBUG messages not controlled by option_debug (DEA). Thanks!
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2007-04-27 08:10:43 +00:00
Joshua Colp
0525347958 Merged revisions 62037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines

Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)

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2007-04-26 16:33:52 +00:00
Joshua Colp
230dc4ad88 Missed an ast_app_group_discard during merge. Thanks blitzrage!
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2007-04-26 03:19:51 +00:00
Joshua Colp
1c6f5bd851 Don't always say that the channel is being paused if it is actually being unpaused in the Manager ack message. (reported by jsmith in #asterisk-bugs)
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2007-04-26 01:48:55 +00:00
Joshua Colp
19a9e28b09 Merged revisions 61958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 lines

Don't count failed include attempts against the configuration include level. (issue #9593 reported by mostyn)

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2007-04-26 01:27:18 +00:00
Kevin P. Fleming
0e525813a8 Merged revisions 61913 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines

handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286)

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2007-04-25 22:29:53 +00:00
Russell Bryant
5574da0c12 Merged revisions 61866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines

If the callerid= option is specified, but empty, clear any previous data.

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2007-04-25 21:59:07 +00:00
Russell Bryant
e6aa1145ed Merged revisions 61862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines

Ensure that callerid settings are reset on a reload.

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2007-04-25 21:13:15 +00:00
Joshua Colp
1d4adc0174 Merged revisions 61804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines

Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)

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2007-04-25 19:21:54 +00:00
Russell Bryant
77c52fd8bf Merged revisions 61798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines

Fix a typo where cid_num got copied instead of cid_ani.  
(issue #9587, reported and patched by xrg)

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2007-04-25 16:22:07 +00:00
Russell Bryant
5bee74171d Merged revisions 61786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | 4 lines

Don't crash if a manager connection provides a username that exists in
manager.conf but does not have a password, and also requests MD5 
authentication. (ASA-2007-012)

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2007-04-24 21:34:53 +00:00
Russell Bryant
456cad8a47 Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list.  I changed the enforced minimum length of a
digit from 100ms to 80ms.  Furthermore, I made it now enforce a gap of 45ms in
between digits.  These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.


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2007-04-24 19:00:06 +00:00
Dwayne M. Hubbard
db55615090 Merged revisions 61777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) | 1 line

removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor()
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2007-04-24 18:43:53 +00:00
Russell Bryant
bdd29c22c2 Add a few more state changes in handle_frame_ownerless() so that the SLA code
will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.


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2007-04-24 16:16:41 +00:00
Joshua Colp
b271179161 Merged revisions 61771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines

Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)

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2007-04-24 16:07:02 +00:00
Russell Bryant
215adb35b6 Some dialplan functions, such as CUT(), expect to operate on variables on a
channel.  So, this little hack lets them work in places where a channel doesn't
exist, such as within DUNDi configuration.
(issue #9465, reported and patched by Corydon76, testing by blitzrage)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-23 18:17:00 +00:00
Russell Bryant
baab073219 Ensure that digits passing through Asterisk have a reasonable minimum length.
It is currently 100 ms.  If someone thinks this should be different, feel free
to speak up.  (related to issues #8944, #9250, and #9348)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-23 17:57:32 +00:00
Jason Parker
9378347264 Avoid invalid seqno cycling detection.
Per comment from Dave Troy:
 This adds back in some simple typecasting I had in an earlier version
 which I realize now may be breaking things.

Issue #9554.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-20 21:35:27 +00:00
Jason Parker
bb59c9520b Merged revisions 61704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 lines

Fix an issue that I noticed while looking over issue 9571.

The reload timestamp was getting set after reloading the built-in stuff, and before the modules.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-20 21:15:29 +00:00
Russell Bryant
477ff09bdf Remove a stray debug message introduced by a recent commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-20 20:42:02 +00:00