Commit Graph

1575 Commits

Author SHA1 Message Date
Joshua Colp c7d51a7fc1 Put my slinfactory changes back in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-07 14:54:42 +00:00
Tilghman Lesher 0e6140c564 Use a 32k file buffer on recordings, which increases the efficiency of file recording.
(closes issue #11962)
 Reported by: garlew
 Patches: 
       recording.patch uploaded by garlew (license 376)
       bug-11962.diff uploaded by snuffy (license 35)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-03 07:49:05 +00:00
Terry Wilson 1eb31edde2 Re-add HTTP post support by moving to res_http_post.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-02 15:25:48 +00:00
Jeff Peeler a5cdd849e5 This adds DNS SRV record support to DNS manager. If there is a SRV record for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 17:53:08 +00:00
Joshua Colp 7dab892401 Merged revisions 112125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines

Ensure that we do not exceed the hold's maximum size with a single frame.
(closes issue #12047)
Reported by: fabianoheringer
Tested by: fabianoheringer

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@112126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-01 16:50:37 +00:00
Russell Bryant afd8783577 Make some notes about common usage of pbx_builtin_getvar_helper() that is not
thread-safe.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28 22:50:46 +00:00
Mark Michelson 3a0f4cc933 Temporary revert of 111662. It's causing lots of trouble and appears to not be
the proper solution to the problem reported anyway.

(related to issue #12884)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28 19:14:51 +00:00
Mark Michelson ca8e44c051 The copy_request function did not take into account the necessary null terminator
for the string to be copied into. This resulted in parse_request reading invalid
memory beyond the end of the string, and in some cases led to crashes. Thanks
to falves11 for providing the valgrind output which led to the closure of this issue.

(closes issue #12284)
Reported by: falves11



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-28 16:36:59 +00:00
Steve Murphy 2427603eaf Merged revisions 111341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines


(closes issue #12302)
Reported by: pj
Tested by: murf

These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch, 
and ever after that, till the end of the exten, we substitute any ${EXTEN} 
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also). 
The reason for this, is that because switches are coded using 
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value, 
which blows the minds of users.


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-27 04:47:12 +00:00
Tilghman Lesher e04025ead9 Simplify new macro, simplify configfile logic, now that list is sorted
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 19:58:09 +00:00
Tilghman Lesher e6fc9ae52c Add a linkedlist macro that maintains a sorted list
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 19:19:31 +00:00
Jason Parker 6412a96e43 Large cleanup of DSP code
Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.

2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.

3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.

4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.

5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.


(closes issue #11968)
Reported by: dimas
Patches:
      v2-11968-dsp.patch uploaded by dimas (license 88)
      v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 19:05:51 +00:00
Tilghman Lesher ef4eff9a9b Add the "config reload <conffile>" command, which allows you to tell Asterisk
to reload any file that references a given configuration file.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26 18:39:06 +00:00
Joshua Colp 358ac2f76a Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-25 14:39:45 +00:00
Joshua Colp 30d85b3144 Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21 17:58:59 +00:00
Russell Bryant bccebdd21f Remove astobj.h from some places where it wasn't needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-20 17:45:29 +00:00
Jason Parker 9e3603dac9 Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading.

Early cleanup for issue #11968


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 22:25:34 +00:00
Jason Parker 8d4276578a Rename very poorly named function to reflect what it actually does. This was causing quite a bit of confusion for me...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19 21:56:15 +00:00
Kevin P. Fleming e191b51a08 start the process of changing HTTP request dispatching to do it based on *both* URI and method, so that POST support can move into a module; move http.c's private function prototypes into _private.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 22:32:26 +00:00
Mark Michelson 4232cef812 Add format attribute to printf-style functions in astmm.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 17:00:53 +00:00
Kevin P. Fleming 00222ab483 ensure that dependencies on AST_C_DEFINE_CHECK symbols work properly
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 15:50:29 +00:00
Terry Wilson b02bc230af Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se.  I also added format attributes to any printf wrapper functions I found that didn't have them.  -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 15:43:34 +00:00
Steve Murphy 4e29ae8806 Merged revisions 109309 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines

(closes issue #11903)
Reported by: atis

Many thanks to atis for spotting this problem and reporting it.
The fix was to straighten out how items are placed on and removed
from the file stack. Regressions as well as the provided test case
helped to straighten out all code paths. valgrind was used to make
sure all memory allocated was freed.

Sorry for not solving this earlier. I got distracted.

Added the ntest23 regression test, which is mainly a copy of ntest22, 
but with a few juicy errors thrown in, to replicate the kind of 
error that atis spotted.



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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 14:09:50 +00:00
Terry Wilson e727d15d34 Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory.
This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros.

If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 22:10:06 +00:00
Steve Murphy 0af58d3f5c (closes issue #12238)
Reported by: mvanbaak
Tested by: murf, mvanbaak

Due to a bug that occurred when merge_contexts_and_delete scanned the "old" or existing contexts, and found a context
that doesn't exist in the new set, yet owned by a different registrar. The context is created in the new set, with the
old registrar, and and all the priorities and extens that have a different registrar are copied into it. But, not the
includes, ignorepats, and switches. I added code to do this immediately after the context is created.

This still leaves a logical hole in the code. If you define a context in two places, (eg. in extensions.conf and also 
in extensions.ael), and they both have includes, but different in composition, no new context will be generated, and
therefore the 'old' includes, switches, and ignorepats will not be copied. I'd have added code to simply add any non-duplicates
into the 'new' context that had a different registrar, but there is one big complication: includes, and switches are definitely
order dependent. (ignorepats I'm not sure about). And we'll have to develop some sort of policy about how we 
merge order dependent lists, especially if the intersection of the two sets is empty. (in other words, they do not have any
elements in common). Do the new go first, or the old? I've elected to punt this issue until a user complains. Hopefully,
this is pretty rare thing.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 17:47:36 +00:00
Russell Bryant 928475694a Doxygenify dnsmgr.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-14 21:20:36 +00:00
Russell Bryant 835df7d30f Merged revisions 108583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines

Fix another issue that was causing crashes in chanspy.  This introduces a new
datastore callback, called chan_fixup().  The concept is exactly like the
fixup callback that is used in the channel technology interface.  This callback
gets called when the owning channel changes due to a masquerade.  Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.

(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 21:40:43 +00:00
Russell Bryant 8bbef5f996 Rename ast_tcptls_server_instance to session_instance, since this pertains to
server and client usage.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 22:13:18 +00:00
Mark Michelson a96b50580c Merged revisions 108227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines

Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when
it is appropriate and when it is not appropriate to use it.

I also removed the part of the debug message that mentions that this is probably a bug because
there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g.
when the scheduler callback manually reschedules with a new id instead of returning non-zero to
tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug
message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro
is probably being used in several places where it shouldn't be. Also removed the redundant line,
file, and function information since that is provided by ast_log.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 21:19:30 +00:00
Joshua Colp a3c7b08d19 Doxygenify slinfactory a bit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 21:06:50 +00:00
Joshua Colp 5fc569f5f5 Merged revisions 108083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines

Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 18:29:33 +00:00
Tilghman Lesher d02f74ebfe An offhand comment from Russell made me realize that the configuration file
caching would not work properly for users.conf and any other file read from
more than one place.  I needed to add the filename which requested the config
file to get it to work properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 22:55:16 +00:00
Kevin P. Fleming 5875146526 Merged revisions 107461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar 2008) | 2 lines

stop checking for mktime() in the configure script... we don't use it, and the test is buggy under gcc 4.3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 14:37:03 +00:00
Kevin P. Fleming 90786d21c9 Merged revisions 107405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar 2008) | 2 lines

fix small bug in IMAP toolkit testing

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 13:58:37 +00:00
Tilghman Lesher bdad3c9889 (closes issue #6019)
Reported by: ssokol
 Patches: 
       20080304__bug6019.diff.txt uploaded by Corydon76 (license 14)
 Tested by: putnopvut


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 21:48:20 +00:00
Joshua Colp af7e1964f2 Merged revisions 107016 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines

Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial.
(closes issue #11516)
Reported by: ys
Patches:
      branch_1.4_cdr.diff uploaded by ys (license 281)
Tested by: anest, jcapp, dartvader

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 14:36:16 +00:00
Steve Murphy 377e51c4d4 (closes issue #6002)
Reported by: rizzo
Tested by: murf

Proposal of the changes to be made, and then an announcement of how they were accomplished:

http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html

and:

http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html

Here is a recap, file by file, of what I have done:

pbx/pbx_config.c
pbx/pbx_ael.c

All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.

We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.

pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and 
then call merge_contexts_and_delete, which will merge (now) existing contexts and 
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then 
destroy the old dialplan.



chan_sip.c
chan_iax.c
chan_skinny.c

All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.

chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.


apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c

All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.


include/asterisk/pbx.h

ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create()  interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.


include/asterisk/pval.h

ast_compile_ael2() interface changed to include the local hashtab table ptr.


main/features.c

For the sake of the parking context, we use ast_context_find_or_create().



main/pbx.c

I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.

refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.

Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.

Added some calls to ast_verb(3,...) for debug messages

Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.

find_or_create was upgraded to handle both local lists/tables as well as the globals.

context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables

ast_merge_contexts_and_delete() was heavily modified.

ast_add_extension2() was also upgraded to handle changes. 

the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.



res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile

Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps.  The main gotcha was I had to 
include lock.h and hashtab.h in several places.


As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.

How's this for verbose commit messages?




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 18:57:57 +00:00
Russell Bryant 121bc8141f Merged revisions 106704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) | 8 lines

Change a warning message to a debug message.  This is happening quite frequently,
and it is not worth spamming users with these messages unless we are pretty confident
that it should never happen.  As it stands today, it _will_ and _does_ happen and
until that gets cleaned up a reasonable amount on the development side, let's not
spam the logs of everyone else.

(closes issue #12154)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 17:17:30 +00:00
Joshua Colp 496adc6fc0 Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines

Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:43:22 +00:00
Tilghman Lesher 8a411ccf83 Create a centralized configuration option for silencethreshold
(closes issue #11236)
 Reported by: philipps
 Patches: 
       20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
 Tested by: philipps


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 16:23:44 +00:00
Tilghman Lesher 7a3f642207 Merged revisions 106015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines

Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
(closes issue #12140)
 Reported by: slavon
 Patches: 
       sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 15:23:32 +00:00
Russell Bryant a760a033e9 Merged revisions 105932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines

Fix a bug that I just noticed in the RTP code.  The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use.  The len field
represents the number of ms of audio that the frame contains.  It would have
set the value to be twice what it should be.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 01:54:16 +00:00
Russell Bryant 96e04792bd add a destroy API call for a server instance
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 22:28:03 +00:00
Russell Bryant cc55483858 More public API name changes to use an appropriate ast_ prefix
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 22:23:21 +00:00
Russell Bryant efb1e30a38 Rename public object server_instance to ast_tcptls_server_instance
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 22:15:18 +00:00
Joshua Colp 4de0d8368f Merged revisions 105674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines

When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 18:08:42 +00:00
Russell Bryant 8372a9bf08 3) In addition to merging the changes below, change trunk back to a regular
LIST instead of an RWLIST.  The way this list works makes it such that
   a RWLIST provides no additional benefit.  Also, a mutex is needed for
   use with the thread condition.


Merged revisions 105563 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines

Merge in some changes from team/russell/autoservice-nochans-1.4

These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.

1) Change it so that autoservice thread doesn't keep looping around calling
   ast_waitfor_n() on 0 channels twice a second.  Instead, use a thread condition
   so that the thread properly goes to sleep and does not wake up until a
   channel is put into autoservice.

   This actually fixes an interesting bug, as well.  If the autoservice thread
   is already running (almost always is the case), then when the thread goes
   from having 0 channels to have 1 channel to autoservice, that channel would
   have to wait for up to 1/2 of a second to have the first frame read from it.

2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
   channels and no fds to poll() on, such as was the case with the previous code
   for the autoservice thread.  In this case, the code would call alloca(0), and
   pass the result as the first argument to poll().  In this case, the 2nd
   argument to poll() specified that there were no fds, so this invalid pointer
   shouldn't actually get dereferenced, but, this code makes it explicit and
   ensures the pointers are NULL unless we have valid data to put there.

(related to issue #12116)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:59:50 +00:00
Terry Wilson 7d1891d5c3 Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue.
(closes issue #11520)
Reported by: pliew
Tested by: otherwiseguy


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-01 01:30:37 +00:00
Russell Bryant 7da06e6cb8 Merged revisions 105116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) | 8 lines

Fix a bug in the lock tracking code that was discovered by mmichelson.  The issue
is that if the lock history array was full, then the functions to mark a lock as
acquired or not would adjust the stats for whatever lock is at the end of the array,
which may not be itself.  So, do a sanity check to make sure that we're updating
lock info for the proper lock.

(This explains the bizarre stats on lock #63 in BE-396, thanks Mark!)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-28 22:39:26 +00:00
Tilghman Lesher 4aff24881b Bring Voicetronix driver up to date with current drivers
(closes issue #12084)
 Reported by: mmickan
 Patches: 
       chan_vpb.cc.diff uploaded by mmickan (license 400)
       module.h.diff uploaded by mmickan (license 400)
       vpb.conf.sample uploaded by mmickan (license 400)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27 08:20:15 +00:00