Commit Graph

1915 Commits

Author SHA1 Message Date
Steve Murphy 6fad66dfb3 Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines

The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the 
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.

If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.

If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.

Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden 
(in trunk).

All the places that previously tested for 
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.

I tested this against the 4 common parking
scenarios:


1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.

2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.

3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.

4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.


No crash.

I also ran the scenarios above against valgrind, and accesses looked good.



........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
Russell Bryant 316f3897a8 Merged revisions 151905 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) | 8 lines

Move AMI initialization to occur after loading modules.  This prevents a
deadlock when someone tries to initiate a module reload from the AMI just
as Asterisk is starting.

(closes issue #13778)
Reported by: hotsblanc
Fix suggested by hotsblanc

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-25 11:02:11 +00:00
BJ Weschke 9aefadd7c1 Do NOT attempt to do anything with the ast_config struct when it's been returned as INVALID by the config file interpreter.
(closes issue #13741)
 


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-20 05:07:25 +00:00
Kevin P. Fleming 1ddc834b39 cleaup of the TCP/TLS socket API:
1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines

2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines)

3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines)

4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied

5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-19 19:11:28 +00:00
BJ Weschke 09e9b5d208 Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766.
We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing.

(closes issue #13715)
reported by: makoto
patch by: bweschke



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-18 02:18:33 +00:00
Michiel van Baak 805556773f Fix CLI command 'channel request hangup'
Prodded on IRC by Russell and fixed by eliel

(closes issue #13730)
Reported by: eliel
Patches:
      main_cli.patch uploaded by eliel (license 64)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-17 17:31:07 +00:00
Mark Michelson 4ad187cba4 Merged revisions 150304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct 2008) | 6 lines

Reverting changes from commits 150298 and 150301 since
I was mistakenly under the assumption that dialplan functions
*always* required that a channel be present. I need to go
home earlier, I think :)


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-16 23:41:16 +00:00
Mark Michelson 8a1d9d1678 Merged revisions 150298,150301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct 2008) | 10 lines

Don't try to call a dialplan function's read callback from
the manager's GetVar handler if an invalid channel has
been specified. Several dialplan functions, including
CHANNEL and SIP_HEADER, do not check for NULL-ness of
the channel being passed in.

(closes issue #13715)
Reported by: makoto


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r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct 2008) | 3 lines

And don't forget to return on the error condition


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-16 23:36:49 +00:00
Mark Michelson 29a8fe20c8 Merged revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines

Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.

Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 23:04:44 +00:00
Tilghman Lesher d5837ba8c2 Add additional memory debugging to several core APIs, and fix several memory
leaks found with these changes.
(Closes issue #13505, closes issue #13543)
Reported by: mav3rick, triccyx
 Patches: 
       20081001__bug13505.diff.txt uploaded by Corydon76 (license 14)
 Tested by: mav3rick, triccyx


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 22:38:06 +00:00
Kevin P. Fleming b17413c992 Merged revisions 148611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct 2008) | 3 lines
  
  it would be nice if this message printing code had actually been tested before it was committed...
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 08:06:45 +00:00
Tilghman Lesher c1351ad237 Merge realtime_update2 branch, which adds a new realtime API call named
'update2', which permits updates which match across multiple columns, instead
of requiring all tables to have a single unique identifier.  All of the other
API calls with the exception of 'update' already had the ability to match on
multiple fields, so it was a missing and very desireable feature that an API
call implementing an update should have this, too.

This does not change any outward performance of Asterisk, but it should make
life easier for application developers who use the RealTime framework.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 00:08:52 +00:00
Steve Murphy db7299f4bc Hmmm. Nobody (but me) is interested in seeing
the trie info when they do 'dialplan show ...'
(even with debug set to non-zero); so I set up a 
   'dialplan debug [context]' cli command instead, 
to explicitly show just the trie info.  I even
added an extension_exists() call to make sure the
trie info is built. I moved the explanatory header
to above the extension loop to ensure it only prints
once. And it will do this now, whether debug is set
or not.

I removed the trie printing from the 'dialplan show' 
command entirely. 



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-13 17:14:38 +00:00
Olle Johansson 32d93bbc0e Highlightning even more bugs in the current tcp/tls implementation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-13 15:49:01 +00:00
Sean Bright 1dedb785ab Don't include logger.h in asterisk.h by default as it is causing problems building
app_voicemail.  Instead, include it where it is needed.  This turned out to be a
relatively minor issue because other headers include logger.h as well.

Need to test -addons before merging this back to 1.6.0.

(closes issue #13605)
Reported by: tomo1657
Patches: 
      13605_seanbright.diff uploaded by seanbright (license 71)
Tested by: mmichelson


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-10 00:42:13 +00:00
Mark Michelson 9851feb8fb The priority was unnecessary for the manager atxfer, so it has
been removed. Furthermore, now we actually use the Context argument
passed to set the transfer context and don't error out if no
context is specified.

This addresses the actual problems outlined in issue 12158. Regarding
the other points brought up, regarding the inability to not transfer
to extensions which cannot be represented by DTMF, it is not enough of
a constraint that it is worth attempting to rework the feature.

(closes issue #12158)
Reported by: davidw



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 23:54:59 +00:00
Mark Michelson 6d70f45506 Merged revisions 146026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines

(closes issue #13579)
Reported by: dwagner

(closes issue #13584)
Reported by: dwagner
Tested by: murf, putnopvut

The thought occurred to me that the res= from the extension spawn
was ending up being returned from the bridge.

"Thou shalt not poison the return value". Made the change
and it appears to allow blind xfers to work as normal.

If I'm wrong, reopen the bugs. But it looks good to me!

Many thanks to putnopvut for helping me reproduce this!


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 23:15:33 +00:00
Tilghman Lesher 8b14e5f493 Reverting format addition for now
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 21:47:02 +00:00
Tilghman Lesher f5d5eb5e19 Fudges for wav16, just like wav49
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 21:37:23 +00:00
Jeff Peeler c897b4e630 (closes issue #13139)
Reported by: krisk84
Tested by: krisk84

This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 19:27:32 +00:00
Steve Murphy e235a07376 (closes issue #13557)
Reported by: nickpeirson
Patches:
      pbx.c.patch uploaded by nickpeirson (license 579)
      replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf

1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
   back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 14:17:33 +00:00
Tilghman Lesher 9335b3ad34 Allow people to select the old console behavior of white text on a black
background, by using the startup flag '-B'.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-07 17:44:32 +00:00
Jeff Peeler 0dc7ac9a6c Explicitly setting these fields to NULL was done because I wasn't sure if they would be NULL otherwise. Since they will be set automatically, removing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-07 16:04:45 +00:00
Jeff Peeler 2ec290b09d Similar to r143204, masquerade the channel in the case of Park being called from AGI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06 23:08:21 +00:00
Jeff Peeler 50923eab5d This commit squashes together three commits because the wrong approach was originally used. (One of the commits was only one line.)
1) r143204:
The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel.

2) r143270:
Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced.

3) r143475: (the one liner)
compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06 22:26:25 +00:00
Jeff Peeler abc88c1d61 fix some comment placement
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06 22:08:40 +00:00
Jeff Peeler 7d8d1f50bb Explicitly set args in park_call_exec NULL so in the case of no options being passed in, there
is no garbage attempted to be used. Also, do not set args to unknown value again if there are
 no options passed in.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06 22:03:01 +00:00
Sean Bright 6855b63d44 Fix a bug with the last item in CLI history getting duplicated when
read from the .asterisk_history file (and subsequently being duplicated
when written).  We weren't checking the result of fgets() which meant
that we read the same line twice before feof() actually returned non-
zero.

Also, stop writing out an extra blank line between each item in the
history file, fix a minor off-by-one error, and use symbolic constants
rather than a hardcoded integer.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-04 16:20:31 +00:00
Jeff Peeler 4b22fb221c remove superfluous reference counting operations in manage_parkinglot since ao2_interator_next increments the ref count automatically
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-03 22:40:59 +00:00
Sean Bright b29fb615ae Resolve a subtle bug where we would never successfully be able to get
the first item in the CLI entry list.  This was preventing '!' from
showing up in either 'help' or in tab completion.

(closes issue #13578)
Reported by: mvanbaak


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-03 22:10:18 +00:00
Terry Wilson 84b0093bef The dialing API should inherit datastores as well as variables
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-03 17:35:37 +00:00
Tilghman Lesher 529874de7b Add schedule extensions to app_meetme. In addition, the reporter found a
problem within strptime(3), which we are correcting here with ast_strptime().
(closes issue #11040)
 Reported by: DEA
 Patches: 
       20080910__bug11040.diff.txt uploaded by Corydon76 (license 14)
 Tested by: DEA


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-01 23:02:25 +00:00
Mark Michelson 7eae109418 Okay, this should really do it now. While I did manage
to fix blind transfers with my last commit here, I also
caused an unwanted side-effect. That is, only the first
priority of the 'h' extension would be executed when
a blind transfer occurred instead of all priorities.

Essentially, my last commit corrected the return value
of ast_bridge_call. However, the implementation still
was not 100% correct. Now it is.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-01 22:23:50 +00:00
Mark Michelson e6799e1c99 if (!(x) == 0) is the same as
if (x).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-01 21:33:11 +00:00
Mark Michelson a23b8c9158 The logic surrounding the return value of ast_spawn_extension
within ast_bridge_call was reversed.

This problem was observed when a blind transfer placed from
the callee channel of a test call failed.

While the problem I am solving here is exactly the same
as what was reported in issue #13584, the difference is
that this fix I am applying is trunk-only. Issue #13584
was reported against the 1.4 branch, and my tests
of 1.4's blind transfers appear to work fine.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-01 21:06:26 +00:00
Michiel van Baak 90751b16ca Merge the cli_cleanup branch.
This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.

This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-28 23:32:14 +00:00
Kevin P. Fleming 629861a705 Merged revisions 144924-144925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) | 6 lines
  
  improve header inclusion process in a few small ways:
  
    - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose
    - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled
    - simplify the usage of some of these headers in the AEL-related stuff in the utils directory
........
  r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) | 2 lines
  
  fix some minor issues with rev 144924
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-27 15:52:56 +00:00
Steve Murphy 579177ae80 Merged revisions 144677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) | 12 lines

(closes issue #13563)
Reported by: mnicholson
Patches:
      found1.diff uploaded by mnicholson (license 96)

This patch was mainly meant to apply to trunk and 1.6.x,
but I'm applying it to 1.4 also, which should be a perfectly
harmless fix to the vast majority of users who are not using
external switches, but the few who might be affected 
will not have to go to the pain of filing a bug report.


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-26 17:50:35 +00:00
Steve Murphy e74584ca3c (closes issue #13557)
Reported by: nickpeirson

The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;

chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-25 22:21:28 +00:00
Steve Murphy 8343faed0e Merged revisions 144066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) | 29 lines

(closes issue #13489)
Reported by: DougUDI
Tested by: murf

(closes issue #13490)
Reported by: seanbright
Tested by: murf

(closes issue #13467)
Reported by: edantie
Tested by: murf, edantie, DougUDI


This crash happens because we are unsafely handling old pointers.
The channel whose cdr is being handled, has been hung up and 
destroyed already. I reorganized the code a bit, and tried not
to lose the fork-cdr-chain concepts of the previous code.
I now verify that the 'previous' channel (the channel we
had when the bridge was started), still exists, by looking it up
by name in the channel list. I also do not try to reset the
CDR's of channels involved in bridges. 

Testing shows it solves the crash problem, and should not
negatively impact previous fixes involving CDR's generated
during/after blind transfers. (The reason we need to reset
the CDR's on the "beginning" channels in the first place).



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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-23 16:52:32 +00:00
Mark Michelson 48ffe6e772 If attempting to free a NULL pointer when MALLOC_DEBUG
is set, don't bother searching for a region to free, just
immediately exit.

This has the dual benefit of suppressing a warning message
about freeing memory at (nil) and of optimizing the free()
replacement by not having to do any futile searching for
the proper region to free.

(closes issue #13498)
Reported by: pj
Patches:
      13498.patch uploaded by putnopvut (license 60)
Tested by: pj



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-17 20:25:40 +00:00
Mark Michelson 224980b07b Merged revisions 143337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) | 6 lines

Allow for "G.729" if offered in an SDP even though
it is not RFC 3551 compliant. Some Cisco switches
will send this in an SDP, and it doesn't hurt to
be able to accept this.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-17 18:26:35 +00:00
Tilghman Lesher 08af5bb312 Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating
when a file is invalid from when a file is missing.  This is most important when
we have two configuration files.  Consider the following example:

Old system:
sip.conf     users.conf     Old result               New result
========     ==========     ==========               ==========
Missing      Missing        SIP doesn't load         SIP doesn't load
Missing      OK             SIP doesn't load         SIP doesn't load
Missing      Invalid        SIP doesn't load         SIP doesn't load
OK           Missing        SIP loads                SIP loads
OK           OK             SIP loads                SIP loads
OK           Invalid        SIP loads incompletely   SIP doesn't load
Invalid      Missing        SIP doesn't load         SIP doesn't load
Invalid      OK             SIP doesn't load         SIP doesn't load
Invalid      Invalid        SIP doesn't load         SIP doesn't load

So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed.  Worse yet, the old
system would do this with no indication that anything was even wrong.

(closes issue #10690)
 Reported by: dtyoo
 Patches: 
       20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 23:30:03 +00:00
Tilghman Lesher bbf8c3d7cc When checking for an encoded character, make sure the string isn't blank, first.
(Closes issue #13470)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 16:54:44 +00:00
Tilghman Lesher 8c53dd7f5e Merged revisions 142740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008) | 4 lines

Don't return a free'd pointer, when a file cannot be opened.
(closes issue #13462)
 Reported by: wackysalut

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 16:29:01 +00:00
Steve Murphy 67f7ac0499 Merged revisions 142675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines

Tested by: sergee, murf, chris-mac, andrew, KNK

This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from 
the ground up!

This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.

Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.

While I dearly hope that this patch overcomes all problems, and 
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.

** trunk note: some code to suppress the h exten being run 
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 04:50:48 +00:00
Steve Murphy c4fc8fe4be Merged revisions 142575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | 20 lines

(closes issue #13364)
Reported by: mdu113

Well, fundamentally, the problems revealed in 13364 are
because of the ForkCDR call that is done before the dial. 
When the bridge is in place, it's dealing with the first
(and wrong) cdr in the list.

So, I wrote a little func to zip down to the first non-locked
cdr in the chain, and thru-out the ast_bridge_call, these
results are used instead of raw chan->cdr and peer->cdr pointers.
This shouldn't affect anyone who isn't forking cdrs before a
dial, and should correct the cdr's of those that do.

So, this change ends up correcting the dstchannel
and userfield; the disposition was fixed by a previous
patch, it was OK coming into this problem.



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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-11 23:12:53 +00:00
Steve Murphy 4fc65a69a2 Merged revisions 142474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | 30 lines

(closes issue #12318)
Reported by: krtorio

I made a small change to the code that handles local channel situations.
In that code, I copy the answer time from the peer cdr, to the bridge_cdr,
but I wasn't also copying the disposition from the peer cdr.

So, Now I copy the disposition, and I've tested against 
these cases:

1. phone 1 never answers the phone; no cdr is generated at all.
   this should show up as a manager command failure or something.

2. phone 2 never answers. CDR is generated, says NO ANSWER

3. phone 2 is busy. CDR is generated, says BUSY

4. phone 2 answers: CDR is generated, times are correct; disposition
   is ANSWERED, which is correct. The start time is the time that
   the manager dialed the first phone. The answer time is the time
   the second phone picks up.

I purposely left the cid and src fields blank; since this call really
originates from the manager, there is no 'easy' data to put in these
fields. If you feel strongly that these fields should be filled in,
re-open this bug and I'll dig further.




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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-10 22:11:27 +00:00
Russell Bryant bdfd731b8b Merged revisions 142354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) | 7 lines

It is a normal situation that a task gets put in the scheduler that should run
as soon as possible.  Accept "0" as an acceptable time to run, and also treat
negative as "run now", and don't print a debug message about it.

(inspired by a message asking about the "request to schedule in the past"
 debug message on the -dev list)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-10 16:41:55 +00:00
Richard Mudgett 75e95cb2b3 Cleaned up comment
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-09 17:30:52 +00:00