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Author SHA1 Message Date
lmadsen e73cab2f3f Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

................
  r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
  
  Merged revisions 328209 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
    
    Introduce <support_level> tags in MODULEINFO.
    This change introduces MODULEINFO into many modules in Asterisk in order to show
    the community support level for those modules. This is used by changes committed
    to menuselect by Russell Bryant recently (r917 in menuselect). More information about
    the support level types and what they mean is available on the wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
  ........
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-14 20:28:54 +00:00
dvossel 4534ea67fb Merged revisions 316650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r316650 | dvossel | 2011-05-04 09:25:03 -0500 (Wed, 04 May 2011) | 15 lines
  
  Merged revisions 316644 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011) | 9 lines
    
    Fixes one-way-audio when chanspy activated with the 'o' option
    
    (closes issue #18382)
    Reported by: jkister
    Patches: 
          0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt uploaded by malin (license )
    Tested by: firstsip, Greenlightcrm, malin, wdoekes, boroda, dvossel
  ........
................


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316657 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-04 14:26:33 +00:00
jrose c1a662b055 Merged revisions 311197 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines
  
  This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
  
  In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy.
  
  (closes issue #18742)
  Reported by: jkister
  Tested by: jkister, jcovert, jrose
  
  Review: http://reviewboard.digium.internal/r/106/
........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@311198 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-17 19:05:42 +00:00
dvossel f27e928f05 Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22 23:04:49 +00:00
dvossel 4aca3187a3 Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03 16:22:10 +00:00
pabelanger 86f1888179 Merged revisions 299865 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r299865 | pabelanger | 2010-12-28 13:53:37 -0500 (Tue, 28 Dec 2010) | 9 lines
  
  Merged revisions 299864 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec 2010) | 2 lines
    
    Documentation typo
  ........
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@299866 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-28 19:00:04 +00:00
twilson 92fefe352c Merged revisions 284921 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r284921 | twilson | 2010-09-03 11:28:18 -0500 (Fri, 03 Sep 2010) | 19 lines
  
  Merged revisions 284897 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r284897 | twilson | 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) | 12 lines
    
    Merged revisions 284881 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines
      
      Properly detect when a sound file doesn't exist
      
      ast_fileexists returns -1 for error and 0 for a non-existant file. The existing
      code treated missing files as though they existed.
    ........
  ................
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284922 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-03 16:42:53 +00:00
tilghman 317ea2e45d Display a list of channel variables in each channel-oriented event.
(Closes AST-33)
Reviewboard:	https://reviewboard.asterisk.org/r/368/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-13 20:42:03 +00:00
jpeeler 39a676a838 Fix the fix for chanspy option o
In 224178, I assumed the uploaded patch was correct as it had received positive
feedback. The flags were being checked in the incorrect location. Upon testing
the fix this time it was also found that the flags from the dialplan weren't
being copied to the chanspy_translation_helper.

(closes issue #16167)
Reported by: marhbere



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228189 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05 21:23:06 +00:00
tilghman 3bacd4082e Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04 14:05:12 +00:00
kpfleming 4f428997ca Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.

During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.

Review: https://reviewboard.asterisk.org/r/379/



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225244 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21 21:08:47 +00:00
jpeeler 3531d342bf Readd removed ability to allow listening to one side of the call in app_chanspy
(Option o)

(closes issue #15675)
Reported by: john8675309
Patches:
      issue15675patchtrunk.txt uploaded by dbrooks (license 790)
Tested by: jgutierrez on users list:
 http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224178 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-15 15:57:14 +00:00
seanbright 805af36f64 Get this compiling under dev-mode.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219230 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-17 16:25:38 +00:00
tilghman a679fc0bd5 Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up.
In addition, there's a bit of cleanup to the arguments and documentation, in which
I discovered that the last feature added to this application duplicated an option
(oops!) and changed that option so that it now works.
(closes issue #14909)
 Reported by: junky
 Patches: 
       __20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10)
 Tested by: amilcar, junky, flujan, lmadsen


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219105 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-17 00:58:10 +00:00
tilghman d1ec1aa57d AST-2009-005
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10 19:20:57 +00:00
russell 23cd457af9 Add 's' option to ChanSpy, which makes the app exit when no channels are left to spy on.
(closes issue #14594)
Reported by: JimDickenson
Patches:
      chanspy.diff uploaded by JimDickenson (license 710)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203842 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26 21:48:41 +00:00
kpfleming ea5a74f18b Merged revisions 200991 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
  
  Improve support for media paths that can generate multiple frames at once.
  
  There are various media paths in Asterisk (codec translators and UDPTL, primarily)
  that can generate more than one frame to be generated when the application calling
  them expects only a single frame. This patch addresses a number of those cases,
  at least the primary ones to solve the known problems. In addition it removes the
  broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
  functions, and cleans up various code paths affected by these changes.
  
  https://reviewboard.asterisk.org/r/175/
........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201056 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16 18:54:30 +00:00
russell f63398715e Global var cleanup - constification and removing unused vars.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199479 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-07 14:55:51 +00:00
mmichelson 6553d7e59e Merged revisions 197537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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  r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
  
  Add flags to chanspy audiohook so that audio stays in sync.
  
  There are two flags being added to the chanspy audiohook here. One
  is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
  we ensure that the read and write slinfactories on the audiohook do
  not skew beyond a certain tolerance.
  
  In addition, there is a new audiohook flag added here,
  AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
  a slinfactory to build up a substantial amount of audio before 
  flushing it. For this particular issue, this means that the person 
  spying on the call will hear the conversations in real time with very 
  little delay in the audio.
  
  (closes issue #13745)
  Reported by: geoffs
  Patches:
        13745.patch uploaded by mmichelson (license 60)
  Tested by: snblitz
........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197543 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28 14:58:06 +00:00
kpfleming 230a66da7d Const-ify the world (or at least a good part of it)
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:

- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments

In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.

Review: https://reviewboard.asterisk.org/r/251/



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21 21:13:09 +00:00
kpfleming f58bc31e46 add 'const' qualifiers in various places where they should have been
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@193832 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-12 13:59:35 +00:00
russell 89175b7e04 Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big 
improvement for performance, stability, code maintainability, 
and ease of future code development.

The channel list is no longer an unsorted linked list.  The main container 
for channels is an astobj2 hash table.  All of the code related to searching 
for channels or iterating active channels has been rewritten.  Let n be 
the number of active channels.  Iterating the channel list has gone from 
O(n^2) to O(n).  Searching for a channel by name went from O(n) to O(1).  
Searching for a channel by extension is still O(n), but uses a new method 
for doing so, which is more efficient.

The ast_channel object is now a reference counted object.  The benefits 
here are plentiful.  Some benefits directly related to issues in the 
previous code include:

1) When threads other than the channel thread owning a channel wanted 
   access to a channel, it had to hold the lock on it to ensure that it didn't 
   go away.  This is no longer a requirement.  Holding a reference is 
   sufficient.

2) There are places that now require less dealing with channel locks.

3) There are places where channel locks are held for much shorter periods 
   of time.

4) There are places where dealing with more than one channel at a time becomes 
   _MUCH_ easier.  ChanSpy is a great example of this.  Writing code in the 
   future that deals with multiple channels will be much easier.

Some additional information regarding channel locking and reference count 
handling can be found in channel.h, where a new section has been added that 
discusses some of the rules associated with it.

Mark Michelson also assisted with the development of this patch.  He did the 
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it 
much easier to deal with holding on to a channel pointer for an extended period 
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.

Thanks to David Vossel for his assistance with this branch, as well.  David 
did the conversion of the DAHDIScan application by making it become a wrapper 
for ChanSpy internally.

The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.

Review: http://reviewboard.digium.com/r/203/


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24 14:04:26 +00:00
mmichelson eca3b22df7 Add manager events for chanspy starting or stopping
(closes issue #14469)
Reported by: caio1982
Patches:
      chanspy_events2.diff uploaded by caio1982 (license 22)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175655 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13 20:35:26 +00:00
mmichelson 11cc929d0f Fix potential for stack overflows in app_chanspy.c
When using the 'g' or 'e' options, the stack allocations that
were used could cause a stack overflow if a spyer stayed on the
line long enough without actually successfully spying on anyone.

The problem has been corrected by using static buffers and copying
the contents of the appropriate strings into them instead of using
functions like alloca or ast_strdupa



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@174805 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10 23:17:03 +00:00
file daf0032d06 Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
(closes issue #14414)
Reported by: bluecrow76


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173902 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06 15:59:17 +00:00
mmichelson 62bbaefff9 Merged revisions 173396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines

Revert my previous change because it was stupid


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173397 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04 17:45:14 +00:00
mmichelson 009cddba58 Merged revisions 173392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines

Add a missing unlock. Extremely unlikely to ever matter, but it's needed.


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173393 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-04 17:41:02 +00:00
russell 66e07d339a Merged revisions 165889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines

Ensure that the chanspy datastore is fully initialized.

This patch resolved some random crash issues observed by a user on a BSD system

(closes issue #14111)
Reported by: ys
Patches:
      app_chanspy.c.diff uploaded by ys (license 281)

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165890 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19 15:05:09 +00:00
file e1f767f777 Only detach and destroy the whisper audiohooks if they are actually in use.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163912 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-13 00:59:24 +00:00
eliel ddd3625c89 - Add more <see-also> based on TFOT.
- Add the 'filename' type to the see-also ref. To be able to reference a filename.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154578 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05 13:07:29 +00:00
kpfleming cc1b2c100f bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153616 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02 18:52:13 +00:00
russell 44147470e5 Fix various spelling and grammatical issues in documentation
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153468 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02 02:50:33 +00:00
russell b1f91b97d2 Merge changes from team/group/appdocsxml
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01 21:10:07 +00:00
russell 31d09879fe Merged revisions 139213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines

Fix a crash in the ChanSpy application.  The issue here is that if you call
ChanSpy and specify a spy group, and sit in the application long enough looping
through the channel list, you will eventually run out of stack space and the
application with exit with a seg fault.  The backtrace was always inside of
a harmless snprintf() call, so it was tricky to track down.  However, it turned
out that the call to snprintf() was just the biggest stack consumer in this
code path, so it would always be the first one to hit the boundary.

(closes issue #13338)
Reported by: ruddy

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139215 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-20 22:16:36 +00:00
mmichelson d7e3c35a50 Merged revisions 138886 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug 2008) | 23 lines

Add a lock and unlock prior to the destruction of the chanspy_ds
lock to ensure that no other threads still have it locked. While
this should not happen under normal circumstances, it appears that
if the spyer and spyee hang up at nearly the same time, the following
may occur.

1. ast_channel_free is called on the spyee's channel.
2. The chanspy datastore is removed from the spyee's channel in 
   ast_channel_free.
3. In the spyer's thread, the spyer attempts to remove and destroy the datastore
   from the spyee channel, but the datastore has already been removed in step 2, 
   so the spyer continues in the code.
4. The spyee's thread continues and calls the datastore's destroy callback, 
   chanspy_ds_destroy. This involves locking the chanspy_ds.
5. Now the spyer attempts to destroy the chanspy_ds lock. The problem is that in step 4, 
   the spyee has locked this lock, meaning that the spyer is attempting to destroy a lock 
   which is currently locked by another thread.

The backtrace provided in issue #12969 supports the idea that this is possible
(and has even occurred). This commit does not close the issue, but should help
in preventing one type of crash associated with the use of app_chanspy.


........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@138887 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-19 18:52:04 +00:00
kpfleming 0891b8a53c make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05 16:56:11 +00:00
mmichelson f76a823f67 merging the zap_and_dahdi_trunk branch up to trunk
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134050 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28 16:00:19 +00:00
mmichelson 5e846e20b2 Merged revisions 133169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines

As suggested by seanbright, the PSEUDO_CHAN_LEN in 
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.

Also changed the next_unique_id_to_use to have the 
static qualifier.

Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133171 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23 19:48:03 +00:00
mmichelson 50d4586dc3 Merged revisions 133104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines

Zap/pseudo is ten characters, but DAHDI/pseudo is
twelve. The strncmp call in next_channel should
account for this.


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133106 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23 19:07:56 +00:00
mmichelson 32e7cdb4b4 Merged revisions 133101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines

Update the "last" channel in next_channel in app_chanspy so
that the same pseudo channel isn't constantly returned.

related to issue #13124


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133102 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-23 18:58:37 +00:00
bbryant 8e222897e6 Janitor patch to change uses of sizeof to ARRAY_LEN
(closes issue #13054)
Reported by: pabelanger
Patches:
      ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130129 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11 18:09:35 +00:00
mmichelson 3225a60b74 Make change proposed by andrew53 on bugtracker
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127857 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-03 20:59:51 +00:00
mmichelson 8653f3a16f Thanks to a suggestion from seanbright, print a warning if the attachment
of the whisper or barge audiohooks fails.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127856 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-03 20:37:21 +00:00
mmichelson 7eeb744249 Fix build
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127852 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-03 20:23:57 +00:00
mmichelson a56376f775 Fix a crash when attempting to spy on an unbridged channel.
(closes issue #12986)
Reported by: andrew53



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127831 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-03 20:19:10 +00:00
jpeeler 490730a6b3 Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12 17:27:55 +00:00
mmichelson 35d48395b0 Merged revisions 118509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May 2008) | 11 lines

Russell noted to me that in the case that separate threads use their
own addressing system, the fix I made for issue 12376 does not guarantee
uniqueness to the datastores' uids. Though I know of no system that works
this way, I am going to change this right now to prevent trying to track
down some future bug that may occur and cause untold hours of debugging
time to track down.

The change involves using a global counter which increases with each new
chanspy_ds which is created. This guarantees uniqueness.


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118514 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-27 19:08:24 +00:00
mmichelson a334014b88 Merged revisions 118365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May 2008) | 14 lines

Add a unique id to the datastore allocated in app_chanspy since
it is possible that multiple spies may be listening to the same
channel.

(closes issue #12376)
Reported by: DougUDI
Patches:
      12376_chanspy_uid.diff uploaded by putnopvut (license 60)
Tested by: destiny6628

(closes issue #12243)
Reported by: atis


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118371 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-27 16:43:36 +00:00
mvanbaak c1210321e7 - revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- make data member of the ast_frame struct a named union instead of a void

Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.

The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.

This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data

Thanks russellb and kpfleming for the feedback.

(closes issue #12674)
Reported by: mvanbaak


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22 16:29:54 +00:00
mmichelson 83a1c36bfe Adding a new option to Chanspy(). The 'd' option allows for the spy to
press DTMF digits to switch between spying modes. Pressing 4 activates spy mode,
pressing 5 activates whisper mode, and pressing 6 activates barge mode. Use of
this feature overrides the normal operation of DTMF numbers. 

This feature is courtesy of Switchvox.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116522 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-14 22:15:12 +00:00