Commit Graph

23606 Commits

Author SHA1 Message Date
Damien Wedhorn
f795062662 Add group and namedgroup pickup to skinny
Above says it all. Code by snuff, cleaned up by me. 

Review: https://reviewboard.asterisk.org/r/2246/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-06 21:09:43 +00:00
Damien Wedhorn
bacc5e6604 Rewrite skinny dialing to remove threaded simpleswitch
This rewrite changes skinny dialing from the threaded simpleswitch
to a scheduled timeout approach. There were some underlying issues
with the threaded simple switch with occasional corruption and
possible segfaults.

Review: https://reviewboard.asterisk.org/r/2240/
........

Merged revisions 378622 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-06 20:45:12 +00:00
Automerge script
2765de67b6 Merged revisions 378593 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378593 | jrose | 2013-01-04 17:14:54 -0600 (Fri, 04 Jan 2013) | 23 lines
  
  res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
  
  Under some circumstances, libsrtp's srtp_create function deallocates memory that
  it wasn't initially responsible for allocating. Because we weren't initially
  aware of this behavior, this memory was still used in spite of being unallocated
  during the course of the srtp_unprotect function. A while back I made a patch
  which would set this value to NULL, but that exposed a possible condition where
  we would then try to check a member of the struct which would cause a segfault.
  In order to address these problems, ast_srtp_unprotect will now set an error value
  when it ends without a valid SRTP session which will result in the caller of
  srtp_unprotect observing this error and hanging up the relevant channel instead of
  trying to keep using the invalid session address.
  
  (closes issue ASTERISK-20499)
  Reported by: Tootai
  Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header
  ........
  
  Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 23:19:39 +00:00
Jonathan Rose
9d5f6e050e res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
Under some circumstances, libsrtp's srtp_create function deallocates memory that
it wasn't initially responsible for allocating. Because we weren't initially
aware of this behavior, this memory was still used in spite of being unallocated
during the course of the srtp_unprotect function. A while back I made a patch
which would set this value to NULL, but that exposed a possible condition where
we would then try to check a member of the struct which would cause a segfault.
In order to address these problems, ast_srtp_unprotect will now set an error value
when it ends without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant channel instead of
trying to keep using the invalid session address.

(closes issue ASTERISK-20499)
Reported by: Tootai
Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header
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Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 23:14:54 +00:00
Automerge script
77150eecd9 Merged revisions 378565,378585 via svnmerge from
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  r378565 | elguero | 2013-01-04 15:20:12 -0600 (Fri, 04 Jan 2013) | 27 lines
  
  Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
  
  On a multihomed server when sending a NOTIFY message, we were not figuring out
  which network should be used to contact the peer.
  
  This patch fixes the problem by calling ast_sip_ouraddrfor() and then
  build_via() so that our NOTIFY message contains the correct IP address.
  
  Also, a debug message is being added to help follow the call-id changes that
  occur.  This was helpful for confirming that the IP address was set properly
  since the call-id contains the IP address.  It also will be helpful for
  troubleshooting purposes when following a call in the debug logs.
  
  (closes issue ASTERISK-20805)
  Reported by: Bryan Hunt
  Tested by: Bryan Hunt, Michael L. Young
  Patches:
      asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2255/
  ........
  
  Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378585 | kmoore | 2013-01-04 16:19:16 -0600 (Fri, 04 Jan 2013) | 13 lines
  
  Fix pjproject compilation in certain circumstances
  
  On a fresh checkout of Asterisk 11, running make before ./configure
  could cause the pjproject subdirectory to get in an odd state that
  would prevent compilation. This patch by Tilghman prevents that from
  occurring.
  
  (closes issue ASTERISK-20681)
  Patch-by: Tilghman Lesher
  ........
  
  Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 22:19:35 +00:00
Kinsey Moore
9e814816cb Fix pjproject compilation in certain circumstances
On a fresh checkout of Asterisk 11, running make before ./configure
could cause the pjproject subdirectory to get in an odd state that
would prevent compilation. This patch by Tilghman prevents that from
occurring.

(closes issue ASTERISK-20681)
Reported by: Dinesh Ramjuttun
Tested by: danilo borges, Steve Lang
patches:
  20121208__ccar_solved.diff.txt uploaded by Tilghman Lesher (license 5003)
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Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 22:19:16 +00:00
Michael L. Young
209373262d Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
On a multihomed server when sending a NOTIFY message, we were not figuring out
which network should be used to contact the peer.

This patch fixes the problem by calling ast_sip_ouraddrfor() and then
build_via() so that our NOTIFY message contains the correct IP address.

Also, a debug message is being added to help follow the call-id changes that
occur.  This was helpful for confirming that the IP address was set properly
since the call-id contains the IP address.  It also will be helpful for
troubleshooting purposes when following a call in the debug logs.

(closes issue ASTERISK-20805)
Reported by: Bryan Hunt
Tested by: Bryan Hunt, Michael L. Young
Patches:
    asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2255/
........

Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 21:20:12 +00:00
Automerge script
98fe918625 Merged revisions 378557 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378557 | file | 2013-01-04 15:18:07 -0600 (Fri, 04 Jan 2013) | 11 lines
  
  Don't pass STUN packets through the SRTP unprotect function.
  
  (closes issue AST-1036)
  Reported by: jbigelow
  ........
  
  Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 21:18:39 +00:00
Joshua Colp
4838d6ff68 Don't pass STUN packets through the SRTP unprotect function.
(closes issue AST-1036)
Reported by: jbigelow
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Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 21:18:07 +00:00
Automerge script
cd0aef75ca Merged revisions 378543 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378543 | lathama | 2013-01-04 10:44:33 -0600 (Fri, 04 Jan 2013) | 6 lines
  
  Doxygen Cleanups
  
  Baseline clean up of formating to make room for extended documentation
  
  (issue ASTERISK-20259)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 17:20:32 +00:00
Andrew Latham
a30e74da4f Doxygen Cleanups
Baseline clean up of formating to make room for extended documentation

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-04 16:44:33 +00:00
Automerge script
7980f15ada Merged revisions 378516 via svnmerge from
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  r378516 | elguero | 2013-01-03 16:14:20 -0600 (Thu, 03 Jan 2013) | 25 lines
  
  Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present
  
  When the "h" extension is present within the context of the queue, all calls
  are being reported COMPLETECALLER even when the agent is hanging up the call.
  
  This patch checks to see if the agent hung-up or not instead of only relying on
  checking if the queue (caller) channel hung-up or not.  It would appear that
  having the h extension in the mix, the pbx goes to the h extension,
  "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.
  
  (closes issue ASTERISK-20743)
  Reported by: call
  Tested by: call, Michael L. Young
  Patches: 
      asterisk-20743-q-cmplt-caller.diff 
                                       uploaded by Michael L. Young (license 5026)
  
  Review: https://reviewboard.asterisk.org/r/2256/
  ........
  
  Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378515 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 22:19:46 +00:00
Michael L. Young
8c0d7005f3 Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present
When the "h" extension is present within the context of the queue, all calls
are being reported COMPLETECALLER even when the agent is hanging up the call.

This patch checks to see if the agent hung-up or not instead of only relying on
checking if the queue (caller) channel hung-up or not.  It would appear that
having the h extension in the mix, the pbx goes to the h extension,
"hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.

(closes issue ASTERISK-20743)
Reported by: call
Tested by: call, Michael L. Young
Patches: 
    asterisk-20743-q-cmplt-caller.diff 
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2256/
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Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 22:14:20 +00:00
Automerge script
d359e1225e Merged revisions 378488 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378488 | rmudgett | 2013-01-03 13:42:54 -0600 (Thu, 03 Jan 2013) | 15 lines
  
  chan_agent: Fix wrapup time wait response.
  
  * Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup 
  time expires.  agent_cont_sleep() had tried but returned the wrong value 
  to stop waiting.  
  
  * Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void
  pointer for better type safety.
  ........
  
  Merged revisions 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378487 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 20:19:37 +00:00
Richard Mudgett
1d685bd28c chan_agent: Fix wrapup time wait response.
* Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup 
time expires.  agent_cont_sleep() had tried but returned the wrong value 
to stop waiting.  

* Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void
pointer for better type safety.
........

Merged revisions 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 19:42:54 +00:00
Automerge script
f824b845ce Merged revisions 378458,378460 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378458 | rmudgett | 2013-01-03 12:47:29 -0600 (Thu, 03 Jan 2013) | 18 lines
  
  chan_agent: Misc code cleanup.
  
  * Fix off-nominal path resource cleanup in agent_request().
  
  * Create agent_pvt_destroy() to eliminate inlined versions in many places.
  
  * Pull invariant code out of loop in add_agent().
  
  * Remove redundant module user references in login_exec().
  
  * Remove unused struct agent_pvt logincallerid[] member.
  ........
  
  Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378457 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378460 | kmoore | 2013-01-03 12:51:43 -0600 (Thu, 03 Jan 2013) | 13 lines
  
  Add missing test event
  
  This test event was missing from channel.c causing the dial_LS_options
  test to fail intermittently because of a race condition where most code
  paths emitted the test event but this one did not. The dial_LS_options
  test should stop bouncing now.
  ........
  
  Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378459 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 19:19:40 +00:00
Kinsey Moore
ad582f9aef Add missing test event
This test event was missing from channel.c causing the dial_LS_options
test to fail intermittently because of a race condition where most code
paths emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
........

Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 18:51:43 +00:00
Richard Mudgett
da7c2e3ffe chan_agent: Misc code cleanup.
* Fix off-nominal path resource cleanup in agent_request().

* Create agent_pvt_destroy() to eliminate inlined versions in many places.

* Pull invariant code out of loop in add_agent().

* Remove redundant module user references in login_exec().

* Remove unused struct agent_pvt logincallerid[] member.
........

Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378457 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 18:47:29 +00:00
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06b4ef8317 Merged revisions 378429 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk

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  r378429 | rmudgett | 2013-01-03 11:48:14 -0600 (Thu, 03 Jan 2013) | 10 lines
  
  chan_agent: Fix agent_indicate() locking.
  
  Avoid deadlock potential with local channels and simplify the locking.
  ........
  
  Merged revisions 378427 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378428 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 18:20:07 +00:00
Richard Mudgett
11571714fe chan_agent: Fix agent_indicate() locking.
Avoid deadlock potential with local channels and simplify the locking.
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2013-01-03 17:48:14 +00:00
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01250e2585 Merged revisions 378410,378412,378414 via svnmerge from
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  r378410 | mjordan | 2013-01-03 09:37:31 -0600 (Thu, 03 Jan 2013) | 13 lines
  
  Prevent crashes in res_xmpp when receiving large messages
  
  Similar to r378287, res_xmpp was marshaling data read from an external source
  onto the stack. For a sufficiently large message, this could cause a stack
  overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by
  removing the stack allocation, as it was unnecessary.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes
  ........
  
  Merged revisions 378409 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378412 | file | 2013-01-03 09:40:21 -0600 (Thu, 03 Jan 2013) | 11 lines
  
  Prevent exhaustion of system resources through exploitation of event cache
  
  This patch changes res_xmpp to no longer cache events under certain circumstances.
  
  (issue ASTERISK-20175)
  Reported by: Russell Bryant, Leif Madsen, Joshua Colp
  Tested by: kmoore
  ........
  
  Merged revisions 378411 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378414 | tilghman | 2013-01-03 10:04:11 -0600 (Thu, 03 Jan 2013) | 11 lines
  
  Add aliases to the Directory.
  
  This is an interesting feature that allows additional strings to be used to
  search the Directory, primarily intended to be used with nicknames, but could
  be used with affiliations and the like.  Because the name field is used in
  more than one place (such as email notifications), it is important that these
  additional strings not be placed in the name field, but be specified
  separately.
  
  Review: https://reviewboard.asterisk.org/r/2244/
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 16:19:40 +00:00
Tilghman Lesher
7739ed9989 Add aliases to the Directory.
This is an interesting feature that allows additional strings to be used to
search the Directory, primarily intended to be used with nicknames, but could
be used with affiliations and the like.  Because the name field is used in
more than one place (such as email notifications), it is important that these
additional strings not be placed in the name field, but be specified
separately.

Review: https://reviewboard.asterisk.org/r/2244/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 16:04:11 +00:00
Joshua Colp
099cd07887 Prevent exhaustion of system resources through exploitation of event cache
This patch changes res_xmpp to no longer cache events under certain circumstances.

(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 15:40:21 +00:00
Matthew Jordan
8bf1f1745b Prevent crashes in res_xmpp when receiving large messages
Similar to r378287, res_xmpp was marshaling data read from an external source
onto the stack. For a sufficiently large message, this could cause a stack
overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by
removing the stack allocation, as it was unnecessary.

(issue ASTERISK-20658)
Reported by: wdoekes
........

Merged revisions 378409 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-03 15:37:31 +00:00
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b375abae50 Merged revisions 378374,378377,378384 via svnmerge from
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  r378374 | rmudgett | 2013-01-02 15:23:16 -0600 (Wed, 02 Jan 2013) | 33 lines
  
  Fix AMI redirect action with two channels failing to redirect both channels.
  
  The AMI redirect action can fail to redirect two channels that are bridged
  together.  There is a race between the AMI thread redirecting the two
  channels and the bridge thread noticing that a channel is hungup from the
  redirects.
  
  * Made the bridge wait for both channels to be redirected before exiting.
  
  * Made the AMI redirect check that all required headers are present before
  proceeding with the redirection.
  
  * Made the AMI redirect require that any supplied ExtraChannel exist
  before proceeding.  Previously the code fell back to a single channel
  redirect operation.
  
  (closes issue ASTERISK-18975)
  Reported by: Ben Klang
  
  (closes issue ASTERISK-19948)
  Reported by: Brent Dalgleish
  Patches:
        jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
  Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
  
  Review: https://reviewboard.asterisk.org/r/2243/
  ........
  
  Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378358 from http://svn.asterisk.org/svn/asterisk/branches/11
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  r378377 | mjordan | 2013-01-02 16:10:32 -0600 (Wed, 02 Jan 2013) | 24 lines
  
  Prevent crashes from occurring when reading from data sources with large values
  
  When reading configuration data from an Asterisk .conf file or when pulling
  data from an Asterisk RealTime backend, Asterisk was copying the data on the
  stack for manipulation. Unfortunately, it is possible to read configuration
  data or realtime data from some data source that provides a large blob of
  characters. This could potentially cause a crash via a stack overflow.
  
  This patch prevents large sets of data from being read from an ARA backend or
  from an Asterisk conf file.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes
  Tested by: wdoekes, mmichelson
  patches:
   * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
   * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
  ........
  
  Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378376 from http://svn.asterisk.org/svn/asterisk/branches/11
................
  r378384 | mjordan | 2013-01-02 16:19:32 -0600 (Wed, 02 Jan 2013) | 11 lines
  
  Clean up app_mysql's application entry points to properly parse arguments
  
  When parsing arguments, application entry points should not attempt to
  directly modify the parameters to the function. This patch properly duplicates
  the passed in parameters before attempting to parse them.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes
  patches:
    issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02 22:19:49 +00:00
Matthew Jordan
473ec0b1c3 Clean up app_mysql's application entry points to properly parse arguments
When parsing arguments, application entry points should not attempt to
directly modify the parameters to the function. This patch properly duplicates
the passed in parameters before attempting to parse them.

(issue ASTERISK-20658)
Reported by: wdoekes
patches:
  issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02 22:19:32 +00:00
Matthew Jordan
89f9e077d7 Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.

This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.

(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
 * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
 * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
........

Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02 22:10:32 +00:00
Richard Mudgett
5601f3be43 Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged
together.  There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.

* Made the bridge wait for both channels to be redirected before exiting.

* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.

* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding.  Previously the code fell back to a single channel
redirect operation.

(closes issue ASTERISK-18975)
Reported by: Ben Klang

(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
      jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode

Review: https://reviewboard.asterisk.org/r/2243/
........

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2013-01-02 21:23:16 +00:00
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................
  r378322 | mjordan | 2013-01-02 12:11:59 -0600 (Wed, 02 Jan 2013) | 33 lines
  
  Prevent exhaustion of system resources through exploitation of event cache
  
  Asterisk maintains an internal cache for devices in the event subsystem. The
  device state cache holds the state of each device known to Asterisk, such that
  consumers of device state information can query for the last known state for
  a particular device, even if it is not part of an active call. The concept of
  a device in Asterisk can include entities that do not have a physical
  representation. One way that this occurred was when anonymous calls are allowed
  in Asterisk. A device was automatically created and stored in the cache for
  each anonymous call that occurred; this was possible in the SIP and IAX2
  channel drivers and through channel drivers that utilized the
  res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
  are never removed from the system, allowing anonymous calls to potentially
  exhaust a system's resources.
  
  This patch changes the event cache subsystem and device state management to
  no longer cache devices that are not associated with a physical entity.
  
  (issue ASTERISK-20175)
  Reported by: Russell Bryant, Leif Madsen, Joshua Colp
  Tested by: kmoore
  patches:
    event-cachability-3.diff uploaded by jcolp (license 5000)
  ........
  
  Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
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................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02 18:20:03 +00:00
Matthew Jordan
8fb5bdce9a Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.

This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.

(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
  event-cachability-3.diff uploaded by jcolp (license 5000)
........

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........

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2013-01-02 18:11:59 +00:00
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................
  r378288 | mjordan | 2013-01-02 09:39:42 -0600 (Wed, 02 Jan 2013) | 36 lines
  
  Resolve crashes due to large stack allocations when using TCP
  
  Asterisk had several places where messages received over various network
  transports may be copied in a single stack allocation. In the case of TCP,
  since multiple packets in a stream may be concatenated together, this can
  lead to large allocations that overflow the stack.
  
  This patch modifies those portions of Asterisk using TCP to either
  favor heap allocations or use an upper bound to ensure that the stack will not
  overflow:
   * For SIP, the allocation now has an upper limit
   * For HTTP, the allocation is now a heap allocation instead of a stack
     allocation
   * For XMPP (in res_jabber), the allocation has been eliminated since it was
     unnecesary.
  
  Note that the HTTP portion of this issue was independently found by Brandon
  Edwards of Exodus Intelligence.
  
  (issue ASTERISK-20658)
  Reported by: wdoekes, Brandon Edwards
  Tested by: mmichelson, wdoekes
  patches:
    ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
    issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
    issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
  ........
  
  Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
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................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-02 16:20:01 +00:00
Matthew Jordan
1fb06fde95 Resolve crashes due to large stack allocations when using TCP
Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.

This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
 * For SIP, the allocation now has an upper limit
 * For HTTP, the allocation is now a heap allocation instead of a stack
   allocation
 * For XMPP (in res_jabber), the allocation has been eliminated since it was
   unnecesary.

Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.

(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
  ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
  issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
  issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
........

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2013-01-02 15:39:42 +00:00
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  r378259 | lathama | 2013-01-01 13:02:52 -0600 (Tue, 01 Jan 2013) | 5 lines
  
  Add UUID packages now required to configure
  
  In ASTERISK-20726 UUID was added to Asterisk.  This commit is to add the dependancies to the install script
........


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2013-01-01 19:19:12 +00:00
Andrew Latham
2db81d7fc2 Add UUID packages now required to configure
In ASTERISK-20726 UUID was added to Asterisk.  This commit is to add the dependancies to the install script



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-01 19:02:52 +00:00
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  r378248 | seanbright | 2013-01-01 11:03:59 -0600 (Tue, 01 Jan 2013) | 2 lines
  
  Bail out early when building an ast_trans_pvt and the translator doesn't supply a 'newpvt'
........
  r378249 | seanbright | 2013-01-01 11:10:42 -0600 (Tue, 01 Jan 2013) | 2 lines
  
  Revert 378248.  I changed the logic of this function unitentionally, pointed out by file.
........


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2013-01-01 17:20:00 +00:00
Sean Bright
cdf8498784 Revert 378248. I changed the logic of this function unitentionally, pointed out by file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-01 17:10:42 +00:00
Sean Bright
9c20603dfc Bail out early when building an ast_trans_pvt and the translator doesn't supply a 'newpvt'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-01 17:03:59 +00:00
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................
  r378220 | kmoore | 2012-12-31 08:46:06 -0600 (Mon, 31 Dec 2012) | 18 lines
  
  Ensure chan_sip rejects encrypted streams without crypto info
  
  This ensures that Asterisk rejects encrypted media streams (RTP/SAVP
  audio and video) that are missing cryptographic keys and ensures that
  the incoming SDP is consistent with RFC4568 as far as having a crypto
  attribute present for any SAVP streams.
  
  Review: https://reviewboard.asterisk.org/r/2204/
  ........
  
  Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
  Merged revisions 378218 from http://svn.asterisk.org/svn/asterisk/branches/10
  ........
  
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-31 15:21:42 +00:00
Kinsey Moore
32472eca70 Ensure chan_sip rejects encrypted streams without crypto info
This ensures that Asterisk rejects encrypted media streams (RTP/SAVP
audio and video) that are missing cryptographic keys and ensures that
the incoming SDP is consistent with RFC4568 as far as having a crypto
attribute present for any SAVP streams.

Review: https://reviewboard.asterisk.org/r/2204/
........

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2012-12-31 14:46:06 +00:00
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................
  r378166 | rmudgett | 2012-12-20 15:51:03 -0600 (Thu, 20 Dec 2012) | 8 lines
  
  Give the causes[] a struct name.
  ........
  
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2012-12-20 22:18:25 +00:00
Richard Mudgett
255d82aae3 Give the causes[] a struct name.
........

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2012-12-20 21:51:03 +00:00
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................
  r378122 | kmoore | 2012-12-18 11:48:36 -0600 (Tue, 18 Dec 2012) | 17 lines
  
  Add test events for time limit-related hangups
  
  This patch adds hangup-related test events in order to support testing
  of time-limited bridges. This aids in testing the S() and L() bridge
  options.
  
  (issue SWP-4713)
  ........
  
  Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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  Merged revisions 378120 from http://svn.asterisk.org/svn/asterisk/branches/10
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2012-12-18 18:18:44 +00:00
Kinsey Moore
dec9ee2f04 Add test events for time limit-related hangups
This patch adds hangup-related test events in order to support testing
of time-limited bridges. This aids in testing the S() and L() bridge
options.

(issue SWP-4713)
........

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2012-12-18 17:48:36 +00:00
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  r378091 | rmudgett | 2012-12-17 17:02:54 -0600 (Mon, 17 Dec 2012) | 22 lines
  
  Make chan_local module references tied to local_pvt lifetime.
  
  The chan_local module references were manually tied to the existence of
  the ;1 and ;2 channel links.
  
  * Made chan_local module references tied to the existence of the local_pvt
  structure as well as automatically take care of the module references.
  
  * Tweaked the wording of the local_fixup() failure warning message to make
  sense.
  
  Review: https://reviewboard.asterisk.org/r/2181/
  ........
  
  Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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................
  r378095 | rmudgett | 2012-12-17 17:10:42 -0600 (Mon, 17 Dec 2012) | 11 lines
  
  Fix potential double free when unloading a module.
  ........
  
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2012-12-17 23:18:40 +00:00
Richard Mudgett
5e2c52a546 Fix potential double free when unloading a module.
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2012-12-17 23:10:42 +00:00
Richard Mudgett
23b94b9211 Make chan_local module references tied to local_pvt lifetime.
The chan_local module references were manually tied to the existence of
the ;1 and ;2 channel links.

* Made chan_local module references tied to the existence of the local_pvt
structure as well as automatically take care of the module references.

* Tweaked the wording of the local_fixup() failure warning message to make
sense.

Review: https://reviewboard.asterisk.org/r/2181/
........

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2012-12-17 23:02:54 +00:00
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  r378081 | rmudgett | 2012-12-17 15:22:21 -0600 (Mon, 17 Dec 2012) | 7 lines
  
  chan_local: Parse dial string consistently.
  
  * Fix local_alloc() unexpected limitation of exten and context length from
  a combined length of 80 characters to a normal 80 characters each.
  
  * Made local_alloc() and local_devicestate() parse the same way.
........


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2012-12-17 22:17:57 +00:00
Richard Mudgett
0494456ae6 chan_local: Parse dial string consistently.
* Fix local_alloc() unexpected limitation of exten and context length from
a combined length of 80 characters to a normal 80 characters each.

* Made local_alloc() and local_devicestate() parse the same way.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-17 21:22:21 +00:00
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................
  r378072 | rmudgett | 2012-12-17 14:34:25 -0600 (Mon, 17 Dec 2012) | 9 lines
  
  chan_local: Misc lock and ref tweaks.
  
  * awesome_locking() does not need to thrash the pvt lock as much.
  
  * local_setoption() does not need to check for NULL pvt on cleanup since
  it will never be NULL.
  
  * Made ref the pvt before locking for consistency.
................
  r378074 | qwell | 2012-12-17 14:59:51 -0600 (Mon, 17 Dec 2012) | 10 lines
  
  Make libasteriskssl.so symlink use a relative path.
  
  This was causing issues when using DESTDIR, since the path to which the link
  pointed is not likely to exist (and not useful to exist) on the target system.
  
  (issue ASTNOW-284)
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2012-12-17 21:18:01 +00:00
Jason Parker
7ecf0aa94b Make libasteriskssl.so symlink use a relative path.
This was causing issues when using DESTDIR, since the path to which the link
pointed is not likely to exist (and not useful to exist) on the target system.

(issue ASTNOW-284)
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Merged revisions 378073 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-17 20:59:51 +00:00