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r169620 | russell | 2009-01-21 04:26:07 -0600 (Wed, 21 Jan 2009) | 10 lines
Fix a regression in TCP support.
This patch fixes a problem that caused chan_sip to think that every open TCP session
was to a remote address of 0.0.0.0:0.
(closes issue #14287)
Reported by: jamesgolovich
Patches:
bug-14287.diff.txt uploaded by jamesgolovich (license 176)
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r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines
Fix device state parsing issues for channel names with multiple slashes
The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.
In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.
(closes issue #14014)
Reported by: kebl0155
Patches:
14014_number2.patch uploaded by putnopvut (license 60)
Tested by: kebl0155
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r169557 | mmichelson | 2009-01-20 14:10:31 -0600 (Tue, 20 Jan 2009) | 19 lines
Convert the character pointers in a sip_request to be pointer offsets
When an ast_str expands to hold more data, any pointers that were pointing
to the data prior to the expansion will be pointing at invalid memory. This
change makes such pointers used in chan_sip.c instead be offsets from the
beginning of the string so that the same math may be applied no matter where
in memory the string resides.
To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added
to chan_sip.c so that given a sip_request and an offset, the string at that
offset is returned.
(closes issue #14220)
Reported by: riksta
Tested by: putnopvut
Review http://reviewboard.digium.com/r/126/
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r169510 | twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines
Make a proper builtin attended transfer to parking work
This is an ugly hack from 1.4 that allows the timeout callback from a parked
call to use the right channel name for the callback when the park is done with
a builtin attended transfer (that isn't completed early). This hasn't ever
worked in trunk and no one has complained yet, so eh.
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r169438 | kpfleming | 2009-01-19 15:42:46 -0600 (Mon, 19 Jan 2009) | 5 lines
ast_str_SQLGetData is *not* part of the ast_str API, it's part of the ast_odbc API and just happens to use an ast_str as the buffer; move all of it to res_odbc.c and res_odbc.h, renaming appropriately
along the way fix some minor coding style issues in strings.h and add some attribute_pure annotations to functions in the ast_str API
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r169211 | mmichelson | 2009-01-19 09:54:06 -0600 (Mon, 19 Jan 2009) | 21 lines
Merged revisions 169210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines
Prevent a crash in chan_local due to a potential NULL pointer dereference
Move the check for if both channels on a local_pvt have generators to below
where p->chan is checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL.
(closes issue #14189)
Reported by: sascha
Patches:
14189.patch uploaded by putnopvut (license 60)
Tested by: sascha
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r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009) | 19 lines
Merged revisions 168716 via svnmerge from
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r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines
Convert call to park_call_full to masq_park_call_announce
Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded
parking, otherwise we will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash.
(closes issue #14215)
Reported by: waverly360
Tested by: otherwiseguy
(closes issue #14228)
Reported by: kobaz
Tested by: otherwiseguy
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r168976 | mmichelson | 2009-01-16 16:43:09 -0600 (Fri, 16 Jan 2009) | 26 lines
Merged revisions 168975 via svnmerge from
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r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines
Account for possible NULL pointer when we receive a 408 in response to a REGISTER
It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash
This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER
(closes issue #14211)
Reported by: aborghi
Patches:
14211.diff uploaded by putnopvut (license 60)
Tested by: aborghi
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r168898 | mmichelson | 2009-01-16 13:54:39 -0600 (Fri, 16 Jan 2009) | 26 lines
Fix a logic error that occur when using the timerfd interface
This sequence of events posed a problem
timerfd_timer_open
timerfd_timer_enable_continuous
timerfd_timer_set_rate
timerfd_timer_disable_continuous
The reason was that the timing module was written under the assumption
that timerfd_timer_set_rate would not be called between enabling and
disabling continuous mode. What happened in this situation was that
timerfd_timer_enable_continuous saved off our previously set timer (in this
situation a 0 timer, meaning it never runs out). Then timerfd_timer_disable_continuous
would restore this 0 timer, even though it logically should set the timer to be whatever
was set in timerfd_timer_set_rate.
Now the behavior in timerfd_timer_set_rate is to overwrite the saved timer that may
or may not have been set in timerfd_timer_enable_continuous. Even if
timerfd_timer_enable_continuous has not been previously called, this will not harm the
operation.
Thanks to Terry Wilson for discovering the problem and giving me a really great debug
capture that pointed out the problem clearly
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r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) | 20 lines
Merged revisions 168745 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines
This patch fixes a problem where a goto (or jump, in this case)
fails a consistency check because it can't find a matching
extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern
into a regex and uses the regex code to determine the match.
I tested using the AEL code the user supplied, and now,
the consistency check passes.
(closes issue #14141)
Reported by: dimas
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r168737 | murf | 2009-01-15 13:54:59 -0700 (Thu, 15 Jan 2009) | 16 lines
This patch allows null args in ast_expr2 func calls, and fixes commas being converted to pipes, which was 1.4 type stuff.
If the user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld);
then it won't complain about the empty arg (c,,...) and fabled's patch
won't let it swap the commas for pipes.
Ran it thru my dialplan and no complaints.
(closes issue #14169)
Reported by: fabled
Patches:
function-argument-separator-fix.diff uploaded by fabled (license 448)
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r168734 | kpfleming | 2009-01-15 14:18:53 -0600 (Thu, 15 Jan 2009) | 5 lines
remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC
this stops modules from being linked against both sets of libraries on systems that have both installed
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r168725 | mmichelson | 2009-01-15 13:00:06 -0600 (Thu, 15 Jan 2009) | 17 lines
Remove an unneeded condition for line addition to a SIP request/response
In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically
allocated buffer to hold the text of the request. There was a check in the
add_line function to not attempt to write the line into the buffer if we
did not have room for it.
In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str
structure is used to hold the text. Since it may grow to fit an arbitrarily
sized string, this check in add_line is no longer valid.
I found this oddity while attempting to fix issue #14220; however, I do not
believe that this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be printed had this
condition been satisfied.
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r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines
Add a missing unlock and properly handle the 'maxusers' setting on MeetMe
conferences. We were using the 'user number' field to compare against the
maximum allowed users, which works assuming users with lower user numbers
didn't leave the conference.
(closes issue #14117)
Reported by: sergedevorop
Patches:
20090114__bug14117-2.diff.txt uploaded by seanbright (license 71)
Tested by: sergedevorop
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r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines
Merged revisions 168628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing
the datastore from the incorrect channel, so this was fixed.
* Most datastore operations regarding the queue_transfer datastore
were being done without the channel locked, so proper channel locking
was added, too.
(closes issue #14086)
Reported by: ZX81
Patches:
14086v2.patch uploaded by putnopvut (license 60)
Tested by: ZX81, festr
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r168615 | seanbright | 2009-01-14 15:58:26 -0500 (Wed, 14 Jan 2009) | 16 lines
Merged revisions 168614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan 2009) | 9 lines
Update autosupport script to supply info for both Zaptel and DAHDI in 1.4 and
be sure to run dahdi_test in 1.6.x and trunk instead of zttest.
(closes issue #14132)
Reported by: dsedivec
Patches:
asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
asterisk-trunk-autosupport.patch uploaded by dsedivec (license 638)
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r168599 | mmichelson | 2009-01-14 10:20:37 -0600 (Wed, 14 Jan 2009) | 15 lines
Blocked revisions 168598 via svnmerge
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r168598 | mmichelson | 2009-01-14 10:19:26 -0600 (Wed, 14 Jan 2009) | 8 lines
Fix a logic error I found while searching through chan_agent.c
I found that the allow_multiple_logins function would never return
0 due to an incorrect comparison being used when traversing the
list of agents. While I was modifying this function, I also did
a little bit of coding guidelines cleanup, too.
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r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines
Merged revisions 168593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
Don't overflow when paging more than 128 extensions
The number of available slots for calls in app_page was hardcoded to 128.
Proper bounds checking was not in place to enforce this limit, so if more than
128 extensions were passed to the Page() app, Asterisk would crash. This patch
instead dynamically allocates memory for the ast_dial structures and removes
the (non-functional) arbitrary limit.
This issue would have special importance to anyone who is dynamically creating
the argument passed to the Page application and allowing more than 128
extensions to be added by an outside user via some external interface.
The patch posted by a_villacis was slightly modified for some coding guidelines
and other cleanups. Thanks, a_villacis!
(closes issue #14217)
Reported by: a_villacis
Patches:
20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
Tested by: otherwiseguy
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r168579 | mmichelson | 2009-01-13 16:30:59 -0600 (Tue, 13 Jan 2009) | 13 lines
Clarify a message that app_queue prints and change to a debug-level message
The "No one is answering..." verbose message contained 3 numbers that were not
explained in any way to whoever was viewing the message. It is more helpful now
since the message explains what the numbers mean. Also, the message has been
downgraded to "DEBUG" level.
(closes issue #14172)
Reported by: caio1982
Patches:
queue_answering_debug.diff uploaded by caio1982 (license 22)
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r168575 | mmichelson | 2009-01-13 15:18:13 -0600 (Tue, 13 Jan 2009) | 13 lines
Allow specifying a port number in the user portion of a register => line in sip.conf
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
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r168523 | mmichelson | 2009-01-12 17:12:30 -0600 (Mon, 12 Jan 2009) | 11 lines
bump the verbosity of a message in srv.c up by one. It used to be
at this level prior to a large patch merge which converted ast_verbose
calls to ast_verb
(closes issue #14221)
Reported by: jcovert
Patches:
srv.c.patch uploaded by jcovert (license 551)
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r168508 | jpeeler | 2009-01-12 14:53:04 -0600 (Mon, 12 Jan 2009) | 15 lines
Merged revisions 168507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines
(closes issue #12269)
Reported by: IgorG
Tested by: denisgalvao
This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock.
Review: http://reviewboard.digium.com/r/35/
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