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r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines
Change the remote user agent session version variable
from an int to a uint64_t. This prevents potential comparison
problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could
not properly place a call on hold since the version in the
SDP of the re-INVITE to place the call on hold greatly
exceeded INT_MAX.
This also aligns with RFC 2327 better since it recommends
using an NTP timestamp for the version (which is a
64-bit number).
(closes issue #13531)
Reported by: sgofferj
Patches:
13531.patch uploaded by putnopvut (license 60)
Tested by: sgofferj
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r158188 | seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 lines
Fix one case where the application argument was not converted from a pipe to
a comma. This was causing problems with switch statements with empty expressions.
(closes issue #13901)
Reported by: smurfix
Patches:
20081118_bug13901.diff uploaded by seanbright (license 71)
Tested by: seanbright
Reviewed by: murf
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r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines
Merged revisions 158071 via svnmerge from
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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
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r158062 | jpeeler | 2008-11-20 11:37:31 -0600 (Thu, 20 Nov 2008) | 6 lines
(closes issue #12929)
Reported by: snyfer
This handles the case for a zero length file to attempt to be streamed. Instead of failing from not playing any data, go ahead and return success as ast_streamfile should consider playing nothing a success when there is nothing to play.
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r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines
Merged revisions 158053 via svnmerge from
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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
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r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov 2008) | 13 lines
Merged revisions 157859 via svnmerge from
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r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
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r157870 | tilghman | 2008-11-19 15:54:39 -0600 (Wed, 19 Nov 2008) | 10 lines
Two new functions, REALTIME_FIELD, and REALTIME_HASH, which should make
querying realtime from the dialplan a little more consistent and easy to use.
The original REALTIME function is preserved, for those who are already
accustomed to that interface.
(closes issue #13651)
Reported by: Corydon76
Patches:
20081119__bug13651__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage, Corydon76
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r157818 | twilson | 2008-11-19 13:25:14 -0600 (Wed, 19 Nov 2008) | 2 lines
Fix checking for CONFIG_STATUS_FILEINVALID so that modules don't crash upon trying to parse an invalid config
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r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines
make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
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r157639 | tilghman | 2008-11-18 19:02:45 -0600 (Tue, 18 Nov 2008) | 7 lines
Starting with a change to ensure that ast_verbose() preserves ABI compatibility
in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also
deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy
in core functions. va_copy() is C99, anyway, and we already require C99 for
other purposes, so this isn't really a big change anyway. This change also
simplifies some of the core ast_str_* functions.
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r157632 | mmichelson | 2008-11-18 18:59:48 -0600 (Tue, 18 Nov 2008) | 10 lines
If malloc returns NULL, we need to return NULL immediately or
else Asterisk will crash when attempting to dereference the NULL
pointer
(closes issue #13858)
Reported by: eliel
Patches:
astmm.c.patch uploaded by eliel (license 64)
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r157600 | seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 lines
Fix a few build problems on Solaris (and check for an md5 utility in
configure instead of the icky loop I was doing before).
(closes issue #13842)
Reported by: snuffy
Patches:
bug13842_20081106.diff uploaded by snuffy (license 35)
13842.diff uploaded by seanbright (license 71)
Tested by: snuffy
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r157592 | mmichelson | 2008-11-18 17:59:02 -0600 (Tue, 18 Nov 2008) | 10 lines
This change prevents a crash from occurring if res_musiconhold.so
is unloaded and then Asterisk is stopped. The problem was that
we are not unregistering the ast_moh_destroy function at exit.
(closes issue #13761)
Reported by: eliel
Patches:
res_musiconhold.c.patch uploaded by eliel (license 64)
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r157562 | mmichelson | 2008-11-18 17:28:23 -0600 (Tue, 18 Nov 2008) | 11 lines
Fix the logic for when delete=yes when IMAP storage
is in use so that the message is deleted from both
local and IMAP storage.
(closes issue #13642)
Reported by: jaroth
Patches:
deleteyes.patch uploaded by jaroth (license 50)
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r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines
Merged revisions 157503 via svnmerge from
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r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines
Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.
(closes issue #13878)
Reported by: nahuelgreco
Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
Tested by: putnopvut
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r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines
Based on Russell's advice on the asterisk-dev list, I have
changed from using a global lock in update_call_counter to
using the locks within the sip_pvt and sip_peer structures
instead.
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r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines
* Add a lock to be used in the update_call_counter function.
* Revert logic to mirror 1.4's in the sense that it will not allow
the call counter to dip below 0.
These two measures prevent potential races that could cause a SIP peer
to appear to be busy forever.
(closes issue #13668)
Reported by: mjc
Patches:
hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)
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r157366 | jpeeler | 2008-11-18 13:16:00 -0600 (Tue, 18 Nov 2008) | 14 lines
Merged revisions 157365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines
(closes issue #13899)
Reported by: akkornel
This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer.
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r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines
Merged revisions 157305 via svnmerge from
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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines
Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
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r157302 | murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines
(closes issue #13420)
Reported by: alex70
Patches:
13420.13539.patch uploaded by murf (license 17)
Tested by: murf, awk
This fixes two problems: a spurious linefeed insertion
probably left over from pre-precomment times. Only
generated when category had no previous comments.
The other problem: Insertions could get the line-numbering
out of whack and generate negative line numbers, causing
chunks of line numbers to be emitted, on the scale of the
number of lines up to that point in the file. In such cases,
abort the looping, and all is well.
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r157253 | tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines
Can't use items duplicated off the stack frame in an element returned from
a function: in these cases, we have to use the heap, or garbage will result.
(closes issue #13898)
Reported by: alecdavis
Patches:
20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: alecdavis
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r157167 | kpfleming | 2008-11-15 20:51:29 +0100 (Sat, 15 Nov 2008) | 2 lines
ensure that if a .i file (preprocessed source) is present, the .o file is made from it, not from the .c file (this only works because GNU makes respects the order the rules are defined)
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r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov 2008) | 13 lines
Merged revisions 157162-157163 via svnmerge from
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r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line
dist-clean should remove dependency information files as well
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r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line
when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice
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r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines
major update to doxygen configuration file:
1) update to doxygen 1.5.x style file, as used in trunk
2) tell doxygen where are header files are, so include-file processing can be done
3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function
4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields
5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit)
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r156962 | mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7 lines
Revision 155513 of chan_sip.c in trunk inadvertently
removed a very important line to set the "len" field
for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk
could do no meaningful processing of anything SIP-related
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r156916 | twilson | 2008-11-14 11:29:05 -0600 (Fri, 14 Nov 2008) | 2 lines
Use Mark's new ast_str_case_hash function instead of jumping through hoops to do insensitive case lookups
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r156817 | mmichelson | 2008-11-14 09:20:03 -0600 (Fri, 14 Nov 2008) | 18 lines
Merged revisions 156816 via svnmerge from
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r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines
If the prompt to reenter a voicemail password timed out, it
resulted in the password not being saved, even if the input matched
what you gave when first prompted to enter a new password. This is
because the return value of ast_readstring was checked, but not checked
properly.
This bug was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!
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r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines
Merged revisions 156755 via svnmerge from
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r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines
ast_waitfordigit() requires that the channel be up, for no good logical
reason. This prevents While/EndWhile from working within the "h"
extension.
Reported by: jgalarneau (for ABE C.2)
Fixed by: me
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r156649 | jpeeler | 2008-11-13 13:17:50 -0600 (Thu, 13 Nov 2008) | 6 lines
(closes issue #13891)
Reported by: smurfix
This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null.
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r156647 | tilghman | 2008-11-13 13:10:28 -0600 (Thu, 13 Nov 2008) | 3 lines
Command offsets were not changed correctly when the command syntax for
'pri set debug' was changed from 'pri debug'.
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