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r170677 | mmichelson | 2009-01-23 14:23:00 -0600 (Fri, 23 Jan 2009) | 22 lines
Merged revisions 170671 via svnmerge from
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r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan 2009) | 14 lines
Update contrib/i18n.testsuite.conf to not use deprecated syntax
* Convert Wait,1 to Wait(1)
* Convert SetLanguage to Set(CHANNEL(language))
* Use 'n' for all priorities beyond the first
Also added test for Chinese numbers, too.
(closes issue #14320)
Reported by: dant
Patches:
i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license 670)
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r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines
Merged revisions 170648 via svnmerge from
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r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines
When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them.
(closes issue #14249)
Reported by: RadicAlish
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r170498 | file | 2009-01-23 13:32:26 -0400 (Fri, 23 Jan 2009) | 4 lines
Reset the ast_str used for escape substitution. We need to do this since it is a thread local variable that may contain the value of a previous substitution.
(closes issue #14312)
Reported by: pj
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r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan 2009) | 36 lines
Merged revisions 170392 via svnmerge from
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r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines
Fix broken call pickup
There was a subtle change in ast_do_masquerade which
resulted in failed attempts to pickup calls. The problem
was that the value of the AST_FLAG_OUTGOING flag was
copied from the clone to the original channel. In the case
of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting
to execute the pickup.
Because this flag was not set, when ast_read came across
an answer frame, it ignored it. The result of this was that
the calling channel was never properly answered.
This fix changes the behavior in ast_do_masquerade to set
the flags on the original channel to the union of the flags
on the clone channel. This way, if the AST_FLAG_OUTGOING
flag is set on either of the two channels involved in the
masquerade, the resulting channel will have the flag set
as well.
(closes issue #14206)
Reported by: francesco_r
Patches:
14206.patch uploaded by putnopvut (license 60)
Tested by: francesco_r, aragon, putnopvut
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r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines
Merged revisions 170147 via svnmerge from
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r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines
If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
(closes issue #14282)
Reported by: cheesegrits
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r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) | 13 lines
Merged revisions 170050 via svnmerge from
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r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 lines
Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness.
(closes issue #14011)
Reported by: dveiga
Patches:
pbx.c.patch uploaded by dveiga (license 665)
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r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines
Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop.
(closes issue #14304)
Reported by: jcovert
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r169794 | mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17 lines
Fix a crash when saying certain numbers in Chinese
This commit fixes a crash that was occurring when attempting to
say a number between 10000 and 100000 due to dividing by 0.
This also removes some places where a "zero" is spoken when it
should not be.
(closes issue #14291)
Reported by: dant
Patches:
say.c-14291.diff uploaded by dant (license 670)
Tested by: dant
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r169673 | murf | 2009-01-21 10:21:40 -0700 (Wed, 21 Jan 2009) | 14 lines
This patch corrects a segfault reported in 14289,
due to a null ptr being refd.
Yes, seanbright is right in the bug comments, that is the fix.
Sorry for this oversight; I guess my personal
usage didn't have this happen!
murf
(closes issue #14289)
Reported by: jamesgolovich
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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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This must have inadvertantly got merged in sometime as the code doesn't handle
this option, and configs/sip.conf.sample doesn't mention it as available. So
just remove it from the WARNING message
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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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r169211 | mmichelson | 2009-01-19 09:54:06 -0600 (Mon, 19 Jan 2009) | 21 lines
Merged revisions 169210 via svnmerge from
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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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r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) | 20 lines
Merged revisions 168745 via svnmerge from
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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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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
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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
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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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r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines
Merged revisions 168593 via svnmerge from
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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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