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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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sized string, this check in add_line is no longer valid.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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This started as work to fix the 'core show sysinfo'
CLI command but while working on it oej
pointed out that read_credentials did not compile neither.
So while being there, fix that as well.
Thanks for all the testing oej!
(closes issue #14129)
Reported by: ys
Tested by: oej, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines
Don't read into a buffer without first checking if a value is beyond the end.
(closes issue #13600)
Reported by: atis
Patches:
20090106__bug13600.diff.txt uploaded by Corydon76 (license 14)
Tested by: atis
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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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan 2009) | 5 lines
I am reverting the fix made in revision 168128 (and its upward merges)
after being contacted by Olle Johansson and being shown how this fix is
incorrect. Thanks to Olle for clearing this up for me.
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