https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
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r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
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r120731 | russell | 2008-06-05 13:01:25 -0500 (Thu, 05 Jun 2008) | 2 lines
Add the UPGRADE.txt file from Asterisk 1.2, for handy reference.
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r120671 | russell | 2008-06-05 11:38:52 -0500 (Thu, 05 Jun 2008) | 5 lines
It turns out that searching on the forwarding station isn't very useful for
most people, so pull in the changes that allow searching for SMDI messages
based on other components of the SMDI message. Also, update the SMDI
documentation.
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and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
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r120371 | russell | 2008-06-04 11:26:43 -0500 (Wed, 04 Jun 2008) | 4 lines
Make the "dialplan remove include" CLI command actually work. Also, tweak
some formatting, and make the success message a little bit more clear.
(closes AST-52)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines
Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch
cannot perform any translation on the extension number before searching for it
in the target context.
(closes issue #12473)
Reported by: chappell
Patches:
pbx_loopback.c.diff uploaded by chappell (license 8)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines
(closes issue #11594)
Reported by: yem
Tested by: yem
This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash.
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and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines
Fix another place where peer->callno could change at a very bad time, and also
fix a place where a peer was used after the reference was released.
(inspired by rev 120001)
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a dynamic realtime queue member is added to the queue, and the
REMOVEMEMBER event is logged when a dynamic realtime member is
removed. Since no calling channel is associated with these events
the string "REALTIME" is placed where the channel's unique id is
normally placed.
(closes issue #12774)
Reported by: atis
Patches:
queue_log_rt_members.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
hold tracking information for mutexes. Now, the "core show locks" output
will output information about who is holding a rwlock when a thread is
waiting on it.
(closes issue #11279)
Reported by: ys
Patches:
trunk_lock_utils.v8.diff uploaded by ys (license 281)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r120061 | tilghman | 2008-06-03 13:23:32 -0500 (Tue, 03 Jun 2008) | 8 lines
When listing the manager users, managers in users.conf are not shown, even
though they are allowed to connect.
(closes issue #12594)
Reported by: bkruse
Patches:
12594-managerusers-2.diff uploaded by qwell (license 4)
Tested by: bkruse
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 lines
Updated the regressions on AEL. Hadn't updated
this for the changes I made to preserve ${EXTEN}
in switches, which affected several tests because
it adds extra priorities, and at least one needed to be updated
because of the removal of the empty extension warning
message.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines
as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
which is a message from Philipp Kempgen, requesting that the WARNING
that an extension is empty be reduced to a NOTICE or less, as empty
extensions are syntactically possible, and no big deal.
With which I agree, and have removed that WARNING message entirely.
I think it is not necessary to see this message. It didn't
state that a NoOp() was inserted automatically on your behalf,
and really, as users, who cares? Why freak out dialplan writers
with unnecessary warnings? The details of the machinations a compiler goes
thru to produce working assembly code is of little interest
to most programmers-- we will follow the unix principal of
doing our work silently.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines
Revert a change made for issue #12479. This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.
(closes issue #12770)
Reported by: dagmoller
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warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being
distributed under the LGPL, so we can move this module into the main tree.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines
Improve CLI command blacklist checking for the command manager action. Previously,
it did not handle case or whitespace properly. This made it possible for blacklisted
commands to get executed anyway.
(closes issue #12765)
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jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line
Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
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