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r150307 | mmichelson | 2008-10-16 19:13:35 -0500 (Thu, 16 Oct 2008) | 14 lines
After a long discussion on #asterisk-bugs, it seems kind of
odd that a channel would be named after the originating port.
For endpoints that always include ":5060" as part
of the From: header, it will mean that you have a ton of
channels with names like "SIP/5060-3ea38a8b."
I am boldly moving forward with this change in trunk, but I'm
not touching other branches with this one since this definitely
would qualify as a behavior change. If there is a problem with
this commit, and I haven't seen the obvious reason why you'd want
to name the channel after the port from which the call originated,
then please feel free to revert this
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r150309 | jpeeler | 2008-10-16 19:14:19 -0500 (Thu, 16 Oct 2008) | 3 lines
Initialize character arrays as they are not guaranteed to be set.
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r150305 | mmichelson | 2008-10-16 18:41:16 -0500 (Thu, 16 Oct 2008) | 14 lines
Merged revisions 150304 via svnmerge from
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r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct 2008) | 6 lines
Reverting changes from commits 150298 and 150301 since
I was mistakenly under the assumption that dialplan functions
*always* required that a channel be present. I need to go
home earlier, I think :)
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r150302 | mmichelson | 2008-10-16 18:36:49 -0500 (Thu, 16 Oct 2008) | 24 lines
Merged revisions 150298,150301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct 2008) | 10 lines
Don't try to call a dialplan function's read callback from
the manager's GetVar handler if an invalid channel has
been specified. Several dialplan functions, including
CHANNEL and SIP_HEADER, do not check for NULL-ness of
the channel being passed in.
(closes issue #13715)
Reported by: makoto
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r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct 2008) | 3 lines
And don't forget to return on the error condition
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r150210 | mmichelson | 2008-10-16 16:23:38 -0500 (Thu, 16 Oct 2008) | 12 lines
Change configure script to search for openais in
both /usr/lib and /usr/lib64 since some distros
place 64-bit libraries only in the /usr/lib64
directory.
(closes issue #13721)
Reported by: jcollie
Patches:
0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie (license 412)
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r150207 | mmichelson | 2008-10-16 15:57:18 -0500 (Thu, 16 Oct 2008) | 12 lines
INVITES with proxy auth were sent with a different branch
than what was in the invite_branch of a sip_pvt, meaning
that if a CANCEL were sent later, the branch in the CANCEL
would not match the branch in the latest INVITE sent out, leading
to some endpoints responding to the CANCEL with a 481.
(closes issue #13714)
Reported by: fnordian
Patches:
invite_branch.patch uploaded by fnordian (license 110)
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r149981 | kpfleming | 2008-10-16 15:28:56 +0200 (Thu, 16 Oct 2008) | 3 lines
return this logic to where it used to be, *after* the dialog->needdestroy flag has been determined to be set; otherwise, we generate these debug messages every time we inspect every active dialog
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r149802 | mmichelson | 2008-10-15 15:55:42 -0500 (Wed, 15 Oct 2008) | 12 lines
Make the sip_proxy struct reference counted. This is
necessary to allow for a sip_pvt to maintain a reference
to a sip_peer's outboundproxy even after the peer has
been freed.
(closes issue #13700)
Reported by: fnordian
Patches:
13700.patch uploaded by putnopvut (license 60)
Tested by: fnordian
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r149637 | tilghman | 2008-10-15 11:41:54 -0500 (Wed, 15 Oct 2008) | 8 lines
When using MALLOC_DEBUG, codec_lpc10 leaks memory, because it matches a library
malloc() with an ast_free (which, of course, doesn't match up with known
allocated memory, so the free fails).
(closes issue #13702)
Reported by: eliel
Patches:
codec_lpc10_lpcini.c uploaded by eliel (license 64)
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r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines
Merged revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines
Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
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r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines
Merged revisions 149200 via svnmerge from
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r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines
Update the queue with the correct number of calls and
whether the call was completed within the service level
when a transfer takes place. This way, we do not "break"
the leastrecent and fewestcalls strategies by not logging
a call until after the transferred call has ended.
(closes issue #13395)
Reported by: Marquis
Patches:
app_queue.c.transfer.patch uploaded by Marquis (license 32)
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r148738 | kpfleming | 2008-10-14 12:33:14 +0200 (Tue, 14 Oct 2008) | 9 lines
Merged revisions 148736 via svnmerge from
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r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct 2008) | 3 lines
on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly
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r148695 | kpfleming | 2008-10-14 11:31:50 +0200 (Tue, 14 Oct 2008) | 1 line
ensure that *all* fields in the req structure are cleared out before reusing it; has_to_tag was not cleared, which caused the second incoming call over a TCP socket to fail if pedantic checking was enabled
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r148519 | murf | 2008-10-13 11:14:38 -0600 (Mon, 13 Oct 2008) | 15 lines
Hmmm. Nobody (but me) is interested in seeing
the trie info when they do 'dialplan show ...'
(even with debug set to non-zero); so I set up a
'dialplan debug [context]' cli command instead,
to explicitly show just the trie info. I even
added an extension_exists() call to make sure the
trie info is built. I moved the explanatory header
to above the extension loop to ensure it only prints
once. And it will do this now, whether debug is set
or not.
I removed the trie printing from the 'dialplan show'
command entirely.
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r148376 | mmichelson | 2008-10-10 16:21:45 -0500 (Fri, 10 Oct 2008) | 13 lines
The logic used when checking a peer got changed subtly
in the "kill the user" commit and caused calls relying
on the insecure setting to not work properly. I changed
for finding a peer back to how it was prior to that
commit.
(closes issue #13644)
Reported by: pj
Patches:
13644_trunkv2.patch uploaded by putnopvut (license 60)
Tested by: pj
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r148373 | mmichelson | 2008-10-10 16:18:10 -0500 (Fri, 10 Oct 2008) | 8 lines
Make sure that the inUse and inRinging fields for
a sip peer cannot go below zero. This is a regression
from 1.4 and so it will be applied to 1.6.0 as well.
(closes issue #13668)
Reported by: mjc
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