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r152369 | tilghman | 2008-10-28 12:07:39 -0500 (Tue, 28 Oct 2008) | 15 lines
Merged revisions 152368 via svnmerge from
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r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) | 8 lines
Reset all DIAL variables back to blank, in case Dial is called multiple times
per call (which could otherwise lead to inconsistent status reports).
(closes issue #13216)
Reported by: ruddy
Patches:
20081014__bug13216.diff.txt uploaded by Corydon76 (license 14)
Tested by: ruddy
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r152174 | tilghman | 2008-10-27 11:44:55 -0500 (Mon, 27 Oct 2008) | 2 lines
Set ARGC in subroutines with the number of arguments passed.
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r152060 | seanbright | 2008-10-26 16:25:08 -0400 (Sun, 26 Oct 2008) | 15 lines
Merged revisions 152059 via svnmerge from
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r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct 2008) | 7 lines
Since passing \0 as the second argument to strchr is valid (and will
match the trailing \0 of a string) we need to check that first, otherwise
we end up with incorrect results. Fix suggested by reporter.
(closes issue #13787)
Reported by: meitinger
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r151906 | russell | 2008-10-25 06:02:11 -0500 (Sat, 25 Oct 2008) | 16 lines
Merged revisions 151905 via svnmerge from
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r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) | 8 lines
Move AMI initialization to occur after loading modules. This prevents a
deadlock when someone tries to initiate a module reload from the AMI just
as Asterisk is starting.
(closes issue #13778)
Reported by: hotsblanc
Fix suggested by hotsblanc
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r151600 | mmichelson | 2008-10-22 15:05:14 -0500 (Wed, 22 Oct 2008) | 10 lines
Change some logical ands to bitwise ands and add
messages alerting that a channel is being ignored
if the PROC_DAHDI_NOCHAN option is set in process_dahdi.
(closes issue #13759)
Reported by: smurfix
Patches:
dahdi.patch uploaded by smurfix (license 547)
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r151554 | russell | 2008-10-22 12:44:05 -0500 (Wed, 22 Oct 2008) | 2 lines
Fix this check to use the proper variable (the result from get_in_brackets)
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r151555 | russell | 2008-10-22 12:45:05 -0500 (Wed, 22 Oct 2008) | 2 lines
Print out the right var in the log message
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r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines
Make the sip_standard_port function more granular by allowing separate
type and port arguments. This is necessary because when building our From
and Contact headers, we need to be absolutely sure that we are placing our
source port there and not the peer's source port.
(closes issue #12761)
Reported by: asbestoshead
Patches:
patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455)
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r151428 | mmichelson | 2008-10-21 18:27:45 -0500 (Tue, 21 Oct 2008) | 14 lines
If a peer uses any transport other than UDP, then MWI will
fail for that peer since sip_alloc will allocate a sip_pvt with
a default transport of UDP. This change resets the socket type
immediately after allocating the sip_pvt in sip_send_mwi_from_peer,
so that the proceeding call to create_addr_from_peer does not fail
right away. The socket data from the peer is properly copied to
the sip_pvt in create_addr_from_peer.
(closes issue #13710)
Reported by: andrew53
Patches:
sip_notify_use_tcp.patch uploaded by andrew53 (license 519)
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r151420 | mmichelson | 2008-10-21 18:08:56 -0500 (Tue, 21 Oct 2008) | 10 lines
When attempting to resolve hostnames, we need to be sure
to remove any parameters from the string so that name
resolution succeeds.
(closes issue #13727)
Reported by: fnordian
Patches:
resolvewithouturiparameter.patch uploaded by fnordian (license 110)
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r151371 | tilghman | 2008-10-21 10:20:50 -0500 (Tue, 21 Oct 2008) | 5 lines
Default file modes should always be full read and write, to allow the system
administrator to make the decision of what permissions will actually be given,
through the use of the process umask.
(Closes issue# 13751)
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r151242 | kpfleming | 2008-10-20 07:59:04 +0300 (Mon, 20 Oct 2008) | 9 lines
Merged revisions 151240 via svnmerge from
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r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct 2008) | 3 lines
break up acinclude.m4 into individual files, which will make it easier to maintain, easier to add new macros (less patching) and will ease maintenance of these macros across Asterisk branches
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r151243 | kpfleming | 2008-10-20 08:00:56 +0300 (Mon, 20 Oct 2008) | 9 lines
Merged revisions 151241 via svnmerge from
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r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct 2008) | 2 lines
rename this macro to properly reflect what it does
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r151101 | kpfleming | 2008-10-19 22:11:28 +0300 (Sun, 19 Oct 2008) | 13 lines
cleaup of the TCP/TLS socket API:
1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines
2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines)
3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines)
4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied
5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address
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r150817 | bweschke | 2008-10-17 22:18:33 -0400 (Fri, 17 Oct 2008) | 8 lines
Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766.
We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing.
(closes issue #13715)
reported by: makoto
patch by: bweschke
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r150729 | qwell | 2008-10-17 16:35:23 -0500 (Fri, 17 Oct 2008) | 1 line
Merge codec_consistency branch. This should make sample usage much happier.
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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
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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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