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r75447 | russell | 2007-07-17 15:51:25 -0500 (Tue, 17 Jul 2007) | 1 line
cast arguments to ast_log so that it builds without warnings for me
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r75445 | russell | 2007-07-17 15:48:21 -0500 (Tue, 17 Jul 2007) | 13 lines
Merged revisions 75444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | 5 lines
Ensure that when encoding the contents of an ast_frame into an iax_frame, that
the size of the destination buffer is known in the iax_frame so that code
won't write past the end of the allocated buffer when sending outgoing frames.
(ASA-2007-014)
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r75439 | file | 2007-07-17 17:40:57 -0300 (Tue, 17 Jul 2007) | 2 lines
Ensure that the pointer to STUN data does not go to unaccessible memory. (ASA-2007-017)
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r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul 2007) | 6 lines
Fixing an error I made earlier. ast_fileexists can return -1 on failure, so I need to be sure that we only enter the if
statement if it is successful.
Related to my fix to issue #10186
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r75403 | russell | 2007-07-17 15:01:12 -0500 (Tue, 17 Jul 2007) | 12 lines
(closes issue #10209)
Reported by: juggie
Patches:
10209-trunk-2.patch uploaded by juggie
Tested by: juggie, blitzrage
In ast_pbx_run(), mark a channel as hung up after an application returned -1,
or when it runs out of extensions to execute. This is so that code can detect
that this channel has been hung up for things like making sure DeadAGI is used
on actual dead channels, and is beneficial for other things, like making sure
someone doesn't try to start spying on a channel that is about to go away.
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in a consistent way. This is meant to replace the custom code
which is repeated all over the place in the various files when
parsing config files, CLI entries and other string information.
Right now the code supports parsing int32, uint32 and sockaddr_in with
optional default values and bound checks. It contains minimal error
checking, but that can be easily extended as the need arises.
Being a new API i am introducing this only in trunk, though I believe
that once the interface has been ironed out it might become a
worthwhile addition to 1.4 as well - basically, the first time
we will need to fix a piece of argument parsing code, we might as
well bring in this change and use the new API instead.
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(and/or the optimization level) may think it is used uninitialized.
The code was indeed correct, but unfortunately the result of
some compiler checks such as -Wunused and -Wuninitialized depends
heavily on the optimization level.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75365 f38db490-d61c-443f-a65b-d21fe96a405b
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r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul 2007) | 8 lines
Restoring functionality from 1.2 wherein Retrydial will not exit if there is no announce file specified.
This change makes it so that if there is no announce file specified, the application will continue until finished (or caller hangs up).
If a bogus announce file is specified, then a warning message will be printed saying that the file could not be found, but execution will
still continue.
(closes issue #10186, reported by jon, patched by me)
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use the \retval tag for documenting return values, fixing various warnings
when generating the documentation, and various other things.
(closes issue #10203, snuffy)
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(const char *), while others define it as (char *). This is done to suppress
compiler warnings about it.
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r75078 | mmichelson | 2007-07-13 15:15:30 -0500 (Fri, 13 Jul 2007) | 13 lines
Merged revisions 75066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul 2007) | 5 lines
Fixed an issue where chanspy flags were uninitialized if no options were passed.
What triggered this investigation was an IRC chat where some people's quiet flags were
set while others' weren't even though none of them had specified the q option.
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r75067 | russell | 2007-07-13 15:10:40 -0500 (Fri, 13 Jul 2007) | 14 lines
Merged revisions 75059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 Jul 2007) | 6 lines
Ensure that adding a user to the list of users of a specific music on hold
class is not done at the same time as any of the other operations on this list
to prevent list corruption. Using the global moh_data lock for this is not
ideal, but it is what is used to protect these lists everywhere else in the
module, and I am only changing what is necessary to fix the bug.
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r75053 | russell | 2007-07-13 14:11:26 -0500 (Fri, 13 Jul 2007) | 20 lines
Merged revisions 75052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | 12 lines
(closes issue #9660)
Reported by: mmacvicar
Patches submitted by: bbryant, russell
Tested by: mmacvicar, marco, arcivanov, jmhunter, explidous
When using a TDM400P (and probably other analog cards) there was a chance that
you could hang up and pick the phone back up where it has been long enough to
be not considered a flash hook, but too soon such that the device reports that
it is busy and the person on the phone will only hear silence. This patch
makes chan_zap more tolerant of this and gives the device a couple of seconds
to succeed so the person on the phone happily gets their dialtone.
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sockets other than RTP ones.
The main change is a new API function in main/rtp.c (see there
for a description)
int ast_stun_request(int s, struct sockaddr_in *dst,
const char *username, struct sockaddr_in *answer)
which can be used to send an STUN request on a socket, and
optionally wait for a reply and store the STUN_MAPPED_ADDRESS
into the 'answer' argument (obviously, the version that
waits for a reply is blocking, but this is no different
from DNS resolutions).
Internally there are minor modifications to let stun_handle_packet()
be somewhat configurable on how to parse the body of responses.
At the moment i am not committing any change to the clients,
but adding STUN client support is extremely simple, e.g. chan_sip.c
could do something like this:
+ add a variable to store the stun server address;
static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */
+ add code to parse a config file of the form "stunaddr=my.stun.server.org:3478"
(not shown for brevity);
+ right after binding the main sip socket, talk to the stun server to
determine the externally visible address
if (stunaddr.sin_addr.s_addr != 0)
ast_stun_request(sipsock, &stunaddr, NULL, &externip);
so now 'externip' is set with the externally visible address.
so it is really trivial.
Similarly ast_stun_request could be called when creating the RTP
socket (possibly adding a struct sockaddr_in field in the struct
ast_rtp to store the externalip).
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+ mark a potentially dangerous write-past-end-of-buffer
+ localize some variables in the block generating stun replies.
As before, not ready yet for a merge to 1.4
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ast_rtp_new_with_bindaddr():
1. add comments to the logic of the main loop;
2. use a common exit point on failure so the cleanup is done only in one place;
3. handle failures in rtp_socket() in the main loop of the function;
No functional changes except for #3 above, so it is not yet
worthwhile merging this and other changes to 1.4
Once the cleanup work on this file will be complete (which among
other things should include some extensions to the stun support)
it might be a good thing to push all the changes to 1.4
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r74767 | russell | 2007-07-11 17:57:07 -0500 (Wed, 11 Jul 2007) | 13 lines
Merged revisions 74766 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | 5 lines
The function make_trunk() can fail and return -1 instead of a valid new call
number. Fix the uses of this function to handle this instead of treating it
as the new call number. This would cause a deadlock and memory corruption.
(possible cause of issue #9614 and others, patch by me)
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