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refinement, but this won't have as many folks bothered.
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r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines
Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list).
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r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) | 2 lines
Bug 9486 - memory leak when opening a filestream
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Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
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r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | 6 lines
It is valid to redirect channels via the manager interface that are not in the
UP state. Instead of checking for that to prevent to ensure a dead channel
doesn't get redirected, just use the ast_check_hangup() API call.
(issue #9457, reported by Callmewind, patch by me)
(related to issue #8977)
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r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | 5 lines
When doing a built-in blind or attended transfer, restore the ability to use '#'
to terminate the number and immediately do the transfer instead of having to
dial the number and just wait for the feature digit timeout.
(issue #8366, xueliangliang)
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r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | 6 lines
Add the SO_REUSEADDR flag to sockets handled by netsock. This is needed by
the patch that went in for issue 7874. chan_iax2 needs to be able to create
socket that is lisetning on INADDR_ANY, but also be able to bind sockets to
specific addresses. (Thanks to Stevenson on the asterisk-dev mailing list
for explaining why this flag was needed.)
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Mainly with CDRs generated from transfer situations.
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r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines
If an error occurs when reading from an RTP socket, and the error code does not
indicate that we should try again, then return NULL instead of a "null frame".
This will prevent Asterisk from trying over and over again, and eventually
causing the system to crash. (issue #8285, john)
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thread-safe, which is a minimum requirement.
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because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
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these chanes will only be done in the trunk.
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appdocs TeX file
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documentation.
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* Convert most of the doc directory into a single LaTeX formatted document
so that we can generate a PDF, HTML, or other formats from this
information.
* Add a CLI command to dump the application documentation into LaTeX format
which will only be include if the configure script is run with
--enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included. However, you
can simply run "make asterisk.pdf" to generate it yourself. We may include
it in release tarballs or have automatically generated ones on the web site,
but that has yet to be decided.
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by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ)
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make_compatible function too. (issue #9221 reported by marcelbarbulescu)
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r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines
Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez)
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attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines
If the channels acquired using the manager Redirect action are not up, then
don't attempt to do anything with them. It could lead to weird behavior,
including crashes. (issue #8977)
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r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line
Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong)
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Shaduri
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r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line
further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction
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Background application when using the m option.
(issue #9176, mjagdis)
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r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 line
a small documentation update, to reflect reality in the goto doc strings, as per 9156, Goto does not proceed to next prio if jump fails
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present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis)
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commands to work. (issue #9135 reported by mvanbaak)
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r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines
Issue 9130 - If prev is the last item on the channel list, then evaluating
additional conditions (e.g. name prefix) will cause a NULL dereference.
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r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines
Fix up a couple more signal handlers to not do bad things that could cause
various undesirable results. The other day, I made Asterisk deadlock by
hitting Control-C because of a bad signal handler. Now, signal handlers
just set a flag and write to an alert pipe for the flag to be handled. Then,
there is another thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the background,
a thread is created to do it.
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and be deleted at close to the same time and unavoidable. If this happens this message creeps up.
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This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines
Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.
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reported by blitzrage)
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doing so that will not be fixed in this version of Asterisk due to its invasiveness
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responses. (requested by Pari)
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Issue 9079
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free to change it. (issue #9045 reported by gork)
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asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
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