Russell Bryant [Mon, 26 Feb 2007 20:42:21 +0000 (20:42 +0000)]
Restore the behavior of Asterisk 1.2 where if a device was not specified in
alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139)
Joshua Colp [Mon, 26 Feb 2007 17:09:53 +0000 (17:09 +0000)]
Use ast_strlen_zero to see if the language and/or context argument is not present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis)
Russell Bryant [Mon, 26 Feb 2007 00:35:04 +0000 (00:35 +0000)]
Blocked revisions 56729 via svnmerge
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r56729 | russell | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines
Ensure that lock.h is included in utils.c with AST_API_MODULE defined so that
the implementations will be properly included when the AST_INLINE_API functions
are not going to be inlined. (issue #9124, festr)
Russell Bryant [Fri, 23 Feb 2007 23:24:18 +0000 (23:24 +0000)]
Merged revisions 56504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
Joshua Colp [Fri, 23 Feb 2007 21:53:41 +0000 (21:53 +0000)]
Change log notice to debug. It is possible for a scheduled item to execute and be deleted at close to the same time and unavoidable. If this happens this message creeps up.
Russell Bryant [Fri, 23 Feb 2007 20:20:00 +0000 (20:20 +0000)]
Merged revisions 56406 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
Russell Bryant [Fri, 23 Feb 2007 17:58:57 +0000 (17:58 +0000)]
The IMAP storage code uses the same code to build the email that is used when
voicemail is sent via email using something like sendmail. In the patch from
bug 8033 to fix various IMAP storage problems, the line endings in the email
file were changed in the code from "\n" to "\r\n". However, this breaks
sending regular voicemail to email. So, this change conditionally sets line
endings to "\r\n" only if IMAP_STORAGE is enabled.
(issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage)
Russell Bryant [Thu, 22 Feb 2007 23:08:36 +0000 (23:08 +0000)]
Merge changes from team/russell/sla_updates.
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.
Kevin P. Fleming [Thu, 22 Feb 2007 00:08:54 +0000 (00:08 +0000)]
disable unloading of embedded modules... there is a fundamental problem with doing so that will not be fixed in this version of Asterisk due to its invasiveness
Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message.
Joshua Colp [Wed, 21 Feb 2007 17:18:19 +0000 (17:18 +0000)]
Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113)
Russell Bryant [Wed, 21 Feb 2007 00:22:29 +0000 (00:22 +0000)]
Blocked revisions 55750 via svnmerge
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r55750 | russell | 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines
Fix random crashes when using the MeetMe application. This patch converts list
handling to use the linked list macros and most importantly, implements
reference counting on the ast_conference objects. The reference counting was
first backported from 1.4. However, that code has some problems that caused
the reference count to never hit zero. Those problems are fixed in this patch
and will be resolved in 1.4 and trunk next, with a different patch.
Joshua Colp [Tue, 20 Feb 2007 23:08:45 +0000 (23:08 +0000)]
Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska)
Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155)
Doug Bailey [Mon, 19 Feb 2007 14:52:59 +0000 (14:52 +0000)]
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
Modified module unload process to avoid deadlocks on pthread cancels
Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)
Russell Bryant [Fri, 16 Feb 2007 22:49:42 +0000 (22:49 +0000)]
Merged revisions 55005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines
Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4,
and trunk. I decided that once a conference is created from meetme.conf,
it is acceptable behavior that the pin can not be changed until the
conference goes away. I also added a note in meetme.conf to describe this
behavior.
We still have another issue in 1.4 and trunk where some conferences with no
users don't go away. That is the real bug that needs to be addressed here.
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
Russell Bryant [Fri, 16 Feb 2007 21:12:18 +0000 (21:12 +0000)]
Merged revisions 54955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines
For conferences that are configured in meetme.conf, check the configuration
file every time someone joins the conference instead of only when the
conference is first created. This is to ensure that changes to the pin
numbers in the config file are always honored. (issue #9073)
Russell Bryant [Fri, 16 Feb 2007 18:03:41 +0000 (18:03 +0000)]
Fix setting "autofallthrough" to yes by default. It was set to enabled in
pbx.c. However, if the option was not present in extensions.conf, then
pbx_config.c would set it back to disabled.
Russell Bryant [Tue, 13 Feb 2007 19:42:00 +0000 (19:42 +0000)]
If we fail to create the SIP socket, then return -1 from reload_config() so
that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
Russell Bryant [Mon, 12 Feb 2007 17:58:43 +0000 (17:58 +0000)]
- Add the ability to register a callback to monitor state changes in an
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
Russell Bryant [Sat, 10 Feb 2007 00:35:09 +0000 (00:35 +0000)]
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
Russell Bryant [Sat, 10 Feb 2007 00:15:50 +0000 (00:15 +0000)]
When the Echo() application receives the digit '#', echo that back as well.
Since we already sent the BEGIN frame for that digit, it makes sense to send
the END as well.
projects / thirdparty / asterisk.git / log
