Jeff Peeler [Thu, 24 Sep 2009 20:29:51 +0000 (20:29 +0000)]
Add bridge related dial flags to the bridge app
Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.
(closes issue #13165)
Reported by: tim_ringenbach
Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
modified by me
Implicitly sending a progress signal breaks some applications.
Call Progress() in your dialplan if you explicitly want progress to be sent.
(Reverts change 216430, closes issue #15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
........
Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward.
(closes issue #15739)
Reported by: DLNoah, jeffg
Patches:
20090914__issue15739.diff.txt uploaded by tilghman (license 14)
20090922__issue15739.diff.txt uploaded by tilghman (license 14)
Tested by: DLNoah, jeffg
Add support for 'setvar=' for MGCP device lines, like other channel drivers provide.
(closes issue #14818)
Reported by: alea-soluciones
Patches:
chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514)
Update fax number to the legal fax, not the generic fax.
(closes issue #15946)
Reported by: jtodd
Patches:
leif-is-a-wuss.txt uploaded by jtodd (license 870)
Tested by: jparker, tilghman, jtodd, russellb, mmichelson, seanbright, kpfleming, and the rest of the usual suspects
This commit adds the doxygen changes that I've made to describe the Mantis
work flow documentation for the open source issue tracker. This should make
it easier to determine the flow of issues through the issue tracker, and what
those statuses mean.
When IMAP variables were changed during a reload, Voicemail did not use the new values.
This change introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to expire
normally.
(closes issue #15934)
Reported by: viniciusfontes
Patches:
20090922__issue15934.diff.txt uploaded by tilghman (license 14)
Tested by: viniciusfontes
........
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
........
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
Missing value setting line for maxsecs/maxmessage
(closes issue #15696)
Reported by: fhackenberger
Patches:
maxsecs.patch uploaded by fhackenberger (license 592)
This patch cleans up the locking logic in chan_sip.c's
handle_invite_replaces() function as well as making use
of ast_do_masquerade() rather than forcing the masquerade
on an ast_read(). The code had several redundant unlocks
that would result in 'freed more times than we've locked!'
errors. I cleaned these up as well as moving all the unlock
logic to the end of the function. This patch should also
resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls.
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up.
In addition, there's a bit of cleanup to the arguments and documentation, in which
I discovered that the last feature added to this application duplicated an option
(oops!) and changed that option so that it now works.
(closes issue #14909)
Reported by: junky
Patches:
__20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10)
Tested by: amilcar, junky, flujan, lmadsen
Detect whether we actually have the long double type, before looking for those functions.
(closes issue #15017)
Reported by: tzafrir
Patches:
20090916__issue15017.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
Fixes CID pattern matching behavior to mirror that of extension pattern matching.
Pattern matching for extensions uses a type of scoring system, giving values for
specificity to each character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific patterns being first
in line for matching, but it will usually get the job done.
This patch merely brings CID matching to the same level as extension matching.
This patch does not attempt to tackle the problem shared by extension matching.
The firmware can now be found on downloads.digium.com, where the rest of our
binary downloads live. This was the last part of our Asterisk tarballs that
was considered non-free by Debian. :-)
If the user enters the same password as before, don't signal an error when the change does nothing.
(closes issue #15492)
Reported by: cbbs70a
Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
........
Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are accountcode and
musicclass, and the inheritance of datastores.
(closes issue #15372)
Reported by: Romik
Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs
........
Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor.
After talking to rmudgett about some of his recent iflist locking changes, it
was determined that the only place that would destroy a channel without being
explicitly to do so was in handle_init_event. The loop to walk the interface
list has been modified to wait to destroy the channel until the dahdi_pvt of
the channel to be destroyed is no longer needed.
Make calltoken support work with realtime users and peers.
In the course of this, I also found that the results of ast_gethostbyname
were being used incorrectly in both chan_iax2 and chan_sip, so both have
been fixed.
Richard Mudgett [Thu, 10 Sep 2009 22:31:12 +0000 (22:31 +0000)]
Cleaned up chan_dahdi iflist handling and locking.
* Fixed walking the iflist so it is always done with the iflock locked.
* Simplified iflist walking routines.
* Created chan_dahdi iflist insertion and extraction routines.
* Fixed duplicate_pseudo() malloc fail handling.
* Fixed infinite loop in action_dahdishowchannels() when showing a single channel.
The IAX2 Call Token security patch inadvertently broke the use of
encryption due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must first
be decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this, decryption
of full frames is once again done before looking into the frame. This
involves searching for an existing callno, checking the pvt to see if
encryption is turned on, and decrypting the packet before the internal
fields of the full frame are accessed.
Jeff Peeler [Thu, 10 Sep 2009 20:18:30 +0000 (20:18 +0000)]
Stop caller id transmission when offhook event detected.
This fixes the problem that would occur if an analog phone was picked up while
the caller id was being sent. The caller id before sent the whole spill even
after pickup and is now corrected.
Remove harmful code that causes endless loops.
Remove code that causes loops in registrations.
We have agreed that the patch that this code was part of was bad. I am ripping out the code that causes
the issue. putnopvut needs to check the rest of the patch, if it needs to be changed as well.
This solves the issue reported in #15540, but needs more work before we close it (as described above).
When MOH is playing on the channel, announcements sent through the conference are not heard.
(closes issue #14588)
Reported by: voipas
Patches:
20090716__issue14588__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, twisted, tilghman
........
Russell Bryant [Tue, 8 Sep 2009 18:06:57 +0000 (18:06 +0000)]
Fix audio problems with format_mp3.
This problem was introduced when the AST_FRIENDLY_OFFSET patch was merged.
I'm surprised that nobody noticed any trouble when testing that patch, but this
fixes the code that fills in the buffer to start filling in after the offset
portion of the buffer.
(closes issue #15850)
Reported by: 99gixxer
Patches:
issue15850.diff1.txt uploaded by russell (license 2)
Tested by: 99gixxer
Ensure that the default autoconf CFLAGS are not used.
A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.
live_ast: Fix asterisk.conf instead of regenerating it
* Don't write asterisk.conf from scratch. Fix the existing one.
* Pass extra 'make' command-line arguments to 'install' and 'samples'.
* Fix some extra typos.
David Vossel [Tue, 8 Sep 2009 14:26:30 +0000 (14:26 +0000)]
caller id number empty
parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.
Olle Johansson [Mon, 7 Sep 2009 18:24:04 +0000 (18:24 +0000)]
- Doxygen additions
- Remove unused string in sip_registry -- "random"
- Someone added a function in the middle of all forward declarations... Weird. Moved it out of that
section.
Olle Johansson [Mon, 7 Sep 2009 18:00:48 +0000 (18:00 +0000)]
Clean up the "offered_media" code
- Add variable for number of known media streams instead of hardcoding in definition of sip_pvt
- Rename "text" to "codecs" - beacuse it's what it is
- Add documentation for future developers so that we make sure that we define new sdp media types
for SRTP-variants
projects / thirdparty / asterisk.git / log
