Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r294639 | jpeeler | 2010-11-11 13:31:00 -0600 (Thu, 11 Nov 2010) | 53 lines
Merged revisions 294384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) | 47 lines
Fix a deadlock in device state change processing.
Copied from some notes from the original author (Russell):
Deadlock scenario:
Thread 1: device state change thread
Holds - rdlock on contexts
Holds - hints lock
Waiting on channels container lock
Thread 2: SIP monitor thread
Holds the "iflock"
Holds a sip_pvt lock
Holds channel container lock
Waiting for a channel lock
Thread 3: A channel thread (chan_local in this case)
Holds 2 channel locks acquired within app_dial
Holds a 3rd channel lock it got inside of chan_local
Holds a local_pvt lock
Waiting on a rdlock of the contexts lock
A bunch of other threads waiting on a wrlock of the contexts lock
To address this deadlock, some locking order rules must be put in place and
enforced. Existing relevant rules:
1) channel lock before a pvt lock
2) contexts lock before hints lock
3) channels container before a channel
What's missing is some enforcement of the order when you involve more than any
two. To fix this problem, I put in some code that ensures that (at least in the
code paths involved in this bug) the locks in (3) come before the locks in (2).
To change the operation of thread 1 to comply, I converted the storage of hints
to an astobj2 container. This allows processing of hints without holding the
hints container lock. So, in the code path that led to thread 1's state, it no
longer holds either the contexts or hints lock while it attempts to lock the
channels container.
(closes issue #18165)
Reported by: antonio
ABE-2583
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294640 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) | 8 lines
Detect GMime properly on systems where gmime flags and libs are configured with pkg-config.
(closes issue #16155)
Reported by: jcollie
Patches:
20100917__issue16155.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294430 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Add connected line update for sig_analog transfers and simplify the
corresponding sig_pri and chan_misdn transfer code.
Note that if you create a three-way call in sig_analog before transferring
the call, the distinction of the caller/callee interception macros make
little sense. The interception macro writer needs to be prepared for
either caller/callee macro to be executed. The current implementation
swaps which caller/callee interception macro is executed after a three-way
call is created.
Review: https://reviewboard.asterisk.org/r/996/
JIRA ABE-2589
JIRA SWP-2372
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294349 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
Fix playback failure when using IAX with the timerfd module.
To fix this issue the alert pipe will now be used when the timerfd module is
in use. There appeared to be a race that was not solved by adding locking in the
timerfd module, but needed to be there anyway. The race was between the timer
being put in non-continuous mode in ast_read on the channel thread and the IAX
frame scheduler queuing a frame which would enable continuous mode before the
non-continuous mode event was read. This race for now is simply avoided.
(closes issue #18110)
Reported by: tpanton
Tested by: tpanton
I put tested by tpanton because it was tested on his hardware. Thanks for the
remote access to debug this issue!
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294278 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
The documentation for ast_rtp_instance_get_(local/remote)_address stated that
they returned 0 for success and -1 on failure. Instead, they returned 0 if the
address structure passed in was already equivalent to the address instance
local/remote address or 1 otherwise. 90% of the calls to these functions
completely ignored the return address and passed in an uninitialized struct,
which would make valgrind complain even though the operation was technically
safe.
This patch fixes the documentation and converts the get_xxx_address functions
to void since all they really do is copy the address and cannot fail.
Additionally two new functions
(ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3
times where the return value was actually checked. The
get_and_cmp_local_address function is currently unused, but exists for the sake
of symmetry.
The only functional change as a result of this change is that we will not do an
ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the
ast_sockaddr_copy() in the get_*_address functions. So, even though it is an
API change, it shouldn't have a noticeable change in behavior.
Review: https://reviewboard.asterisk.org/r/995/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293803 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
libpri.
Fixes our Bamboo builds.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293046 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
libpri.
Fixes our Bamboo builds.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292906 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
While testing chan_gtalk I noticed jabber was using my IPv6 address
and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip()
to return both IPv6 and IPv4 results. Adding a family parameter gives you
the ablility to choose.
Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results.
Review: https://reviewboard.asterisk.org/r/973/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291758 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This patch includes several chan_gtalk enhancements.
Two new gtalk.conf options have been added, externip
and stunadd. Setting externip allows us to
manually specify what the external IP address is
outside of a NAT environment. Setting the stunaddr
option to a valid stun server allows for that external
ip to be retrieved via a STUN server automatically. This
external IP is then advertised during call setup as
a possible candidate.
I have also attempted to clean up chan_gtalk's code
so it meets our coding guidelines. During this cleanup
I noticed several things that need to be done in the
code and made a TODO section at the top of the file.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291192 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r290751 | qwell | 2010-10-07 15:57:14 -0500 (Thu, 07 Oct 2010) | 16 lines
Merged revisions 290750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | 9 lines
Allow PRI to build properly when using --with-pri.
Use the directories found for the parent when using lib dependencies.
(closes issue #17314)
Reported by: tzafrir
Patches:
17314-withdeps.diff uploaded by qwell (license 4)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290752 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This patch was written by Philippe Sultan (phsultan). Thanks
for keeping this up to date!
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290479 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289840 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289543 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Review: https://reviewboard.asterisk.org/r/942/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289104 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r288637 | tilghman | 2010-09-23 22:36:01 -0500 (Thu, 23 Sep 2010) | 9 lines
Merged revisions 288636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines
Solaris compatibility fixes
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288638 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
So far all our tools for viewing and manipulating media streams
within Asterisk have been entirely focused on audio. That made
sense then, but is not scalable now. The FrameHook API lets us
tap into and manipulate _ANY_ type of media or signaling passed
on a channel present today or in the future. This tool is a step
in the direction of expanding Asterisk's boundaries and will help
generate some rather interesting applications in the future.
In addition to the FrameHook API, a simple dialplan function
exercising the api has been included as well. This function
is called FRAME_TRACE(). FRAME_TRACE() allows for the internal
ast_frames read and written to a channel to be output. Filters
can be placed on this function to debug only certain types of frames.
This function could be thought of as an internal way of doing
ast_frame packet captures.
Review: https://reviewboard.asterisk.org/r/925/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@287647 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286931 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
Merged revisions 286059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286189 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines
Another fix for Mac OS X.
While trying to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285962 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285931 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280020 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
The version of libedit that is bundled with asterisk is old and has some bugs.
This patch updates the bundled version of libedit within asterisk, and also
updates asterisk to use the system libedit instead if one is available (and
pkg-config is available). This review integrates several patches from other
users specifically kkm and tzafrir.
(closes issue #15929)
Reported by: kkm
Patches:
015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
(issue #16858)
Reported by: jw-asterisk
(closes issue #17039)
Reported by: tzafrir
Patches:
0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/807/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280019 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r279946 | dvossel | 2010-07-27 15:54:32 -0500 (Tue, 27 Jul 2010) | 24 lines
Merged revisions 279945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that
list stays on the channel until the channel is destroyed. There
is no reason to keep that list on the channel if it becomes empty.
If it is empty that just means we are doing needless translating
for every ast_read and ast_write. This patch removes the audiohook
list from the channel once it is detected to be empty on either a
read or write. If a audiohook is added back to the channel after
this list is destroyed, the list just gets recreated as if it never
existed to begin with.
(closes issue #17630)
Reported by: manvirr
Review: https://reviewboard.asterisk.org/r/799/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279949 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | 5 lines
Really fix sounds Makefile (and make it readableish).
There was a rather large syntax error that should have caused ALL versions of GNU make to fail.
I don't know how it worked.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279658 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 Jul 2010) | 2 lines
Use a special Makefile for noobs who still have GNU Make 3.80.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279562 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279504 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #17677)
Reported by: outcast
Patches:
issue0017677.patch uploaded by pabelanger (license 224)
Tested by: elguero
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279280 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278957 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Review: https://reviewboard.asterisk.org/r/795
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278908 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278579 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Review: https://reviewboard.asterisk.org/r/793/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278538 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r278167 | tilghman | 2010-07-20 15:59:06 -0500 (Tue, 20 Jul 2010) | 4 lines
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278272 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277814 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r277738 | tilghman | 2010-07-17 11:59:11 -0500 (Sat, 17 Jul 2010) | 5 lines
Remove uclibc cross-compile triplet, as uclibc has a working fork()... it's only uclinux that does not.
(closes issue #17616)
Reported by: pprindeville
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277775 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines
Since we split values at the semicolon, we should store values with a semicolon as an encoded value.
(closes issue #17369)
Reported by: gkservice
Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277773 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
xmllint seems to be more commonly available since it comes with libxml2.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277703 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
cancelling lagid.
No, replacing usleep(1) with sched_yield() did not have an effect.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277484 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #17644)
Reported by: pprindeville
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276769 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #17475)
Reported by: tilghman
Review: https://reviewboard.asterisk.org/r/695/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276490 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
tracking down the source.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275105 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r275021 | russell | 2010-07-09 10:33:08 -0500 (Fri, 09 Jul 2010) | 4 lines
Document that a leading and trailing slash is expected for test categories.
Also, emit a warning if a test is registered without one of these.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275022 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274907 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274727 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273830 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273198 f38db490-d61c-443f-a65b-d21fe96a405b
|