........
r121403 | russell | 2008-06-09 19:43:06 -0500 (Mon, 09 Jun 2008) | 4 lines
Merge a couple of configure script checks in from team/russell/events. This adds
the checks for the CLM and EVT services from the SAForum AIS. I'm going to work
on merging in changes from this branch in pieces.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008) | 18 lines
Merged revisions 121280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines
Do not attempt to do emulation if an END digit is received and the length is
less than the defined minimum digit length, and the other end only wants END
digits (SIP INFO, for example).
(closes issue #12778)
Reported by: tsearle
Patches:
12778.rev1.txt uploaded by russell (license 2)
Tested by: tsearle
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r121230 | mmichelson | 2008-06-09 10:08:58 -0500 (Mon, 09 Jun 2008) | 27 lines
Merged revisions 121229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Note that this is being merged to trunk/1.6.0 because
it may affect non-callback agents with ackcall set)
........
r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines
A unique situation of timeouts brought forth a failure situation for
autologoff in chan_agent. If using AgentCallbackLogin-style agents,
then if the timeout specified by the Dial() to reach the agent's phone
was shorter than the timeout specified in queues.conf, then autologoff
would only work if the caller hung up while the agent's phone was ringing.
This patch allows autologoff to work in this situation when the call in
queue transfers to the next available agent (as it would have if the timeout
in queues.conf were less than the timeout in the Dial()).
(closes issue #12754)
Reported by: Rodrigo
Patches:
12754.patch uploaded by putnopvut (license 60)
Tested by: Rodrigo
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r121197 | mvanbaak | 2008-06-08 13:40:44 +0200 (Sun, 08 Jun 2008) | 12 lines
add a new argument to PrivacyManager to specify a context
where the entered phone number is checked.
You can now define a set of extensions/exten patterns that describe
valid phone numbers. PrivacyManager will check that context for a match
with the given phone number.
This way you get better control. For example people blindly hitting
10 digits just to get past privacymanager
Example line in extensions.conf:
exten => incoming,n,PrivacyManager(3,10,route-outgoing)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines
This was accidentally reverted.
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07 Jun 2008) | 2 lines
Fixes segfault when using ParkAndAnnounce. Also, loop made more efficient as announce template only needs to be checked until the number of colon separated arguments run out, not the entire pointer storage array. Was done in a similiar fashion in 1.4, but here we're using less variables.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@121138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines
Merged revisions 120863,120885 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
........
r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r120789 | tilghman | 2008-06-05 14:07:27 -0500 (Thu, 05 Jun 2008) | 3 lines
Merge the adaptive realtime branch, which will make adding new required fields
to realtime less painful in the future.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r120673 | bbryant | 2008-06-05 11:41:36 -0500 (Thu, 05 Jun 2008) | 1 line
Update CHANGES file for the things done in revision 120635.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r120635 | bbryant | 2008-06-05 11:24:19 -0500 (Thu, 05 Jun 2008) | 14 lines
This patch adds more detailed statistics for RTP channels, and provides an API call to access it, including maximums, minimums, standard deviatinos,
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r120602 | tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines
Conditionally load the AGI command gosub, depending on whether or not res_agi
has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r120230 | tilghman | 2008-06-03 18:17:33 -0500 (Tue, 03 Jun 2008) | 7 lines
Add a function, CHANNELS(), which retrieves a list of all active channels.
(closes issue #11330)
Reported by: rain
Patches:
func_channel-channel_list_function.diff uploaded by rain (license 327)
(with some additional changes by me, mostly to meet coding guidelines)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008) | 16 lines
Merged revisions 120226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines
Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch
cannot perform any translation on the extension number before searching for it
in the target context.
(closes issue #12473)
Reported by: chappell
Patches:
pbx_loopback.c.diff uploaded by chappell (license 8)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008) | 14 lines
Merged revisions 120173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines
(closes issue #11594)
Reported by: yem
Tested by: yem
This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines
Move compatibility options into asterisk.conf, default them to on for upgrades,
and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r120166 | mmichelson | 2008-06-03 16:22:52 -0500 (Tue, 03 Jun 2008) | 13 lines
Adding two new queue log events. The ADDMEMBER event is logged when
a dynamic realtime queue member is added to the queue, and the
REMOVEMEMBER event is logged when a dynamic realtime member is
removed. Since no calling channel is associated with these events
the string "REALTIME" is placed where the channel's unique id is
normally placed.
(closes issue #12774)
Reported by: atis
Patches:
queue_log_rt_members.patch uploaded by atis (license 242)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r120129 | russell | 2008-06-03 14:48:37 -0500 (Tue, 03 Jun 2008) | 2 lines
Use proper return values for a few application modules
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r120064 | russell | 2008-06-03 13:26:51 -0500 (Tue, 03 Jun 2008) | 10 lines
Add lock tracking for rwlocks. Previously, lock.h only had the ability to
hold tracking information for mutexes. Now, the "core show locks" output
will output information about who is holding a rwlock when a thread is
waiting on it.
(closes issue #11279)
Reported by: ys
Patches:
trunk_lock_utils.v8.diff uploaded by ys (license 281)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue, 03 Jun 2008) | 16 lines
Merged revisions 119966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 lines
Updated the regressions on AEL. Hadn't updated
this for the changes I made to preserve ${EXTEN}
in switches, which affected several tests because
it adds extra priorities, and at least one needed to be updated
because of the removal of the empty extension warning
message.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@120000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) | 24 lines
Merged revisions 119929 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines
as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
which is a message from Philipp Kempgen, requesting that the WARNING
that an extension is empty be reduced to a NOTICE or less, as empty
extensions are syntactically possible, and no big deal.
With which I agree, and have removed that WARNING message entirely.
I think it is not necessary to see this message. It didn't
state that a NoOp() was inserted automatically on your behalf,
and really, as users, who cares? Why freak out dialplan writers
with unnecessary warnings? The details of the machinations a compiler goes
thru to produce working assembly code is of little interest
to most programmers-- we will follow the unix principal of
doing our work silently.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r119892 | russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines
Do a deep copy of file and function strings to avoid a potential crash when
modules are unloaded.
(closes issue #12780)
Reported by: ys
Patches:
logger.diff uploaded by ys (license 281)
-- modified by me for coding guidelines
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r119839 | russell | 2008-06-02 15:08:24 -0500 (Mon, 02 Jun 2008) | 15 lines
Merged revisions 119838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines
Revert a change made for issue #12479. This change caused a regression such that
a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's'
extension anymore.
(closes issue #12770)
Reported by: dagmoller
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008) | 4 lines
Add app_fax from asterisk-addons, with some additional changes to resolve compiler
warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being
distributed under the LGPL, so we can move this module into the main tree.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r119799 | russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines
After determining that the version of spandsp installed is an acceptable version,
do a build and link test to ensure that the library is usable, and that libtiff
is also available
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008) | 13 lines
Merged revisions 119742 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines
Improve CLI command blacklist checking for the command manager action. Previously,
it did not handle case or whitespace properly. This made it possible for blacklisted
commands to get executed anyway.
(closes issue #12765)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines
Do not link the guest account with any configured XMPP client (in
jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008) | 9 lines
Merged revisions 119585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line
Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44).
........
................
r119637 | crichter | 2008-06-02 04:35:04 -0500 (Mon, 02 Jun 2008) | 9 lines
Merged revisions 119636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line
fixed compile issue when dev-mode is enabled
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119690 65c4cc65-6c06-0410-ace0-fbb531ad65f3