Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r149279 | mmichelson | 2008-10-14 18:57:46 -0500 (Tue, 14 Oct 2008) | 7 lines
When specifying an invalid timeout to Dial, take it
to mean that no timeout is desired.
(closes issue #13625)
Reported by: atis
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@149280 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r145962 | russell | 2008-10-02 14:30:45 -0500 (Thu, 02 Oct 2008) | 2 lines
The 'P' command for ExternalIVR was also added in 1.6.0
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@145964 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r145959 | russell | 2008-10-02 14:27:37 -0500 (Thu, 02 Oct 2008) | 2 lines
TCP support for ExternalIVR went in to 1.6.1, not 1.6.0
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@145960 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r140491 | jpeeler | 2008-08-29 12:53:32 -0500 (Fri, 29 Aug 2008) | 2 lines
Added the option s to the Park application which will silence the announcement of the parking space number. Also, fixes the bug of just clearing the flags instead of actually parsing the arguments to Park.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@140492 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135717 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
memory allocator. See doc/hoard.txt for more details.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135405 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This set of changes enhances IAX2 encryption support by adding key rotation
to provide enhanced security. The key used for encryption is rotated right
after the call gets set up, and then again every few minutes. This was
discussed at the last AstriDevCon. For interoperability with older versions
of Asterisk, there is an option that disables key rotation.
(closes issue #13018)
Reported by: bbryant
Patches:
07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
Tested by: russell, bbryant
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135158 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134443 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
implementations. Asterisk has, for a long time,
had its own implementation of poll(2) which
just used the input arguments to call select(2).
In 1.4, this internal implementation was used
for Darwin systems. This was removed in Asterisk
trunk at some point, but it seems as though this
was not the right move to make.
On Mac OS X, it appears as though the poll used
to gather CLI input does not respond properly
when connecting via a remote Asterisk console.
Reverting to the use of Asterisk's poll fixed
the issue.
Also, there is now an option for the configure
script, --enable-internal-poll, which will allow
for anyone to use Asterisk's internal poll
implementation in case they suspect that their
system's poll implementation is buggy.
closes issue #11928)
Reported by: adriavidal
Patches:
1.6.0-configurev2.patch uploaded by putnopvut (license 60)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134125 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #13134)
Reported by: eliel
Patches:
app_image.c.patch uploaded by eliel (license 64)
UPGRADE.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134088 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
deprecation, of course)
(closes issue #13016)
Reported by: caio1982
Patches:
dialplan_globals6.diff uploaded by caio1982 (license 22)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131606 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130044 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
explanation of the change may be found in configs/queues.conf.sample
(closes issue #12690)
Reported by: atis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127720 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127558 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
to how language is handled between two channels whose native
language is different. Prior to this patch, app_dial would have
the callee inherit the caller's language, and app_queue would not.
After this patch, app_dial no longer has the language inheritance
capability. This seems to make the most sense since it seems more
natural for a person to hear files played back in his/her native
language instead of the language of the person on the far end of
the call. See the CHANGES file for hints on how to keep the
previous behavior of app_dial if desired.
(closes issue #12489)
Reported by: bcnit
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125647 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124835 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124125 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12830)
Reported by: jcollie
Patches:
astertisk-trunk-121496-alternate-extensions.patch uploaded by jcollie (license 412)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124049 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
in 12494. Due to changes in trunk to use the astobj2 i/f in the sip channel driver, the order of the entries in the config file was lost, thus the output was in a random order, but no longer.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123448 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line
Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122128 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines
(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia
Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.
The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.
The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.
The T option was added to forkCDR to force
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.
The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via
email, irc, etc, over the past months/year)
The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.
Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122091 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121559 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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: http://svn.digium.com/svn/asterisk/trunk@121197 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12562)
Reported by: michael-fig
Patches:
20080515__bug12562.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121042 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120673 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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: http://svn.digium.com/svn/asterisk/trunk@120166 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
properly.
(closes issue #12760)
Reported by: Corydon76
Patches:
20080530__bug12760.diff.txt uploaded by Corydon76 (license 14)
Tested by: tim_ringenbach, Corydon76
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119296 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118647 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
If a deadlock is detected, then the typical lock information will be
printed along with a backtrace of the stack for the offending threads.
Use of this requires compiling with DETECT_DEADLOCKS and having glibc
installed.
Furthermore, issuing the "core show locks" CLI command will print the
normal lock information as well as a backtraces for each lock. This
requires that DEBUG_THREADS is enabled and that glibc is installed.
All the backtrace features may be disabled by running the configure
script with --without-execinfo as an argument
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118173 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
If you give chanisavail a list of channels, it will only
return the first available channel.
When this option is set, it will return all the available
channels from the given list.
(closes issue #12248)
Reported by: dagmoller
Patches:
app_chanisavail-snv.patch-v2.txt uploaded by dagmoller (license 436)
- major changes by me because russellb pointed out some buffer overflows
and codeguideline issues.
Converted it all to the ast_str_* api
Tested by: dagmoller, mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118101 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12705)
Reported by: ctooley
Patches:
new_externalivr_argument_format-v2.diff uploaded by ctooley (license 136)
new_externalivr_documentation.diff uploaded by ctooley (license 136)
and a few additional fixes by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117725 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(Related to issue #12677)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117264 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
'unknown', and better document the use of each parameter.
(closes issue #12633)
Reported by: tzafrir
Patches:
pridialplan_unknown_2.diff uploaded by tzafrir (license 46)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117182 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
press DTMF digits to switch between spying modes. Pressing 4 activates spy mode,
pressing 5 activates whisper mode, and pressing 6 activates barge mode. Use of
this feature overrides the normal operation of DTMF numbers.
This feature is courtesy of Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116522 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116237 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This is to enable video and text in originated calls. Development sponsored
by Omnitor AB, Sweden. (http://www.omnitor.se)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116229 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115588 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115586 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #12090)
Reported by: jaroth
Patches:
vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7)
20080509__bug12090.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115582 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(Closes issue AST-34)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115301 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
will show the version of each library respectively.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115078 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.
This originally came up as a suggestion on the asterisk-dev mailing list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115021 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
basis in the register line. This comes from a Switchvox patch. (issue AST-24)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114912 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
to announce-position, "limit" and "more," as well as a new option,
announce-position-limit. For more information on the use of these options,
see CHANGES or configs/queues.conf.sample.
(closes issue #10991)
Reported by: slavon
Patches:
app_q.diff uploaded by slavon (license 288)
Tested by: slavon, putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114906 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114874 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
the spied-on
party to be spoken instead of the channel name or number.
This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.
This change comes as a suggestion from Switchvox, which already has this feature. AST-23
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114813 f38db490-d61c-443f-a65b-d21fe96a405b
|