The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.
(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This channel driver is a replacement for both chan_gtalk and chan_jingle but adds additional features not found in either.
These features include full configuration reload, video, full codec support, bidirectional cause code mapping, hold,
unhold, and ringing indication. It is also compliant with the current published Jingle and Google Jingle specifications.
The original Google Talk protocol is also supported for Google Voice interoperability.
You may ask yourself though where the name motif comes from... and I would say to you... music!
motif: a perceivable or salient recurring fragment or succession of notes
Sorta like a jingle!
Review: https://reviewboard.asterisk.org/r/1917/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ANI2 information is now parsed out of SIP From headers when present in
the oli, isup-oli, and ss7-oli parameters and is available via the
CALLERID(ani2) dialplan function.
(closes issue ASTERISK-19912)
Patch-by: Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1947/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.
This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.
Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.
* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.
Review: https://reviewboard.asterisk.org/r/1829/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in
favor of the Gosub method without a means of backwards compatibility.
(issue ASTERISK-18809)
(closes issue ASTERISK-19457)
Reported by: Matt Jordan
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1855/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated. This also adds
deprecation warnings for those features when used and in documentation.
Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change permits each verbose destination (consoles, logger) to have its
own concept of what the verbosity level is. The big feature here is that
the logger will now be able to capture a particular verbosity level without
condemning each console to need to suffer that level of verbosity.
Additionally, a stray 'core set verbose' will no longer change what will go
to the log.
Review: https://reviewboard.asterisk.org/r/1599/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes res_ais and introduces a new module, res_corosync.
The OpenAIS project is deprecated and is now just a wrapper around
Corosync. This module provides the same functionality using the same
core infrastructure, but without the use of the deprecated components.
Technically res_ais could have been used with an AIS implementation other
than OpenAIS, but that is the only one I know of that was ever used.
Review: https://reviewboard.asterisk.org/r/1700/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk now supports using ODBC with databases where a single schema must be
selected. Previously, INSERTs would fail because they did not take into
account extra fields cause by having multiple schemas. This also corrects
some SQL resource leaks.
(closes issue ASTERISK-17106)
Patch-by: Alexander Frolkin
Patch-by: Tilgnman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add an AMI event in the Call category that is issued when a call is terminated
due to either RTP stream inactivity or SIP session timer expiration.
Event description:
Event: SessionTimeout
Source: source
Channel: channel-name
Uniqueid: channel-unique-id
`source` can be either RTPTimeout or SIPSessionTimer
(closes issue ASTERISK-16467)
Patch-by: Kirill Katsnelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Adds per-parking lot options comebackcontext and comebackdialtime
* Makes comebacktoorigin settable per parking lot
* Sets a PARKER channel variable when comebacktoorigin is disabled
(closes issue ASTERISK-16643)
Reported by: Mitch Sharp (bluecrow76)
Patches:
asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff by Mitch Sharp (bluecrow76) license 5231
with updates by me.
Review: https://reviewboard.asterisk.org/r/1674
Review: https://reviewboard.asterisk.org/r/963
Reviewed by Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Users created by users.conf with hasvoicemail=yes have been documented as
using a Gosub to stdexten since v1.6.0. However, the code still generates
dialplan to access stdexten as a Macro as documented in v1.4; which does
not work with the newer extensions.conf.sample file.
* Make generated dialplan access the stdexten dialplan with the documented
Gosub instead of the older Macro style.
(closes issue ASTERISK-18809)
Reported by: Jay Allen
Patches:
gosub_patch-pbx_config.patch (license #6323) patch uploaded by Jay Allen (modified)
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655). This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.
I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.
(issue ASTERISK-18974)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
........
r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) | 8 lines
Add astdb conversion utility for Berkeley to SQLite 3
If someone wants to backtrack from Asterisk 1.8 to 10 they can use the
astdb2bdb utility to convert the database back to the Berkeley format
that Asterisk 1.8 uses.
Review: https://reviewboard.asterisk.org/r/1502/
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r340220 | twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
Add a missing file for the astdb2bdb conversion utility
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ENUM dialplan functions do not report DNS query errors properly. It is
useful to differentiate between failed query (e.g. non-existent domain) vs. no
data records of the appropriate type. This is required to make overlapped
dialing work.
(closes issue ASTERISK-13769)
Review: https://reviewboard.asterisk.org/r/1355/
Patch-by: Timo Teras
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
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r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines
Bump the AMI protocol version to 1.2
As a result of converting Unlink events that were missed in the AMI
1.1 update to Bridge events, the AMI protocol version is being incremented.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.
Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.
We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Make autoservice_start() and autoservice_stop() return nothing. Also check if
the autoservice flag is set before starting or stopping the autoservice and
stop and start the autoservice when returning control to and getting control
from the pbx engine.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the moh_signaling option to specify what to do when the channel's
bridged peer puts the ISDN channel on and off of hold.
Implemented as a FSM to control libpri ISDN signaling when the bridged
peer places the channel on and off of hold with the AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD control frames.
JIRA SWP-2687
JIRA ABE-2691
Review: https://reviewboard.asterisk.org/r/1063/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The documentation for this option did not match the code. Fix that along with
some minor cleanups to the code along the way. Document a slight change in
behavior (to something that was previously undocumented) in UPGRADE.txt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
This prevents handling the call failure by just continuing on in the dialplan.
Since this would be a change in behavior, the new option to disable this
behavior is forwardloopdetected which defaults to 'yes'.
Review: https://reviewboard.asterisk.org/r/764/
........
(no option for trunk, just changing the behavior)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity
written unless cdr.conf exists and is configured.
(closes issue #17373)
Reported by: wdoekes
Tested by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear
their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber.
This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape.
If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour.
Reported by: alecdavis
Tested by: alecdavis
Patch
vm_a_extension.diff2.txt uploaded by alecdavis (license 585)
Review: https://reviewboard.asterisk.org/r/489/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since ISDN works like SIP and not analog ports in regard to devices, the
device state based on the ISDN channel number could not work. This has
not been an issue until the advent of PTMP NT mode. Previously, ISDN
lines were used as trunks and did not have to keep track of specific
devices.
As an interim solution until device states are properly implemented, the
channel name is being changed to the following format to use the generic
device state support:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
Dialplan hints would thus be:
exten => xxx,hint,DAHDI/i2/5551212
This will work with the following restrictions:
* The number of devices/phones cannot exceed the number of B channels.
(i.e., BRI has 2)
* Each device/phone can only have one number. No shared MSN's.
* The phones/devices probably should not use subaddressing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines
Remove the IAXy firmware from Asterisk.
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. :-)
(closes issue #15838)
Reported by: paravoid
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
(closes issue #15022)
Reported by: greenfieldtech
Patches:
20090519__issue15022.diff.txt uploaded by tilghman (license 14)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
hashing used by app_queue.c to be case-insensitive.
This is accomplished by adding a new case-insensitive
hashing function.
This was necessary to prevent bad refcount errors
(and potential crashes) which would occur due to the
fact that queues were initially read from the config
file in a case-sensitive manner. Then, when a user
issued a CLI command or manager action, we allowed
for case-insensitive input and used that input to
directly try to find the queue in the hash table. The result
was either that we could not find a queue that was input or
worse, we would end up hashing to a completely bogus value
based on the input.
This commit resolves the problem presented in
issue #13703. However, that issue was reported against
1.6.0. Since this fix introduces a behavior change, I am
electing to not place this same fix in to the 1.6.0 or 1.6.1
branches, and instead will opt for a change which does not
change behavior.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
implementation from using the volatile libtds, to using the db-lib front end.
The unintended side effect of this is that we support (at least) versions 0.62
through 0.82 of the FreeTDS distribution without any #ifdef ugliness.
(closes issue #12844)
Reported by: jcollie
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function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
Reported by: mostyn
Patches:
iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
(with some additional cleanup by me)
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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
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for the sake of backwards compatibility, since it is a non-trivial task to convert
existing large dialplans that depend on Macro() to use GoSub(), instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 110869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines
due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves
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(closes issue #11553)
Reported by: johan
Patches:
UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
CHANGES.channelredirect.patch uploaded by johan (license 334)
app_channelredirect-20080219.patch uploaded by johan (license 334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using
'1' as a queue-round-seconds value is no longer valid.
(closes issue #9736)
Reported by: caio1982
Patches:
queue_announce5.diff uploaded by caio1982 (license 22)
Tested by: caio1982, putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This way, if people need to go back and review what was deprecated in previous
major releases, it is readily available to them. Thanks for the suggestion!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
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- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
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using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1 line
support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary.
........
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
SQLite3 database. (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
support for SQLite version 2. I decided that this was ok since we didn't have
any realtime support for version 3. If someone ports this to version 3, then
version 2 support can be removed or marked deprecated.
(issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.
Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality. Those are:
* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matthew Jordan
Mark Michelson
Joshua Colp
Kinsey Moore
Richard Mudgett
Sean Bright
Igor Goncharovskiy
Tilghman Lesher
Russell Bryant
Walter Doekes
Terry Wilson
Paul Belanger
Olle Johansson
Matthew Nicholson
Jason Parker
Leif Madsen
David Vossel
Gregory Nietsky
Bradley Latus
Alec L Davis
TransNexus OSP Development
David Ruggles
Kevin P. Fleming
Michiel van Baak
Steve Murphy
Dwayne M. Hubbard