Like Dial and Queue, FollowMe needs to deal with
AST_CONTROL_CONNECTED_LINE information so when the parties are initially
bridged, the connected line information will be correct.
* Added the 'I' option just like the app_dial and app_queue 'I' option.
(closes issue ASTERISK-18969)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1656/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel. The channel driver
thread and the PBX thread running dialplan.
* Add lock protection around CDR API calls that access an ast_channel
pointer.
(closes issue ASTERISK-18836)
Reported by: gpluser
Review: https://reviewboard.asterisk.org/r/1628/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The addition of the Connected Line support changed how CallerID is passed
to outgoing calls. The FollowMe application was not updated to pass
CallerID to the outgoing calls.
* Fix FollowMe CallerID on outgoing calls.
* Restructured findmeexec() to fix several memory leaks and eliminate some
duplicated code.
* Made check the return value of create_followme_number(). Putting a NULL
into the numbers list is bad if create_followme_number() fails.
* Fixed a couple uses of ast_strdupa() inside loops.
* The changes to bridge_builtin_features.c fix a similar CallerID issue
with the bridging API attended and blind transfers. (Not used at this
time.)
(closes issue ASTERISK-17557)
Reported by: hamlet505a
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes an issue where the voicemail duration was being reported
with a duration significantly less than the actual sound file duration.
Voicemails that contained mostly silence were reporting the duration of
only the sound in the file, as opposed to the duration of the file with
the silence. This patch fixes this by having two durations reported in
the __ast_play_and_record family of functions - the sound_duration and the
actual duration of the file. The sound_duration, which is optional, now
reports the duration of the sound in the file, while the actual full duration
of the file is reported in the duration parameter. This allows the voicemail
applications to use the sound_duration for minimum duration checking, while
reporting the full duration to external parties if the voicemail is kept.
(issue ASTERISK-2234)
(closes issue ASTERISK-16981)
Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r297713 | tilghman | 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) | 15 lines
Merged revisions 297689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines
Don't create a Local channel if the target extension does not exist.
(closes issue #18126)
Reported by: junky
Patches:
followme.diff uploaded by junky (license 177)
(partially restructured by me to avoid a possible memory leak)
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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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r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines
Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are accountcode and
musicclass, and the inheritance of datastores.
(closes issue #15372)
Reported by: Romik
Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for the "pls_hold_prompt" option. This does not affect any released
version of Asterisk, so there is no need to update the CHANGES
file for this.
(closes issue #13893)
Reported by: eliel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines
Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines
Fix a couple uses of sprintf. The second one could actually cause an overflow
of a stack buffer. It's not a security issue though, it only depends on your
configuration.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines
Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution.
(closes issue #10327)
Reported by: kkiely
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
