Followme answers an incoming call if it hasn't already been answered and starts
MOH. Some poorly designed autodialers see the answer and start playing their
message to the hold music. The 'N' option has been added to indicate ringing and
not answer until the call is accepted.
(closes issue #18479)
Reported by: ianc
Patches:
trunk_followme.diff uploaded by ianc (license 998)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sometimes during troubleshooting it can be useful to disable T.38 usage in order
to narrow down a problem. This patch adds an 'n' option to SendFAX and ReceiveFAX
so that can be done without having to disable T.38 usage entirely for the peer
that Asterisk is communicating with.
(inspired by trying to assist Bryant Zimmerman on asterisk-users)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r304097 | seanbright | 2011-01-25 20:26:26 -0500 (Tue, 25 Jan 2011) | 19 lines
Merged revisions 304096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines
Per the man page, setvbuf() must be called before any other operation on an open file.
We use setvbuf() to associate a buffer with a stream, but we have already written
to the open file. This works (by chance) on Linux, but fails on other platforms,
such as OpenSolaris.
(closes issue #16610)
Reported by: bklang
Patches:
setvbuf.patch uploaded by crjw (license 963)
Tested by: bklang, asgaroth, efutch
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r304007 | rmudgett | 2011-01-25 17:28:25 -0600 (Tue, 25 Jan 2011) | 22 lines
Merged revisions 304006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r304006 | rmudgett | 2011-01-25 17:25:32 -0600 (Tue, 25 Jan 2011) | 15 lines
Merged revisions 304005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) | 8 lines
DTMF attended transfers sometimes fail for no apparent reason.
The loop in feature_request_and_dial() can exit when Party C has answered
without processing an AST_CONTROL_ANSWER. Also sometimes an
AST_CONTROL_ANSWER never happens even though Party C has answered.
Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER.
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303962 | twilson | 2011-01-25 16:09:01 -0600 (Tue, 25 Jan 2011) | 30 lines
Merged revisions 303960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines
Merged revisions 303906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines
Guard against retransmitting BYEs indefinitely
In the case of an attended transfer (A calls B, A atxfers to C) where
A becomes unreachable before replying to Asterisk's BYE, Asterisk can
sometimes retransmit the BYE indefinitely. This is because
__sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out,
it will not ever be marked as ALREADYGONE, so when __sip_autodestruct
is called again, we end up starting the cycle over.
This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt
in the case of a BYE that has timed out. This should prevent Asterisk
from trying to transmit new BYE messages in the future.
Review: https://reviewboard.asterisk.org/r/1077/
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines
Merged revisions 303769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines
Merged revisions 303765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines
Sending out unnecessary PROCEEDING messages breaks overlap dialing.
Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing
through Asterisk. There is not enough information available at this point
to know if dialing is complete. The ast_exists_extension(),
ast_matchmore_extension(), and ast_canmatch_extension() calls are not
adequate to detect a dial through extension pattern of "_9!".
Workaround is to use the dialplan Proceeding() application early in
non-dial through extensions.
* Effectively revert issue #16789.
* Allow outgoing overlap dialing to hear dialtone and other early media.
A PROGRESS "inband-information is now available" message is now sent after
the SETUP_ACKNOWLEDGE message for non-digital calls. An
AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE
messages for non-digital calls.
* Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was
inconsistent with the cause codes.
* Added better protection from sending out of sequence messages by
combining several flags into a single enum value representing call
progress level.
* Added diagnostic messages for deferred overlap digits handling corner
cases.
(closes issue #17085)
Reported by: shawkris
(closes issue #18509)
Reported by: wimpy
Patches:
issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664)
Expanded upon issue18509_early_media_v1.8_v3.patch to include analog
and SS7 because of backporting requirements.
Tested by: wimpy, rmudgett
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines
Merged revisions 303677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines
Merged revisions 303676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines
Fix voicemail sequencing for file based storage.
A previous change was made to account for when the number of voicemail messages
exceeds the max limit to be handled properly, but it caused gaps in the messages
to not be properly handled. This has now been resolved.
In later non 1.4 branches, it appears that resequencing wasn't even occurring
due from what appears and accidental code removal.
(closes issue #18498)
Reported by: JJCinAZ
Patches:
bug18498v2.patch uploaded by jpeeler (license 325)
(closes issue #18486)
Reported by: bluefox
Patches:
bug18486.patch uploaded by jpeeler (license 325)
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
Merged revisions 303548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
escaped.
This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future.
The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs.
The unit tests for these functions have also been updated.
ABE-2705
Review: https://reviewboard.asterisk.org/r/1081/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Discussion on #asterisk on 2011-01-19:
(02:07:03 PM) boch: i wonder how to cycle all entries in a tree
(02:07:11 PM) leifmadsen: use While()
(02:07:17 PM) leifmadsen: you need to know the tree structure already though
(02:07:36 PM) boch: what you mean?
(02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan
(02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything
(02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script
(02:10:13 PM) boch: for example i need to know all entries in the tree
(02:10:15 PM) boch: got it
(02:10:20 PM) leifmadsen: exactly
(02:10:22 PM) leifmadsen: that's the problem
(02:10:22 PM) boch: thank you
(02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over.
(02:15:35 PM) leifmadsen: database shows everything
(02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>'
(02:16:41 PM) leifmadsen: guess no one has found that important enough to program :)
(02:16:52 PM) leifmadsen: at that point you should probably just use a relational database...
(02:17:10 PM) mateu: i dunno
(02:17:16 PM) mateu: seems pretty basic to me.
(02:17:16 PM) leifmadsen: me either
(02:17:19 PM) leifmadsen: sure does
(02:17:24 PM) leifmadsen: no one has programmed it though
(02:17:28 PM) ***leifmadsen shrugs
(02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen
(02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ?
(02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT()
(02:30:31 PM) leifmadsen: although HASHKEYS() might work
(02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS()
(02:31:01 PM) leifmadsen: DBKEYS() I guess?
(02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family?
(02:34:02 PM) leifmadsen: ya
(02:34:16 PM) leifmadsen: how would you iterate through layers of them?
(02:34:30 PM) leifmadsen: i.e. family/key/key/key ?
(02:34:43 PM) Corydon76-home: Essentially, yes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303153 | rmudgett | 2011-01-20 14:31:20 -0600 (Thu, 20 Jan 2011) | 22 lines
Merged revision 303098 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines
CC_INTERFACES does not get built correctly with local channels.
If local channels are used with CCSS, CC_INTERFACES gets garbage prepended
to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended
to it.
* Initialize the buffer to eliminate the prepended garbage.
* Filter out the empty interface strings to eliminate the latter.
* Added a diagnostic message if the CC_INTERFACES is ever empty.
JIRA ABE-2740
JIRA SWP-2848
..........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303107 | sruffell | 2011-01-20 13:57:31 -0600 (Thu, 20 Jan 2011) | 23 lines
Merged revisions 303106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) | 15 lines
main/features: Use POLLPRI when waiting for events on parked channels.
This change resolves a regression in the 1.6.2 when converting from
select to poll. The DAHDI timers use POLLPRI to indicate that the timer
fired, but features was not waiting for that flag. The result was no
audio for MOH when a call was parked and res_timing_dahdi was in use.
This patch is slightly modified from the one on the mantis issue. It does
not set an exception on the channel if the POLLPRI flag is set.
(closes issue #18262)
Reported by: francesco_r
Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines
Merged revisions 303008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines
Merged revisions 303007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines
Add new queue strategy to preserve behavior for when queue members moved to ao2.
Add queue strategy called "rrordered" to mimic old behavior from when queue
members were stored in a linked list.
ABE-2707
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines
Merged revisions 302917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines
Option L() is milliseconds, not seconds.
> Change the verbose output of option L() to say milliseconds and not seconds
> as the value is in milliseconds.
>
> (closes issue #18264)
> Reported by: jacco
> Patches:
> app_dial_patch.txt uploaded by lmadsen (license 10)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15 lines
Resolve a memory leak with the manager interface is disabled.
The intent of this check as it stands in previous versions of Asterisk was to
check if there are any active sessions. If there were no sessions, then the
function would return immediately and not bother with queueing up the manager
event to be processed. Since the conversion of storing sessions in an astobj2
container, this check will always pass. I changed it to go back to checking
what was intended.
The side effect of this was that if the AMI is disabled, the manager event
queue is populated anyway, but the code that runs to clear out the queue
never runs. A producer with no consumer is a bad thing.
Reported internally by kmorgan.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r302713 | rmudgett | 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) | 29 lines
Merged revisions 302693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines
Merged revisions 302671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines
DTMF transfer plays the wrong sounds for wrong number or other call failure.
* Set the default for features.conf.sample xferfailsound option to "beeperr"
as documented instead of "pbx-invalid" and corrected the use of it in DTMF
blind transfer (#1).
* Improved DTMF blind transfer handling of wrong numbers.
Most of the concerns in this issue were taken care of by the patch for
issue 17999: Issues with DTMF triggered attended transfers.
(closes issue #18379)
Reported by: gincantalupo
Tested by: rmudgett
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r302634 | tilghman | 2011-01-19 14:24:57 -0600 (Wed, 19 Jan 2011) | 22 lines
Merged revisions 302599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines
Kill zombies.
When we ast_safe_fork() with a non-zero argument, we're expected to reap our
own zombies. On a zero argument, however, the zombies are only reaped when
there aren't any non-zero forked children alive. At other times, we
accumulate zombies. This code is forward ported from res_agi in 1.4, so that
forked children are always reaped, thus preventing an accumulation of zombie
processes.
(closes issue #18515)
Reported by: ernied
Patches:
20101221__issue18515.diff.txt uploaded by tilghman (license 14)
Tested by: ernied
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r302555 | seanbright | 2011-01-19 14:03:32 -0500 (Wed, 19 Jan 2011) | 14 lines
Merged revisions 302554 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan 2011) | 7 lines
Don't call strlen() when we only need to look at the next character or two.
(closes issue #18042)
Reported by: wdoekes
Patches:
astsvn-inefficient-ast-uri-decode.patch uploaded by wdoekes (license 717)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r302549 | seanbright | 2011-01-19 13:43:11 -0500 (Wed, 19 Jan 2011) | 17 lines
Merged revisions 302548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines
Properly handle partial reads from fgets() when handling AGIs.
When fgets() failed with EAGAIN, we were continually decrementing the available
space left in our buffer, resulting in botched command handling.
(closes issue #16032)
Reported by: notahat
Patches:
agi_buffer_patch2.diff uploaded by fnordian (license 110)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, 19 Jan 2011) | 10 lines
Use appropriate type for requested format in chan_local.
We were passing and storing the requested format as an int instead of format_t
resulting in truncation.
(closes issue #18238)
Reported by: whizemen
Patches:
0018238_speex16.patch uploaded by whizemen (license 1143)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302413 65c4cc65-6c06-0410-ace0-fbb531ad65f3