https://origsvn.digium.com/svn/asterisk/trunk
........
r231556 | dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines
app_queue crashes randomly, often during call-transfers
This patch adds a ref to the queue_ent object's parent call_queue
in queue_exec() so the call_queue won't be destroyed
while the the queue_ent still holds a pointer to it.
(closes issue 0015686)
Tested by: dvossel, aragon
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@231560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
Fix the fix for chanspy option o
In 224178, I assumed the uploaded patch was correct as it had received positive
feedback. The flags were being checked in the incorrect location. Upon testing
the fix this time it was also found that the flags from the dialplan weren't
being copied to the chanspy_translation_helper.
(closes issue #16167)
Reported by: marhbere
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines
Merged revisions 226889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines
Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
while the called party had not yet answered.
This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction
file under all scenarios. This was done to preserve the behavior that has existed for quite some time.
(closes issue #14674)
Reported by: ulogic
Patches:
bug14674.patch uploaded by jpeeler (license 325)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This backport resolves some issues handling audio frames during FAX processing,
and ensures that the FAX application doesn't accidentally get notified of a T.38
switchover at the end of a successful FAX.
(issue #16127)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines
Merged revisions 223804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines
Ensure ringing continues for branched calls after progress is received
While waiting for an answer, don't send progress for branched calls
for which ringing was sent.
(closes issue #15028)
Reported by: fnordian
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines
Remove automatic switching from T.38 to voice mode in chan_sip.
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and changes app_fax to
explicitly switch off T.38 mode when the FAX transmission process ends.
(closes issue #16025)
Reported by: jamicque
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines
Initiate T.38 switchover when acting as called party, regardless of FAX direction.
SendFAX() and ReceiveFAX() can be given options to indicate whether they should
act as the calling or called party; this mode should be used to decide whether
to initiate a switchover to T.38, not the direction that the FAX transfer will
take place.
(closes issue #16039)
Reported by: jamicque
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines
Recorded merge of revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines
Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't
actually being expunged. When imapgreetings was set to no, the file based
message was not being deleted at all. All good now!
(closes issue #14949)
Reported by: noahisaac
Patches:
vm_tempgreeting_removal.patch uploaded by noahisaac (license 748),
modified by me
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@220834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines
Merged revisions 219816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines
When IMAP variables were changed during a reload, Voicemail did not use the new values.
This change introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to expire
normally.
(closes issue #15934)
Reported by: viniciusfontes
Patches:
20090922__issue15934.diff.txt uploaded by tilghman (license 14)
Tested by: viniciusfontes
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@219819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines
Merged revisions 216430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@216645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r215338 | dhubbard | 2009-09-01 20:16:59 -0500 (Tue, 01 Sep 2009) | 18 lines
Merged revisions 215270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines
Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names
In general channel names are in the form Foo/Bar-Z, but the channel name
could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to
truncate the channel name at the last hyphen.
(closes issue #15810)
Reported by: dhubbard
Patches:
dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@215352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) | 14 lines
Fix storage of greetings when using IMAP_STORAGE
Fix checking if the imapgreetings option is turned on to store the greeting
in IMAP.
(closes issue #14950)
Reported by: noahisaac
Patches:
14950.patch uploaded by mmichelson (license 60)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@213837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009) | 12 lines
Fix greeting retrieval from IMAP
Properly check for the current voicemail state and if it doesn't exist,
create it.
(closes issue #14597)
Reported by: wtca
Patches:
14597_v2.patch uploaded by mmichelson (license 60)
Tested by: jpeeler
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@213411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) | 21 lines
Merged revisions 211038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines
QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername.
This is a partial revert of revision 82590, which was an attempted cleanup,
but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended
as a method by which component interfaces could be queried from the queue.
Membername isn't useful here, because that field cannot be used to obtain
further information about the member. See the documentation on
QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various
AMI commands which take a member argument for further justification.
(closes issue #15664)
Reported by: rain
Patches:
app_queue-queue_member_list.diff uploaded by rain (license 327)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@211044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r209839 | russell | 2009-08-01 06:02:07 -0500 (Sat, 01 Aug 2009) | 20 lines
Merged revisions 209838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines
Modify how Playtones() is used in Milliwatt() to resolve gain issue.
When Milliwatt() was changed internally to use Playtones() so that the proper
tone was used, it introduced a drop in gain in the output signal. So, use
the playtones API directly and specify a volume argument such that the output
matches the gain of the original Milliwatt() code.
(closes issue #15386)
Reported by: rue_mohr
Patches:
issue_15386.rev2.diff uploaded by russell (license 2)
Tested by: rue_mohr
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
