This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.
During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.
Review: https://reviewboard.asterisk.org/r/379/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@223330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@220833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.
(closes issue #13165)
Reported by: tim_ringenbach
Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In addition, there's a bit of cleanup to the arguments and documentation, in which
I discovered that the last feature added to this application duplicated an option
(oops!) and changed that option so that it now works.
(closes issue #14909)
Reported by: junky
Patches:
__20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10)
Tested by: amilcar, junky, flujan, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines
If the user enters the same password as before, don't signal an error when the change does nothing.
(closes issue #15492)
Reported by: cbbs70a
Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
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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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r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines
When MOH is playing on the channel, announcements sent through the conference are not heard.
(closes issue #14588)
Reported by: voipas
Patches:
20090716__issue14588__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, twisted, tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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 :-)
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The store macro was not getting called preventing storage of IMAP greetings
at all. This has been corrected along with fixing checking if the
imapgreetings option is turned on to store the greeting in IMAP. Lastly,
the attachment filename was incorrectly using the full path instead of just
the basename, which was causing problems with retrieval of the greeting.
(closes issue #14950)
Reported by: noahisaac
(closes issue #15729)
Reported by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch modifies app_voicemail's response to mailbox status subscriptions
(via the internal event system) to ensure that a subscription triggers an
explicit poll of the mailbox, so the subscriber can get an immediate cached
event with that status. Previously, the cache was only populated with the
status of non-realtime mailboxes.
(closes issue #15717)
Reported by: natmlt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/trunk@213404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch makes some small changes to handle watchdog timeouts in a better way,
and also uses a 'cleaner' method of including the spandsp header files.
(closes issue #14769)
Reported by: andrew
Patches:
app_fax-20090406.diff uploaded by andrew (license 240)
v1-14769.patch uploaded by dimas (license 88)
Tested by: freh, deti, caspy, dimas, sgimeno, Dovid
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Kevin P. Fleming
Tilghman Lesher
Joshua Colp
Jeff Peeler
Terry Wilson
Mark Michelson
Matthias Nick
Sean Bright
Matthew Nicholson
Olle Johansson
Dwayne M. Hubbard
David Vossel
Russell Bryant