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We require box numbers, not names as the documentation implies.
(issue #14740)
Reported by: pj
Patches:
__20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10)
Tested by: seanbright, lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221085 f38db490-d61c-443f-a65b-d21fe96a405b
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fact that app_confbridge does not automatically answer the channel.
(closes issue #15964)
Reported by: shrift
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220904 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@220833 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@220344 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
Implicitly sending a progress signal breaks some applications.
Call Progress() in your dialplan if you explicitly want progress to be sent.
(Reverts change 216430, closes issue #15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
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(closes issue #15739)
Reported by: DLNoah, jeffg
Patches:
20090914__issue15739.diff.txt uploaded by tilghman (license 14)
20090922__issue15739.diff.txt uploaded by tilghman (license 14)
Tested by: DLNoah, jeffg
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219987 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@219818 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15696)
Reported by: fhackenberger
Patches:
maxsecs.patch uploaded by fhackenberger (license 592)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219412 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219230 f38db490-d61c-443f-a65b-d21fe96a405b
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hangs up.
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: http://svn.digium.com/svn/asterisk/trunk@219105 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@218731 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@218579 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines
Don't say "Please try again" if we don't give the user another chance to try again.
(issue #15055, SWP-129)
Reported by: jthurman
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@218361 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines
Ensure we don't pickup ourselves when doing pickup by exten.
(closes issue #15100)
Reported by: lmsteffan
Patches:
(modified) pickup.patch uploaded by lmsteffan (license 779)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@218224 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines
Don't ring another channel, if there's not enough time for a queue member to answer.
(Fixes AST-228)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217990 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by ebroad in #asterisk-bugs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217286 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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: http://svn.digium.com/svn/asterisk/trunk@216593 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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: http://svn.digium.com/svn/asterisk/trunk@215338 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@213833 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@213697 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15146)
Reported by: arabe
Patches:
asterisk-trunk.patch uploaded by arabe (license 786)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213414 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@213404 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r213283 | jpeeler | 2009-08-20 14:53:34 -0500 (Thu, 20 Aug 2009) | 2 lines
Make all the symbols for the C-client callbacks global
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) | 8 lines
Fixes memory leak caused by incorrectly freeing mixmonitor
(closes issue #15699)
Reported by: edantie
Patches:
mixmonitor.patch uploaded by edantie (license 862)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212844 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15720)
Reported by: tobias_e
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212627 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug 2009) | 10 lines
This patch adds additional checking when generating queue log TRANSFER events.
The additional checks prevent generation of false TRANSFER events in certain situations.
(closes issue #14536)
Reported by: aragon
Patches:
queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
Tested by: aragon, mnicholson
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15617)
Reported by: rain
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211232 f38db490-d61c-443f-a65b-d21fe96a405b
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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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(closes issue #15557)
Reported by: rain
Patches:
20090723__issue15557.diff.txt uploaded by tilghman (license 14)
Tested by: rain
Review: https://reviewboard.asterisk.org/r/316/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210908 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@210777 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@209839 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15595)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209554 f38db490-d61c-443f-a65b-d21fe96a405b
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Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages,
clean up some looping logic, and correct an improper use of ast_free() for
freeing an ast_frame.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209279 f38db490-d61c-443f-a65b-d21fe96a405b
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In receive mode, if the channel that ReceiveFAX is running on supports T.38,
we should *always* attempt to switch T.38, rather than listening for an incoming
CNG tone and only triggering on that. The channel may be using a low-bitrate
codec that distorts the CNG tone, the sending FAX endpoint may not send CNG
at all, or there could be a variety of other reasons that we don't detect it,
but in all those cases if T.38 is available we certainly want to use it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209256 f38db490-d61c-443f-a65b-d21fe96a405b
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"initialize"
(closes issue #15571)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209098 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines
Do not log an ERROR if autoservice_stop() returns -1.
This does not indicate an error. A return of -1 just means that the channel
has been hung up.
(reported in #asterisk-dev)
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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: http://svn.digium.com/svn/asterisk/trunk@208464 f38db490-d61c-443f-a65b-d21fe96a405b
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This x is one crafty little bugger...
It was used for 2 different things (one of which was only done on PPC) in 1.4.
One of the uses were removed in trunk, and with it went the declaration.
(closes issue #14038)
Reported by: ffloimair
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208113 f38db490-d61c-443f-a65b-d21fe96a405b
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A shallow pointer copy was causing an ast_party_connected_line
structure to be freed multiple times, thus causing a crash.
(closes issue #15441)
Reported by: lmsteffan
Patches:
15441.patch uploaded by mmichelson (license 60)
Tested by: lmsteffan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208017 f38db490-d61c-443f-a65b-d21fe96a405b
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This helps to prevent a crash which may occur due to our freeing
garbage due to a struct being uninitialized.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207522 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by "Hoggins!" on asterisk-dev list.
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lengths more consistent.
(closes issue #15493)
Reported by: lasko
Patches:
meetme.diff uploaded by lasko (license 833)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206567 f38db490-d61c-443f-a65b-d21fe96a405b
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If a redirecting control frame is processed or a call forward occurs,
we need to reset the sentringing flag so that we can send another ringing
indication to the phone that may contain a connected line update.
AST-164
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206455 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15427)
Reported by: brushtyler
Patches:
app_voicemail.c.diff uploaded by brushtyler (license 821)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206185 f38db490-d61c-443f-a65b-d21fe96a405b
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