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r198665 | tilghman | 2009-06-01 15:07:04 -0500 (Mon, 01 Jun 2009) | 7 lines
If using the old deprecated format, a reload would cause the class to disappear.
(closes issue #14759)
Reported by: lidocaineus
Patches:
20090518__issue14759.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved more static docs to XML (pplications and manager actions):
Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
aji_io_recv takes the maximum number of bytes to read (instead of the total
buffer size), so we have to subtract 1 from our buffer size. Without this, when
we receive packets that are larger than our buffer, iksemel will choke and
things get wonky.
(closes issue #15232)
Reported by: lp0
Patches:
05302009_res_jabber.c.patch uploaded by seanbright (license 71)
Tested by: seanbright, lp0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines
Properly terminate AMI JabberSend response messages.
The response message (either Error or Success) needs an extra trailing \r\n
after the fields to inform the client that the message is complete.
(closes issue #14876)
Reported by: srt
Patches:
05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71)
asterisk_14876.patch uploaded by srt (license 378)
trunk-14876-2.diff uploaded by phsultan (license 73)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code was there because of the AgentCallbackLogin() application.
->loginchan[] member was only used by AgentCallbackLogin().
Agent where dumped to astdb if they where logged in using AgentCallbacklogin()
so they are not being dumper anymore.
Review: https://reviewboard.asterisk.org/r/267/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The situation that caused this problem was when continuous mode was being
turned on and off while a rate was set for a timing interface. A very easy
way to replicate this bug was to do a Playback() from behind a Local channel.
In this scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about every 20 ms
as frames get queued on to the PBX side channel from the other side of the
Local channel.
Essentially, this module treated continuous mode and a set rate as mutually
exclusive states for the timer to be in. When I dug deep enough, I observed
the following pattern:
1) Set timer to tick every 20 ms.
2) Wait almost 20 ms ...
3) Continuous mode gets turned on for a queued up frame
4) Continuous mode gets turned off
5) The timer goes back to its tick per 20 ms. state but starts counting
at 0 ms.
6) Goto step 2.
Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick,
but not most of the time. This is what produced the choppy sound (or sometimes
no sound at all).
Now, the module treats continuous mode and a set rate as completely independent
timer modes. They can be enabled and disabled independently of each other and
things work as expected.
(closes issue #14412)
Reported by: dome
Patches:
issue14412.diff.txt uploaded by russell (license 2)
issue14412-1.6.1.0.diff.txt uploaded by russell (license 2)
Tested by: DennisD, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Instead of specifying every object file, use make's magic to generate
it.
This will generate less conflicts in team branches when a new file is
added in trunk.
(closes issue #15226)
Reported by: eliel
Patches:
makefile uploaded by eliel (license 64)
Review: http://reviewboard.asterisk.org/r/269/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This branch splits all the analog signaling logic out of chan_dahdi.c into
sig_analog.c. Functionality in theory should not change at all. As noted
in the code, there is still some unused code remaining that will be cleaned
up in a later commit.
Review: https://reviewboard.asterisk.org/r/253/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines
Fix 'make config' target for Slackware.
There was a missing semi-colon after the echo statement in the Makefile that was
causing problems for some users. Fix suggested by reporter.
(closes issue #15225)
Reported by: pdavis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Set the invitestate to INV_CALLING when we send a connected line reinvite.
This prevents us from potentially rapid-firing reinvites to a single peer.
* Use the astdb to store a peer's allowed methods. This prevents us from sending
an UPDATE during the interval between startup and the peer's first registration
if the peer does not support the UPDATE method.
* Handle Polycom's method of indicating allowed methods in REGISTER. Instead of
using an Allow header, they place the allowed methods in a methods= parameter
in the Contact header.
ABE-1873
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I still need to set up tests to make sure my examples are completely correct,
but I ran out of time tonight and felt that they at least would give an idea as
to how to use calendaring. I will try to test the examples and do some cleanup
on the docs tomorrow night.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize away the Local
channel when using this option.
(closes issue #14829)
Reported by: licedey
Tested by: mmichelson, licedey, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for a stringfield manager structure (and pool) structure to be allocated
as part of a larger structure allocation (using ast_calloc_with_strinfields()); when
this is done, the stringfield pool cannot be separately freed, but users of the tructure
may not be aware (and shouldn't have to be aware) of whether the pool was embedded.
This patch modifies the behavior so that they can always call ast_string_field_free_memory()
and the function will do the right thing for both embedded and non-embedded situations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This makes sure that we mark a method as being unallowed if we
receive a 405 response so that we don't continue to try to
send that same type of message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).
Features include:
Querying a calendar for events over a specific time range
Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange only)
Handling calendar event notifications through the dialplan
(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines
Use the address we already know when reloading a peer with nat=yes.
If we already have an address for a peer, and we are reloading the sip
configuration, try to use that address to contact the peer, instead of
getting it from the Contact.
(closes issue #15194)
Reported by: ibc
Patches:
sip.patch uploaded by eliel (license 64)
Tested by: manwe
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
Allow for media to arrive from an alternate source when responding to a reinvite with 491.
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
