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r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009) | 2 lines
Fix documentation for FIELDQTY.
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r198442 | eliel | 2009-05-30 22:09:06 -0400 (Sat, 30 May 2009) | 3 lines
Filter the say.o object, it is being added later.
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r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) | 11 lines
Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded.
if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash
when calling ast_unregister_timing_interface() with a NULL pointer.
(closes issue #15234)
Reported by: eliel
Patches:
timing_dahdi1.diff uploaded by eliel (license 64)
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r198438 | russell | 2009-05-30 20:40:02 -0500 (Sat, 30 May 2009) | 2 lines
Constification and remove some unused code.
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r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines
Merged revisions 198311 via svnmerge from
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r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines
Fix a crash that occurred when MWI SMDI messages expired.
(closes issue #14561)
Reported by: cmoss28
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r198434 | russell | 2009-05-30 20:19:30 -0500 (Sat, 30 May 2009) | 2 lines
Constify the ast_frame arg to ast_queue_frame().
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r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines
Properly terminate the receive buffer before sending to iksemel.
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
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r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines
Merged revisions 198370 via svnmerge from
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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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r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines
Merged revisions 198251 via svnmerge from
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r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines
Treat an empty FORWARD_CONTEXT the same way we treat a missing one.
(closes issue #15056)
Reported by: p_lindheimer
Patches:
05292009_bug15056.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer
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r198248 | file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines
When removing all packets from a dialog we also need to free the data if present.
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r198217 | eliel | 2009-05-29 21:04:57 -0400 (Fri, 29 May 2009) | 10 lines
Remove not used code in the Agent channel.
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/
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r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines
Suggesting that only a single timing module be loaded is no longer necessary.
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r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) | 2 lines
Improve handling of trying to ACK too many timer expirations.
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r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines
Resolve issues with choppy sound when using res_timing_pthread.
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
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r198088 | jpeeler | 2009-05-29 14:19:51 -0500 (Fri, 29 May 2009) | 9 lines
New signaling module to handle analog operations in chan_dahdi
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/
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r198139 | eliel | 2009-05-29 15:46:07 -0400 (Fri, 29 May 2009) | 15 lines
Simplify the Makefile and avoid needing to specify each object file.
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/
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r198083 | eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines
Apply anti-spam obfuscation to an email address.
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r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines
Merged revisions 198068 via svnmerge from
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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r198064 | file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines
Fix a memory leak of the write buffer when writing a file.
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r197959 | mmichelson | 2009-05-29 10:48:04 -0500 (Fri, 29 May 2009) | 16 lines
A few fixes to SIP with regards to connected line updates during transfers.
* 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
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r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May 2009) | 15 lines
Merged revisions 197998 via svnmerge from
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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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r197996 | file | 2009-05-29 14:51:06 -0300 (Fri, 29 May 2009) | 2 lines
Fix a bug where the default setting did not perform a remote bridge when it should have.
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r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009) | 2 lines
Trim trailing whitespace so that I can work on this bug without it bothering me. :-)
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r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009) | 8 lines
Update documentation in MixMonitor.
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
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r197861 | seanbright | 2009-05-28 18:42:27 -0400 (Thu, 28 May 2009) | 2 lines
Update references to downloads.digium.com to its new URL.
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r197824 | seanbright | 2009-05-28 17:50:27 -0400 (Thu, 28 May 2009) | 2 lines
Update references to bugs.digium.com and reviewboard.digium.com to the new URLs.
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r197738 | twilson | 2009-05-28 14:57:18 -0500 (Thu, 28 May 2009) | 19 lines
Add Calendaring support for Asterisk
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
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r197775 | kpfleming | 2009-05-28 15:36:49 -0500 (Thu, 28 May 2009) | 10 lines
Ensure that accidental calls to ast_string_field_free_memory() on embedded stringfield pools are safe.
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.
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r197740 | mmichelson | 2009-05-28 15:17:24 -0500 (Thu, 28 May 2009) | 7 lines
Treat 405 responses the same way we would a 501.
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.
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r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines
Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload.
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r197701 | mmichelson | 2009-05-28 13:48:56 -0500 (Thu, 28 May 2009) | 3 lines
Add missing lock to local_indicate function for connected line frames.
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r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines
Merged revisions 197562 via svnmerge from
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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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r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
Recorded merge of revisions 197588 via svnmerge from
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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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r197570 | file | 2009-05-28 12:23:29 -0300 (Thu, 28 May 2009) | 5 lines
Fix an incorrect call to ast_string_field_free_memory which caused a crash in the logger.
Since the message structure is allocated using ast_calloc_with_stringfields we do not need to
free the memory used for the stringfields as it will get freed when the message structure is.
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r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines
Merged revisions 197537 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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r197538 | file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines
Fix a bug in stringfields where it did not actually free the pools of memory.
(closes issue #15074)
Reported by: pj
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r197535 | seanbright | 2009-05-28 10:39:21 -0400 (Thu, 28 May 2009) | 3 lines
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.
Let's try that again, this time removing trailing whitespace and not leading
whitespace. I can't believe no one noticed.
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r197528 | seanbright | 2009-05-28 10:32:03 -0400 (Thu, 28 May 2009) | 2 lines
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.
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r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines
Merged revisions 197466 via svnmerge from
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r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines
Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting.
The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
(or it passes through unauthenticated) the proper nat flag is set.
(closes issue #13823)
Reported by: dimas
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r197374 | tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines
Revert commit 192032. This define is needed on Mac OS X.
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r197338 | russell | 2009-05-27 17:42:15 -0500 (Wed, 27 May 2009) | 2 lines
Don't do a pointer comparison before setting the remote address.
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r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May 2009) | 3 lines
Ensure that this header includes xmldoc.h, since it depends on it.
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r197260 | seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 lines
Use bash explicitly when calling build_tools/mkpkgconfig from the Makefile.
Since we use bashisms in build_tools/mkpkgconfig, we should call on bash
explicitly when running from the Makefile, otherwise we get errors during a
'make install.'
(closes issue #15209)
Reported by: seandarcy
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r197209 | tilghman | 2009-05-27 14:20:56 -0500 (Wed, 27 May 2009) | 12 lines
Recorded merge of revisions 197194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines
Use a different determinator on whether to print the delimiter, since leading fields may be blank.
(closes issue #15208)
Reported by: ramonpeek
Patch by me, though inspired in part by a patch from ramonpeek
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r197189 | seanbright | 2009-05-27 14:25:33 -0400 (Wed, 27 May 2009) | 2 lines
Remove a file sample configuration file that is no longer used.
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The bridge was terminating immediately after the attended transfer was
completed. The problem was because upon reentering ast_channel_bridge
nexteventts was checked to see if it was set and if so could possibly
return AST_BRIDGE_COMPLETE.
(closes issue #15183)
Reported by: andrebarbosa
Tested by: andrebarbosa, tootai, loloski
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r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines
Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in
the sample configuration files.
(closes issue #15207)
Reported by: seandarcy
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r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines
Display an error message when chan_alsa fails to load due to a missing
or inaccessible configuration file.
Before this change, when chan_alsa failed to load due to a missing or
inaccessible configuration file, no message would be displayed. With this
change, when chan_alsa fails to load due to a missing or inaccessible
configuration file, a message will be displayed.
(closes issue #14760)
Reported by: Nick_Lewis
Patches:
chan_alsa.c-confload.patch uploaded by Nick (license 657)
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r196948 | seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8 lines
Reset the terminal to the correct fg/bg after XML documenation is rendered.
(closes issue #15200)
Reported by: ajohnson
Patches:
05262009_xmldoc.patch uploaded by seanbright (license 71)
Tested by: ajohnson
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