Russell Bryant [Wed, 6 Aug 2008 18:58:37 +0000 (18:58 +0000)]
Remove the use of --no-print-directory when compiling subdirectories.
This allows vim :make functionality to work properly when errors have
occurred in the build. Without printing the directories, vim did not
know how to find the file that the error occurred in.
If the extra bit of build noise annoys anyone, just let me know, and
I'll make this optional.
Mark Michelson [Wed, 6 Aug 2008 15:58:40 +0000 (15:58 +0000)]
Since adding the AST_CONTROL_SRCUPDATE frame type,
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
Mark Michelson [Wed, 6 Aug 2008 00:25:10 +0000 (00:25 +0000)]
Merging the issue11259 branch.
The purpose of this branch was to take into account
"burps" which could cause jitterbuffers to misbehave.
One such example is if the L option to Dial() were used
to inject audio into a bridged conversation at regular
intervals. Since the audio here was not passed through
the jitterbuffer, it would cause a gap in the jitterbuffer's
timestamps which would cause a frames to be dropped for a
brief period.
Now ast_generic_bridge will empty and reset the jitterbuffer
each time it is called. This causes injected audio to be handled
properly.
ast_generic_bridge also will empty and reset the jitterbuffer
if it receives an AST_CONTROL_SRCUPDATE frame since the change
in audio source could negatively affect the jitterbuffer.
All of this was made possible by adding a new public API call
to the abstract_jb called ast_jb_empty_and_reset.
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
Tilghman Lesher [Tue, 5 Aug 2008 21:34:46 +0000 (21:34 +0000)]
In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID
was perverted. This change reverts IAX2 to the original meaning, which was,
that the callerid set on the client should be overridden on the server, even if
that means the resulting callerid is blank. In other words, if you set
"callerid=" in the IAX config, then the callerid should be overridden to blank,
even if set on the client. Note that there's a distinction, even on realtime,
between the field not existing (NULL in databases) and the field existing, but
set to blank (override callerid to blank).
Specify codecs in callfiles and manager, to allow video calls to be set up
from callfiles and AMI.
(closes issue #9531)
Reported by: Geisj
Patches:
20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14)
20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
OK, a few days of debugging, a bunch of instrumentation
in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid
notebook pages of notes later, I have made the small
tweek necc. to get the start time right on the second
CDR when:
A Calls B
B answ.
A hits Xfer button on sip phone,
A dials C and hits the OK button,
A hangs up
C answers ringing phone
B and C converse
B and/or C hangs up
But does not harm the scenario where:
A Calls B
B answ.
B hits xfer button on sip phone,
B dials C and hits the OK button,
B hangs up
C answers ringing phone
A and C converse
A and/or C hangs up
The difference in start times on the second CDR is because
of a Masquerade on the B channel when the xfer number is
sent. It ends up replacing the CDR on the B channel with
a duplicate, which ends up getting tossed out. We keep
a pointer to the first CDR, and update *that* after the
bridge closes. But, only if the CDR has changed.
I hope this change is specific enough not to muck
up any current CDR-based apps. In my defence, I
assert that the previous information was wrong,
and this change fixes it, and possibly other
similar scenarios.
I wonder if I should be doing the same thing
for the channel, as I did for the peer, but
I can't think of a scenario this might affect.
I leave it, then, as an exersize for the users,
to find the scenario where the chan's CDR
changes and loses the proper start time.
Mark Michelson [Thu, 31 Jul 2008 15:56:18 +0000 (15:56 +0000)]
Add more timeout checks into app_queue, specifically
targeting areas where an unknown and potentially
long time has just elapsed. Also added a check
to try_calling() to return early if the timeout
has elapsed instead of potentially setting a negative
timeout for the call (thus making it have *no* timeout
at all).
Qwell pointed out, via IRC, that the previous fix only worked when explicitly
set. When nothing is set, and the option is implied, it breaks, because
configure sets the prefix to 'NONE'. Fixing.
launch_netscript sometimes returns -1, which fails to set AGISTATUS. Map
failure to -1, so that AGISTATUS is always set.
(closes issue #13199)
Reported by: smw1218
Mark Michelson [Tue, 29 Jul 2008 15:54:48 +0000 (15:54 +0000)]
Merging the imap_consistency branch.
The main aim of this branch was to make the IMAP
code function in the same manner as the ODBC code
does, eliminating the need for so many IMAP-specific
code chunks. The focal point of all of this work was
to make the various macros (e.g. RETRIEVE, DISPOSE)
functionally equivalent.
While doing the above work, I also fixed a few bugs
that I came across in my testing. Among these were
1. Fixed message forwarding. This was completely
broken when using IMAP.
2. Fixed the inability to save new messages as old
and vice versa.
3. Fixed the "delete" options in voicemail.conf when
using IMAP storage.
Even though a few bugs were fixed and the code is
a lot more consistent, the one thing that was *not*
improved in this branch was performance.
The merge of this to trunk may not come immediately
due to the amount of work it will probably involve.
Detect when sox fails to raise the volume, because sox can't read the file.
(closes issue #12939)
Reported by: rickbradley
Patches:
20080728__bug12939.diff.txt uploaded by Corydon76 (license 14)
Tested by: rickbradley
Fix some errant device states by making the devicestate API more strict in
terms of the device argument (only without the unique identifier appended).
(closes issue #12771)
Reported by: davidw
Patches:
20080717__bug12771.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw, jvandal, murf
Kevin P. Fleming [Wed, 23 Jul 2008 20:42:30 +0000 (20:42 +0000)]
make some more changes to the dahdi/zap channel name support stuff to ensure allthe globals are 'const', and clean up mmichelson's changes to app_chanspy to simplify the code
Mark Michelson [Wed, 23 Jul 2008 19:39:47 +0000 (19:39 +0000)]
As suggested by seanbright, the PSEUDO_CHAN_LEN in
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
Mark Michelson [Wed, 23 Jul 2008 17:50:01 +0000 (17:50 +0000)]
Small cleanup. Move the declaration of the DAHDI_SPANINFO
variable to the block where it is used. This allows one
less #ifdef HAVE_PRI to clutter things up.
Thanks to Tzafrir for pointing this out on #asterisk-dev
Kevin P. Fleming [Wed, 23 Jul 2008 15:52:29 +0000 (15:52 +0000)]
ensure that after a channel is created, if it happened to be in 'channel alarm' state, when that alarm clears we won't generate a spurious 'alarm cleared' message
Kevin P. Fleming [Wed, 23 Jul 2008 11:52:18 +0000 (11:52 +0000)]
minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool)
another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users.
Mark Michelson [Tue, 22 Jul 2008 22:14:24 +0000 (22:14 +0000)]
Allow Spiraled INVITEs to work correctly within Asterisk.
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
Kevin P. Fleming [Tue, 22 Jul 2008 22:11:01 +0000 (22:11 +0000)]
fix up namespace pollution for dahdi_chan_mode enum
correct registration of AMI actions in chan_dahdi; in zap-only mode, only register the Zap flavors of the actions (and use Zap prefixes for headers and acks), but in dahdi+zap mode, register both Zap and DAHDI flavors of actions
Kevin P. Fleming [Tue, 22 Jul 2008 21:17:23 +0000 (21:17 +0000)]
ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense'
Olle Johansson [Tue, 22 Jul 2008 20:10:26 +0000 (20:10 +0000)]
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
Fix a bug in 1.4 branch with iax2 channels not being removed when a call was rejected (from the calling box, not the box that denied the
registration). Related to revisions 132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently tested 1.4 with a backport from those revisions,
so I didn't see this problem (oops).
Kevin P. Fleming [Sat, 19 Jul 2008 16:45:52 +0000 (16:45 +0000)]
grant a license exception to allow distribution of Asterisk binaries that use the UW IMAP Toolkit (which is licensed under a non-GPL-compatible license)
Revert part of issue #5620 (revision 6965) as it appears that it was in error.
This should fix talk call progress on analog lines.
(closes issue #12178)
Reported by: michael-fig
Patches:
20080717__bug12178.diff.txt uploaded by Corydon76 (license 14)
Russell Bryant [Wed, 16 Jul 2008 20:47:53 +0000 (20:47 +0000)]
Always ensure that the channel's tech_pvt reference is NULL after calling the
destroy callback.
(closes issue #13060)
Reported by: jpgrayson
Patches:
chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license 492)
Mark Michelson [Wed, 16 Jul 2008 20:23:02 +0000 (20:23 +0000)]
Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a
better way.
Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.
Mark Michelson [Wed, 16 Jul 2008 18:57:34 +0000 (18:57 +0000)]
Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.
Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.
Steve Murphy [Wed, 16 Jul 2008 17:53:43 +0000 (17:53 +0000)]
(closes issue #13090)
Reported by: murf
The problem was that, esoteric as it is, because the hangerupper
context immediately preceded the std-priv-extent macro, that
the checking code accidentally would fall from traversing hangerupper
into the std-priv-exten macro, where it would hit the hangerupper
in the 'includes', and proceed into an infinite recursion.
A small fix to traverse into the statements of the context instead
of the context solves this issue.
I also added some commented out printfs for debug, which were pretty
handy in the face of a dorky gdb.
This was a problem around since the package was first written;
but evidently pretty rare in turning up in the field.
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
Override the callerid in all cases when the callerid is set in the user, not
just when a remote callerid is set. Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
Reported by: dimas
Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
Mark Michelson [Mon, 14 Jul 2008 17:50:21 +0000 (17:50 +0000)]
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
notify the user that dnsmgr refresh wont work when dnsmgr is not enabled.
Previously this command would automagically appear and disappear.
This was confusing.
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