There is no section 22.2.2 in rfc 3261. I believe 26.2.2 is what was meant:
Note that in the SIPS URI scheme, transport is independent of TLS,
and thus "sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
........
Steve Murphy [Wed, 24 Dec 2008 00:52:12 +0000 (00:52 +0000)]
Merged revisions 166665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
Due to non-symmetrical updating, I had some fairly
interesting conflicts to straighten out in this
release. The changes were such that I was compelled
to run thru all the same tests as trunk, which turned
up some problems, which I fixed.
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Fix a deadlock relating to channel locks and autoservice
It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of
autoservice.
The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.
The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.
In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.
It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.
Russell Bryant [Mon, 22 Dec 2008 17:14:18 +0000 (17:14 +0000)]
Merged revisions 166282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r166282 | russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines
Introduce ast_careful_fwrite() and use in AMI to prevent partial writes.
This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes. It is currently used in AMI to
address the issue that has been reported. However, there are probably a few
other places where this could be used.
(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/
Mark Michelson [Mon, 22 Dec 2008 16:30:25 +0000 (16:30 +0000)]
When merging the fix for issue #14118, I found that
the issue didn't affect 1.6.0, but in this case that's
not an especially good thing, because it means that
the fix for issue #13496 was not merged into 1.6.0 in
the first place. This commit kills two birds with one
stone by putting both fixes in the 1.6.0 branch
Mark Michelson [Mon, 22 Dec 2008 16:12:34 +0000 (16:12 +0000)]
Back these changes out for now. The initial refcounting
for filestreams didn't get merged into 1.6.0. I'll have
to get that in first and then put this change in
Fix a file playback crash and explicitly initialize values in func_timeout.c
A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.
A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.
Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port.
(closes issue #13628)
Reported by: pananix
Patches:
bug13628.patch uploaded by jpeeler (license 325)
Tested by: file, blitzrage
........
Russell Bryant [Mon, 22 Dec 2008 14:18:08 +0000 (14:18 +0000)]
Merged revisions 166258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r166258 | russell | 2008-12-22 08:16:54 -0600 (Mon, 22 Dec 2008) | 26 lines
Remove AST_PBX_KEEPALIVE usage from res_agi.
This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage
was for the AGI command, "asyncagi break". This patch removes this feature.
Normally, a feature would not be removed like this. However, this code is
broken and usage of it will result in a memory leak.
Usage of this feature will make the AGI code return a result of
AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed
ownership of the channel. The channel thread will exit without destroying the
channel. Unfortunately, _no_ thread has ownership of the channel at this
point. There are a couple of serious problems here:
1) The only way to recover the caller is to issue a channel redirect. This
will work, but this will be done with a masquerade, and the old ast_channel
structure will be lost.
2) Until the channel redirect happens, there is no code servicing the channel.
That means nothing is reading audio or handling events coming from the
channel. This is very bad.
The recommended way to get this same "break" functionality is to issue the
redirect while the channel is still being handled by the AGI code. That way,
there will be no memory leak, and there will be no period of time that the
channel is not being serviced.
This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
Russell Bryant [Fri, 19 Dec 2008 14:43:13 +0000 (14:43 +0000)]
Blocked revisions 165883 via svnmerge
........
r165883 | russell | 2008-12-19 08:42:51 -0600 (Fri, 19 Dec 2008) | 8 lines
Introduce commit message formatting guidelines.
This documents the recommended outline to use for commit message. It also
covers information on special tags that can be used in commit messages, as well
as the template functionality that is available on bugs.digium.com.
Add mutexes around accesses to the IMAP library interface. This prevents
certain crashes, especially when shared mailboxes are used.
(closes issue #13653)
Reported by: howardwilkinson
Patches:
asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590)
Tested by: jpeeler
........
................
Russell Bryant [Thu, 18 Dec 2008 21:45:38 +0000 (21:45 +0000)]
Merged revisions 165801 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008) | 19 lines
Merged revisions 165796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines
Make ast_carefulwrite() be more careful.
This patch handles some additional cases that could result in partial writes
to the file description. This was done to address complaints about partial
writes on AMI.
(issue #13546) (more changes needed to address potential problems in 1.6)
Reported by: srt
Tested by: russell
Review: http://reviewboard.digium.com/r/99/
Russell Bryant [Thu, 18 Dec 2008 19:40:14 +0000 (19:40 +0000)]
Merged revisions 165723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines
Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.
This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source. While this usage was perfectly safe,
there are others that are problematic. Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.
Further changes to get rid of KEEPALIVE and related code is being done by
murf. There is a patch up for that on review 29.
Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
(closes issue #13545)
Reported by: davidw
........
................
Do not crash if we are not passed in a followme id.
(closes issue #14106)
Reported by: ys
Patches:
app_followme.c.2.diff uploaded by ys (license 281)
........
................
Remove duplicate code from the ast_str API. We now use __AST_STR_* to
access 'struct ast_str' members, but this must only be used inside the API implementation.
Reverse the fix from issue #6176 and add proper handling for that issue.
(Closes issue #13962, closes issue #13363)
Fixed by myself (license 14)
........
................
Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING.
........
Use the create_vm_state_from_user function in a place where
it was not being used before. Also, I've moved the urgent
folder check in messagecount() up a bit so that the flow is
a bit better.
This was something I noticed while taking a look at issue
#13973, although I don't think this is the underlying cause
of the issue.
A possibly "horrible fix" for a "horribly broken"
situation.
As stuff shifts around in the asterisk code, the
miscellaneous inclusions from the standalone stuff
gets broken. There's no easy fix for this situation.
I made sure that everything in utils builds without
problem ***AND*** that aelparse runs the regressions
correctly with the following make menuselect options
both on and off:
I think from now on, I'm going to #undef
all these features in the various utils native
files; I guess I could do the same for the
copied-in files, surrounded by STANDALONE ifdef.
A standalone isn't going to care about threads,
mutexes, etc.
After looking through SIP registration code most of the day, this
is one of the few things I could find that was just plain wrong.
Even though it probably isn't possible for it to happen, it seems weird
to have code that checks if a pointer is NULL and then immediately dereferences
that pointer if it was NULL.
Russell Bryant [Tue, 16 Dec 2008 21:13:18 +0000 (21:13 +0000)]
Merged revisions 164877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines
Merged revisions 164876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines
Do not dereference the channel if AST_PBX_KEEPALIVE has been returned.
This is a bug I noticed while looking at the code for app_macro. This return code
means that another thread has assumed ownership of the channel and it can no longer
be touched. (I hate this return code with a passion, by the way.)
Russell Bryant [Tue, 16 Dec 2008 20:43:16 +0000 (20:43 +0000)]
Merged revisions 164807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008) | 17 lines
Merged revisions 164806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines
Add "restart gracefully" to the AMI blacklist of CLI commands.
"module unload" was already identified as a command that can not be used
from the AMI. "restart gracefully" effectively unloads all modules, and will
run in to the same problems.
Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time.
(closes issue #13217)
Reported by: cervajs
........
introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl.
(closes issue #13433)
Reported by: mvanbaak
Patches:
2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7)
with two free calls replaced with ast_free based on feedback on reviewboard
Review:
http://reviewboard.digium.com/r/91/
........
OK, Well this issue has had its share of flip-flopping.
I found the following:
1. the code in question, in ext_cmp1 in pbx.c, would not
allow two extensions that vary only by any dashes contained
within them, to be defined in the same context.
2. for input dialstrings, dashes are NOT ignored.
So, skipping them when sorting patterns seemed a bit silly.
Thus, you might declare ext 891 in a context, but
if you try dialing 8-9-1, it will NOT match 891.
So, I proposed to remove the code from ext_cmp1 to
skip the spaces and dashes. Just kept us from
declaring 891 and 8-9-1 in the same context,
forcing users to generate otherwise uselessly
obfuscated dialplan code to get the same effect.
Then, I tried out 1.4, and found that:
1. you can declare 891 and 8-9-1 in the
same context!
2. You can't define 891, and have 8-9-1 match
it! Nor can you define 8-9-1, and have 891
match it!
So, it appears that my proposal simply restores
the pbx to behaving as it did in 1.4.
Russell Bryant [Tue, 16 Dec 2008 17:15:43 +0000 (17:15 +0000)]
Merged revisions 164737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) | 22 lines
Merged revisions 164736 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines
Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
One issue was that the ast_mutex_* API was being used within the context of the
thread local data destructors. We would go off and allocate more thread local data
while the pthread lib was in the middle of destroying it all. This led to a memory
leak.
Another issue was an invalid argument being provided to the the object_add
API call.
(closes issue #13678)
Reported by: ys
Tested by: Russell
Be more detailed about why the include did not get included.
(closes issue #14071)
Reported by: kshumard
Patches:
pbx_config.patch.improvederroroutput.txt uploaded by kshumard (license 92)
........
Russell Bryant [Tue, 16 Dec 2008 16:02:22 +0000 (16:02 +0000)]
Merged revisions 164675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines
Merged revisions 164672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines
Fix a memory leak related to the use of the "setvar" configuration option.
The problem was that these variables were being appended to the list of vars
on the sip_pvt every time a re-registration or re-subscription came in.
Since it's just a waste of memory to put them there unless the request was an
INVITE, then the fix is to check the request type before copying the vars.
(closes issue #14037)
Reported by: marvinek
Tested by: russell
When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself.
(closes issue #13634)
Reported by: performer
........
I added a sentence to clarify why - and ' ' are ignored in patterns
as per bug 14076. Leif says he'll put some stuff about it in the
extensions.conf sample, etc.
I was getting this warning during a compile
on a 64-bit machine running ubuntu server 8.10,
and gcc-4.3.2:
[CXXi] chan_vpb.ii -> chan_vpb.oo
cc1plus: warnings being treated as errors
In file included from /home/murf/asterisk/trunk/include/asterisk/utils.h:671,
from chan_vpb.cc:46:
/home/murf/asterisk/trunk/include/asterisk/strings.h: In function ‘char* ast_str_truncate(ast_str*, ssize_t)’:
/home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: comparison between signed and unsigned integer expressions
make[1]: *** [chan_vpb.oo] Error 1
make: *** [channels] Error 2
Do not try to unlock a non-existant channel if the transfer fails.
(closes issue #13800)
Reported by: dwagner
Patches:
asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608)
........
................
Use autoconf logic to determine whether the system has timersub or not. Do not blindly assume Solaris does not.
(closes issue #13838)
Reported by: ano
........
................
Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module.
(closes issue #14079)
Reported by: elguero
........
Russell Bryant [Mon, 15 Dec 2008 16:26:09 +0000 (16:26 +0000)]
Merged revisions 164203 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r164203 | russell | 2008-12-15 08:40:24 -0600 (Mon, 15 Dec 2008) | 39 lines
Merged revisions 164201 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines
Handle a case where a call can be bridged to a channel that is still ringing.
The issue that was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once the parked
call got taken out of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing indication. This patch
fixes this case so that the caller receives ringback once it comes out of
parking until the other side answers.
The fixes are:
- Make sure we remember that a channel was an outgoing channel when doing
a masquerade. This prevents an erroneous ast_answer() call on the channel,
which causes a bogus 200 OK to be sent in the case of SIP.
- Add some additional comments to explain related parts of code.
- Update the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that are events
or commands should be ignored.
- When a bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still ringing.
- Rework ast_indicate_data() a bit for the sake of readability.
Russell Bryant [Mon, 15 Dec 2008 16:19:00 +0000 (16:19 +0000)]
Merged revisions 164272 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r164272 | russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines
When a reload is issued, always process the configuration for dundi.conf.
The reason is that a reload can be used to refresh DNS lookups for defined peers.
Even if the config file hasn't changed, we want to process it for that purpose.
* Fix reference counting used in the __queues_show function
* Add code to be sure that the "queue show" command does not
print information for a realtime queue which has been deleted
from the backend
* Add a missing unref to the realtime queue loading function for
the case where a queue is in the module's container but has been
deleted from the realtime backend
Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret.
(closes issue #14073)
Reported by: seandarcy
........
When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered.
Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a
pointer inside editline to look back to asterisk.c, so others don't spend
as much time as I did looking (in the wrong place) for the appropriate
function.
Reported by: ZX81, via the #asterisk-users channel
Fixed by: me (license 14)
........
................
Russell Bryant [Fri, 12 Dec 2008 18:52:14 +0000 (18:52 +0000)]
Merged revisions 163670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines
Rename a number of tcptls_session variables. There are no functional changes here.
The name "ser" was used in a lot of places. However, it is a relic from when
the struct was a server_instance, not a session_instance. It was renamed since
it represents both a server or client connection.
When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found.
(closes issue #13811)
Reported by: pj
........
Behaviour now is that general codec config flows to default_line and default_device. [devices] stuff amends default_device and similar for [lines]. These are copied to individual device and line as they are created.
Added confcapability and confprefs for the configured stuff which doesn't change as device and so on are connected. prefs are based on line prefs if they exist, else the device prefs are used (prefs identifies codec order).
(closes issue #13806)
Reported by: pj
Patches:
codecs.diff uploaded by wedhorn (license 30)
Tested by: pj and me
........
Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven.
(closes issue #13525)
Reported by: pj
........
Russell Bryant [Fri, 12 Dec 2008 14:05:01 +0000 (14:05 +0000)]
Merged revisions 163449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines
Merged revisions 163448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines
Resolve issues that could cause DTMF to be processed out of order.
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal
is messed up. By intercepting those events with a signal handler in the remote
console, we can avoid those issues.
(closes issue #13464)
Reported by: tzafrir
Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
........
................
projects / thirdparty / asterisk.git / log
