If a REGISTER request with a Call-ID matching an existing transaction is received
it was possible that the REGISTER request would overwrite the initreq of the
private structure. This info is used to generate messages for other responses in
the transaction. This patch ignores REGISTER requests that match non-REGISTER
transactions.
(closes issue #18051)
Reported by: eeman
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/1050/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297959 f38db490-d61c-443f-a65b-d21fe96a405b
Some previous behavior was attempted to be restored, but mistakingly I did
not realize that the previous behavior was incorrect. This fixes DTMF not
being detected since DTMF shouldn't cause the SSRC to change.
(related to issue #17404)
(closes issue #18189)
(closes issue #18352)
Reported by: marcbou
Tested by: cmbaker82
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297823 f38db490-d61c-443f-a65b-d21fe96a405b
(closes issue #18126)
Reported by: junky
Patches:
followme.diff uploaded by junky (license 177)
(partially restructured by me to avoid a possible memory leak)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297689 f38db490-d61c-443f-a65b-d21fe96a405b
The changes here attempt to more strictly follow RFC 3261 section 10.3.
Basically the following will now cause a 400 Bad Response to be returned, if:
- multiple Contact headers are present with one set to expire all bindings ("*")
- wildcard parameter is specified for Contact without Expires header or Expires
header is not set to zero.
ABE-2442
ABE-2443
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297603 f38db490-d61c-443f-a65b-d21fe96a405b
When the adaptive jitter buffer is enabled in sip.conf, the first frame placed
in the jitter buffer fails with something like:
jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466,
threshold 1000, new offset 215886466
This happens because the offset is not initialized before calling jb_put(). This
patch modifies jb_put_first_adaptive() to set the offset to the frame's
timestamp.
Review: https://reviewboard.asterisk.org/r/1041/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297310 f38db490-d61c-443f-a65b-d21fe96a405b
This is in response to some questions on IRC. To the user, there was nothing
that made it obvious that this error had anything to do with DAHDI not being
loaded.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297228 f38db490-d61c-443f-a65b-d21fe96a405b
The problem here is only present when local channels are used with the MOH
passthru option as well as no optimization (/nm). I will describe the slightly
bizarre scenario that was used to test, where phones B and C are queue members:
Phone A dials into a queue with two members using local channels and the above
options. Phone B answers. Phone A blind transfers phone B into the same queue.
Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH.
In this scenario, the unhold frame that should have gotten to phone B never
arrived due to the masquerade from the blind transfer. This is usually fine
since app_queue manages the starting and stopping of MOH. However, with the
passthrough option enabled when app_queue attempts to stop MOH it tries to do
so on the local channel rather than the real channel. The easiest solution
was to just make sure to send an unhold frame during the transfer since it
wouldn't make sense to have MOH playing after a transfer anyway. This only
modifies SIP transfers, but the other transfers did not seem to be a problem.
If DTMF based transfers were a problem it might be okay to add ast_moh_stop
to finishup, but I didn't want to have to add that unless required.
ABE-2624
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297072 f38db490-d61c-443f-a65b-d21fe96a405b
This mainly deals with the problem of constructors on platforms where an
explicit constructor order cannot be specified (any system with gcc 4.2
or less). However, this is only a problem on those systems where we need
to initialize mutexes with a constructor, because we have other code that
also relies upon constructors, and we cannot specify that mutexes are
initialized first (and destroyed last).
There are two approaches to dealing with this issue, related to whether the
code exists in the core Asterisk binary or in a separate code module. In the
core case, constructors are run immediately upon load, and the file_versions
list mutex needs to be already initialized, as it is referenced in the first
constructor within each core source file. In this case, we use pthread_once
to ensure that the mutex is initialized immediately before it is used for the
first time. The only caveat is that the mutex is not ever destroyed, but
because this is the core, it makes no real difference; the only time when
destruction is safe would be just prior to process destruction, which takes
care of that anyway. And due to using pthread_once, the mutex will never be
reinitialized, which means only one structure has leaked at the end of the
process. Hence, it is not a problematic leak.
The second approach is to use the load_module and unload_module routines,
which, for obvious reasons, exist only in loadable modules. In this second
case, we don't have a problem with the constructors, but only with destructor
order, because mutexes can be destroyed before their final usage is employed.
However, we need the mutexes to still be destroyed, in certain scenarios: if
the module is unloaded prior to the process ending, it should be clean, with
no allocations by the module hanging around after that point in time.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296867 f38db490-d61c-443f-a65b-d21fe96a405b
Since we might not put a new translation path on the channel, go ahead and
set it to NULL right after destroying the old one to ensure we don't try
to free an invalid translation path later on.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296213 f38db490-d61c-443f-a65b-d21fe96a405b
The FXS connected phone has to have CW/CID support to fail, as it will
send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal
phone with no CID never fails. Also the SIP phone does not hear MOH when
the CW call is answered.
The DTMF end frame is suppressed when the phone acknowledges the CW signal
for CID. The problem is the DTMF begin frame needs to be suppressed as
well. The DTMF begin frame is causing SIP to start sending the DTMF RTP
frames. Since the DTMF end frame is suppressed, SIP will not stop sending
those DTMF RTP packets.
* Suppress the DTMF begin and end frames when the channel driver is
looking for DTMF digits.
* Fixed a couple issues caused by not cleaning up the CID spill if you
answer the CW call while it is sending the CID spill.
* Fixed not sending CW/CID spill to the phone when the call is natively
bridged. (Fixed by not using native bridge if CW/CID is possible.)
* Suppress received audio when sending CW/CID spills. The other parties
involved do not need to hear the CW/CID spills and may be confused if the
CW call is for them.
(closes issue #18129)
Reported by: alecdavis
Patches:
issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
NOTE:
* v1.4 does not have the main problem fixed by suppressing the DTMF start
frames. The other three items fixed are relevant.
* If you really must restore native bridging between analog ports, you
need to disable CW/CID either by configuring chan_dahdi.conf
callwaitingcallerid=no or dialing *70 before dialing the number to
temporarily disable CW.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296165 f38db490-d61c-443f-a65b-d21fe96a405b
In the case that the native format was able to be changed to match the
new requested format, the code proceeded to attempt to build a translation
path, anyway. The result would be NULL, since no translation path is
necessary and resulted in this function thinking an error has occurred.
This case is now specifically caught and no attempt to build a translation
path is attempted.
Thanks to our automated tests and bamboo.asterisk.org for catching this problem
and making a whole lot of noise when things started failing. :-)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296082 f38db490-d61c-443f-a65b-d21fe96a405b
The problem scenario occurred on a heavily loaded system that was using the
codec_dahdi module and exceeded the hardware transcoding capacity. The failure
mode at that point was not good. The report came in to us as an Asterisk
lock-up. The "core show locks" shows a ton of threads locked up (but no
obvious deadlock). Upon deeper investigation, when the system is in this
state, the CPU was maxed out. The CPU was being consumed by the Asterisk
logger spewing messages on every audio frame for calls set up after transcoder
capacity was reached.
The purpose of this patch is to make Asterisk handle failures to create a
translation path in a more graceful manner. If we can't translate, then the
call just needs to be dropped, as it's not going to work. These are the
changes:
1) In set_format() of channel.c (which is called by set_read_format() and
set_write_format()), it was ignoring if ast_translator_build_path() failed and
returned NULL. It now pays attention to that case and returns a result
reflecting failure. With this change in place, the bridging code will
immediately detect a failure and end the bridge instead of proceeding to try to
bridge frames that can't be translated and making channel drivers freak out by
sending them frames in a format they weren't expecting.
2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was
ignored. It is now reflected in the return value of the function. This didn't
turn out to have any affect on the bug, but seemed like a good change to leave
in.
3) In app_dial(), when only sending a call to a single endpoint, it will
attempt to do some bridging of its own of early audio. It uses
make_compatible() when it's going to do this. However, it ignored failure from
make compatible. So, even with the fix from #1, if there was early audio going
through app_dial, there would still be a period of invalid frames passing
through. After detecting failure here, Dial() exits.
ABE-2658
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296000 f38db490-d61c-443f-a65b-d21fe96a405b
To recreate the problem:
1) Party A calls Party B
2) Invoke CLI "channel redirect" command to redirect channel call leg
associated with A.
3) All associated channels are hung up.
Note that if the CLI command were done on the channel call leg associated
with B it works.
This regression was a result of the fix for issue #16946
(https://reviewboard.asterisk.org/r/740/).
The regression affects all features that use an async goto to execute the
dialplan because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection.
The struct ast_channel._softhangup code is a mess. The variable is used
for several purposes that do not necessarily result in the call being hung
up. I have added doxygen comments to describe how the various _softhangup
bits are used. I have corrected all the places where the variable was
tested in a non-bit oriented manner.
The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak
hangup request so the soft hangup requests that do not normally result in
a hangup do not hangup.
JIRA SWP-2470
JIRA SWP-2489
(closes issue #18171)
Reported by: SantaFox
(closes issue #18185)
Reported by: kwemheuer
(closes issue #18211)
Reported by: zahir_koradia
(closes issue #18230)
Reported by: vmarrone
(closes issue #18299)
Reported by: mbrevda
(closes issue #18322)
Reported by: nerbos
Review: https://reviewboard.asterisk.org/r/1013/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295790 f38db490-d61c-443f-a65b-d21fe96a405b
Add the ability to play a sound file, listen for more than just one digit, specify
escape characters. Backwards compatible (to work with only timeout specified).
(closes issue #15531)
Reported by: diLLec
Patches:
asterisk-res_agi-203638-patched.patch uploaded by diLLec (license 839)
Tested by: diLLec, espiceland
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295552 f38db490-d61c-443f-a65b-d21fe96a405b
It seems the fix to issue 17103 was a little overzealous and removed the code
that backed up the textfile containing the original message duration. This
code has now been restored.
(related to issue #17103)
ABE-2654
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295200 f38db490-d61c-443f-a65b-d21fe96a405b
lmadsen