A call retrieved from hold may wind up with no audio.
If the retrieved call is natively bridged then the call may not have any
audio path. The following warning message is given:
"Failed to add <dfd> to conference <chan>/<chan>: Invalid argument".
* Open the media on a B channel when pri_fixup_principle() moves the call
from a no_b_channel channel to a real channel.
* Added lock protection while pri_fixup_principle() moves a call from one
private structure to another.
* Made some pri_fixup_principle() messages more meaningful.
..........
This only skips authentication on retransmissions that are already
authenticated. A similar method is already used for INVITES. This
is the kind of thing we end up having to do when we don't have a
transaction layer...
Leif Madsen [Mon, 3 Jan 2011 13:14:25 +0000 (13:14 +0000)]
Increase side of mapping response field.
I've increased the size of the response field in a DUNDi mapping because of
some documentation I'm writing. Previously it was set to AST_MAX_EXTENSION which
is only 80 characters, which is far too small when you're using some dialplan
functions to craft a response. The example I'm using is:
Alexandr Anikin [Sat, 25 Dec 2010 16:12:28 +0000 (16:12 +0000)]
Change order of sending TCS and MSD packets
Change order of sending Terminal Capability Set and MasterSlave
Determination packets, MSD send when TCS exchange procedure is done
(we send tcs ack to remote and we have remote tcs ack already
or we receive tcs ack from remote and we have send our tcs ack to
remote already). Some endpoints can work in this sequence only,
i suggest they can't work with both (tcs and msd) exchange procedures
simultaneously.
Also changed StartH245 facility message sending. It send on
incoming calls only due to some endpoints can't proccess properly
this facility messages on their incoming calls.
Richard Mudgett [Wed, 22 Dec 2010 02:10:39 +0000 (02:10 +0000)]
Chan_dahdi sends an empty COLP on the bridged channel.
Chan_dahdi always inserts a connected party IE when you call from one
dahdi channel to another dahdi channel, even if no such information was
received on the 2nd channel. This clears the display of many phones.
* Removed leftover artifact from before the valid flag was added.
* Updated all of the channel's caller id information with the new
connected line information instead of just the string parts.
Remove changes to via processing that were not supposed to go into the last commit.
........
r299220 | mnicholson | 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines
Let Asterisk find better backtrace information with libbfd.
The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search
for better symbol information within both the Asterisk binary, as well as
loaded modules, to assist when using inline backtraces to track down problems.
........
................
Fix improper hangup when doing an attended transfer to queue.
Had to indicate ringing in wait_for_answer so the attended transfer code would
not try and hang up the local channel it created, which would kill the call.
Outgoing PRI/BRI calls cannot do DTMF triggered transfers.
Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING
message is not received. The debug output shows that the DTMF begin event
is seen, but the DTMF end event is missing. When the DTMF begin happens,
the call is muted so we now have one way audio (until a DTMF end event is
somehow seen).
* Made set the proceeding flag when the PRI_EVENT_ANSWER event is
received.
* Made absorb the DTMF begin and DTMF end events if we are overlap dialing
and have not seen a PROCEEDING message.
* Added a debug message when absorbing a DTMF event.
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.
Display the capabilities requested when requesting a fax session fails instead of displaying a hex value.
Tweak the way fax stats are calculated so that all fax attempts and faliures are logged. Also make ensure faxes are either counted as completed or falied and never both.
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.
Don't create a Local channel if the target extension does not exist.
(closes issue #18126)
Reported by: junky
Patches:
followme.diff uploaded by junky (license 177)
(partially restructured by me to avoid a possible memory leak)
........
................
Improve handling of REGISTER requests with multiple contact headers.
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.
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.
Russell Bryant [Thu, 2 Dec 2010 13:20:19 +0000 (13:20 +0000)]
Merged revisions 297229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines
Merged revisions 297228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines
Add "DAHDI" to a couple of app_meetme error messages.
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.
........
................
Fix not stopping MOH when transfered local channel queue member is answered.
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.
Russell Bryant [Mon, 29 Nov 2010 21:26:44 +0000 (21:26 +0000)]
Complete some error handling in transmit_publish() in chan_sip.c.
This error handling block caught my eye. It was missing a couple of things,
but it should be safe now. Thanks to mmichelson for the quick peer review
on IRC.
Invalid mISDN PTMP redirecting signaling as TE towards NT.
The mISDN PTMP redirection signaling (NOTIFY redirecting number and
notification code, SETUP redirecting number) is also sent in PTMP/TE mode.
It should only apply in PTMP/NT mode. The call setup proceeds but the
network (Deutsche Telekom) reacts with ugly ISDN STATUS messages.
Also don't send the redirecting number ie when PTP is also sending the
DivertingLegInformation2 facility. The redirecting number ie is redundant
and the network (Deutsche Telekom) complains about it.
Patches:
abe_2651_v4.patch uploaded by rmudgett (license 664)
Brad Watkins [Fri, 26 Nov 2010 18:19:02 +0000 (18:19 +0000)]
Fix reloading of peer when a user is requested.
Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. This had the effect of sending one NOTIFY for every time a sip peer made a call, in one case eventually overwhelming the phone and causing it to reboot.
Russell Bryant [Wed, 24 Nov 2010 23:29:44 +0000 (23:29 +0000)]
Merged revisions 296221 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r296221 | russell | 2010-11-24 17:28:19 -0600 (Wed, 24 Nov 2010) | 13 lines
Merged revisions 296213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) | 6 lines
Make Asterisk less crashy.
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.
........
................
Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
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.
* 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.
........
................
Russell Bryant [Wed, 24 Nov 2010 20:23:46 +0000 (20:23 +0000)]
Merged revisions 296083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r296083 | russell | 2010-11-24 14:23:11 -0600 (Wed, 24 Nov 2010) | 19 lines
Merged revisions 296082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) | 12 lines
Fix false reporting of an error by set_format().
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. :-)
........
................
Russell Bryant [Wed, 24 Nov 2010 17:13:08 +0000 (17:13 +0000)]
Merged revisions 296001 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r296001 | russell | 2010-11-24 11:03:16 -0600 (Wed, 24 Nov 2010) | 45 lines
Merged revisions 296000 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) | 38 lines
Handle failures building translation paths more effectively.
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.
The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
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.
Richard Mudgett [Sat, 20 Nov 2010 03:11:15 +0000 (03:11 +0000)]
One way audio before answering call waiting call on analog port.
* Analog call waiting Caller ID spills could get stuck resulting in one
way audio until the waiting call is answered. This only happens on the
second (and later) call waiting call if the active call is not the first
call.
* The CLI/AMI "dahdi show channel" command could report the wrong channel
information.
Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer
in sync.
Russell Bryant [Sat, 20 Nov 2010 00:50:00 +0000 (00:50 +0000)]
Merged revisions 295710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines
Fix cache of device state changes for multiple servers.
This patch addresses a regression where device states across multiple servers
were not being processing completely correctly. The code works to determine
the overall state by looking at the last known state of a device on each
server. However, there was a regression due to some invasive rewrites of how
the cache works that led to the cache only storing the last device state change
for a device, regardless of which server it was on.
The code is set up to cache device state change events by ensuring that each
event in the cache has a unique device name + entity ID (server ID). The code
that was responsible for comparing raw information elements (which EID is)
always returned a match due to a memcmp() with a length of 0.
There isn't much code to fix the actual bug. This patch also introduces a new
CLI command that was very useful for debugging this problem. The command
allows you to dump the contents of the event cache.
(closes issue #18284)
Reported by: klaus3000
Patches:
issue18284.rev1.txt uploaded by russell (license 2)
Tested by: russell, klaus3000
This is not a perfect solution as headers that are joined via commas are not
detected. This is a parsing issue that to fix "correctly" would necessitate
a new SIP parser.
Richard Mudgett [Fri, 19 Nov 2010 16:47:11 +0000 (16:47 +0000)]
Bring sig_analog extraction more into alignment with orig-trunk/v1.6.2 chan_dahdi.
* Restore SMDI support.
* Fixed initial value of struct analog_pvt.use_callerid. It may get
forced on depending upon other config options.
* Call analog_dnd() instead of manual inlined code.
* Removed unused struct analog_pvt.usedistinctiveringdetection.
* Removed the struct analog_pvt.unknown_alarm flag. It was really the
struct analog_pvt.inalarm flag.
* Use ast_debug() instead of ast_log(LOG_DEBUG).
* Rename several function's index variable to idx.
* Some formatting tweaks.
projects / thirdparty / asterisk.git / log
