Wenpeng Song authored 1 year ago and Grzegorz Sluja committed 10 months ago

Crash due to a race condition between this ref correction
https://dev.iopsys.eu/voice/asterisk/-/blob/devel/channels/chan_pjsip.c#L2088-2095
and the unref under chan_voicemngr_attended_call_transfer
https://dev.iopsys.eu/voice/asterisk-chan-voicemngr/-/blob/devel/src/channels/chan_voicemngr.c?ref_type=heads#L5646-5651.
When the correction happened and finished before the unref during the transfer, which leading
the result of ref=1 after the unref, then the crash happened during hangup the channel, add an
extra correction for ref<2 could resolve this issue.

Yalu Zhang authored 1 year ago and Grzegorz Sluja committed 10 months ago

Asterisk can crash if two FXS ports on the same device act as transferor and
transfer target when asterisk.tel_options.hold_target_before_refer='yes',
which is the default config. The issues happens after a couple of calls.

The solution is to revert d38e42d8 about
reference count decreasing.
Some validation check is also added to chan_pjsip_hangup().

TODO: After this commit, there will be issues if
asterisk.tel_options.hold_target_before_refer='no'.
- Remaining pjsip channels after a couple of attended transfers in the above
  scenario
- Transferor might not be able to call transfer target because the maximum
  sessions per line exceed the specified number
These issues will be solved in a new commit.

This reverts commit 3c8116c8. It has
been a merge that has been acidentally squashed into one commit with the
consequence that all history has been lost. The next commit will be a
proper merge commit rectifying this.

Grzegorz Sluja authored 1 year ago and Yalu Zhang committed 1 year ago

Our commits to the previous version have been rebased.

This fixes a segment fault caused by ast_channel_name(NULL) for internal CBBS.

Currently, both pulse and tone dialing are always enabled
on all FXS lines, with no way of disabling one or the other.

In some circumstances, it is desirable or necessary to
disable one of these, and this behavior can be problematic.

A new "dialmode" option is added which allows setting the
methods to support on a per channel basis for FXS (FXO
signalled lines). The four options are "both", "pulse",
"dtmf"/"tone", and "none".

Additionally, integration with the CHANNEL function is
added so that this setting can be updated for a channel
during a call.

Resolves: #35
ASTERISK-29992

UserNote: A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.

Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.

(cherry picked from commit 82d7bb49ddc15cff5ea2e50f5ee6934102fe13d1)

Wenpeng Song authored 1 year ago and Yalu Zhang committed 1 year ago

Add the emergency headers if a new session is an emergency call. An emergency flag was set
by chan_voicemngr.

ASTERISK_29392 (a security fix) introduced a regression by
not processing frames when we don't have an audio format.

Currently, chan_iax2 only calls jb_get to read frames from
the jitterbuffer when the voiceformat has been set on the pvt.
However, this only happens when we receive a voice frame, which
means that prior to receiving voice frames, other types of frames
get stalled completely in the jitterbuffer.

To fix this, we now fallback to using the format negotiated during
call setup until we've actually received a voice frame with a format.
This ensures we're always able to read from the jitterbuffer.

ASTERISK-30354 #close
ASTERISK-30162 #close

Change-Id: Ie4fd1e8e088a145ad89e0427c2100a530e964fe9

Wenpeng Song authored 2 years ago and Yalu Zhang committed 2 years ago

A separate package asterisk-chan-voicemngr has been added to replace chan_brcm.

cherry picked from commit 32337872

- EARLY_OFFHOOK
- EARLY_ONHOOK
- MEDIA

Wenpeng Song authored 2 years ago and Yalu Zhang committed 2 years ago

The commit c6a0ad61 changes sub->parent->context during an outgoing
call that causes future outgoing calls fail.

Wenpeng Song authored 2 years ago and Yalu Zhang committed 2 years ago

In function brcm_start_calling(), the type of sub->parent is struct brcm_pvt*. But it was wrongly
casted to struct ast_channel* so the contents of sub->parent were messed up. 

Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).

ASTERISK-30319 #close

Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645

-----------------

This commit reinstates MES with some casting fixes to the
functions in time.h that convert between doubles and timeval
structures.  The casting issues were causing incorrect
timestamps to be calculated which caused transcoding from/to
G722 to produce bad or no audio.

ASTERISK-30391

-----------------

This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038

This reverts commit d454801c.

Reason for revert: Issue when transcoding to/from g722

Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86

This module has been updated to provide additional
quality statistics in the form of an Asterisk
Media Experience Score.  The score is avilable using
the same mechanisms you'd use to retrieve jitter, loss,
and rtt statistics.  For more information about the
score and how to retrieve it, see
https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score

* Updated chan_pjsip to set quality channel variables when a
  call ends.
* Updated channels/pjsip/dialplan_functions.c to add the ability
  to retrieve the MES along with the existing rtcp stats when
  using the CHANNEL dialplan function.
* Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
  checks for debugging purposes.
* Added several function to time.h for manipulating time-in-samples
  and times represented as double seconds.
* Updated rtp_engine.c to pass through the MES when stats are
  requested.  Also debug output that dumps the stats when an
  rtp instance is destroyed.
* Updated res_rtp_asterisk.c to implement the calculation of the
  MES.  In the process, also had to update the calculation of
  jitter.  Many debugging statements were also changed to be
  more informative.
* Added a unit test for internal testing.  The test should not be
  run during normal operation and is disabled by default.

ASTERISK-30280

Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a

Hemlata authored 2 years ago and Yalu Zhang committed 2 years ago

Before this commit, a normal dial tone is played if mwi_dialtone_state in "pjsip show endpoint" is
not "off", "congestion" or "special".
After this commit, in the above condition, a stutter dial tone is played if mwi_enable is 1, otherwise
a normal dial tone is played.

Hemlata authored 2 years ago and Yalu Zhang committed 2 years ago

- Include the header in requests and responses as per the RFC
- Play early media if the configuration allows

Grzegorz Sluja authored 2 years ago and Yalu Zhang committed 2 years ago

An incoming SIP request will be silently dropped if the following conditions are all met:
- The request is about to create a new session
- The source IP address of the request is not the address of the configured proxy

Note
- The source port of the request is not checked since we have not found the good way to
  retrieve the port that is used for transmission
- If the configured proxy is a domain name other than IP address, the resolved IP address
  might be different from the one which is being used in existing sessions

ASTERISK_28702 previously attempted to fix an
issue with flash hook hold timing out after
just under 17 minutes, when it should have never
been timing out. It fixed this by changing 999999
to INT_MAX, but it did so in chan_dahdi, which
is the wrong place since ss_thread is now in
sig_analog and the one in chan_dahdi is mostly
dead code.

This fixes this by porting the fix to sig_analog.

ASTERISK-30336 #close

Change-Id: I05eb69cc0b5319d357842a70bd26ef64d145cb15

Currently, chan_dahdi will wait for at least one
ring before an incoming call can enter the dialplan.
This is generally necessary in order to receive
the Caller ID spill and/or distinctive ringing
detection.

However, if neither of these is required, then there
is nothing gained by waiting for one ring and this
unnecessarily delays call setup. Users can now
use immediate=yes to make FXO channels (FXS signaled)
begin processing dialplan as soon as Asterisk receives
the call.

ASTERISK-30305 #close

Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5

Grzegorz Sluja authored 2 years ago and Yalu Zhang committed 2 years ago

In brcm_request() there were posibilities that we called ast_channel_name(tmp)
if tmp is NULL, it leads to crash. Also fixed other deadlock possibilities where
mutex would not be unlocked in some cases.

- R0. If more than one account is registered, and there has another incoming call(ringing) to other accounts,
  it will be rejected as well when the ongoing one doing R0. (A, B registered on DUT; A<=>C(ongoing call);
  D=>B(ringing); E=>A(ringing,cw); A press R0 then both D and E be rejected)
- 3-way conference back to 2-way call, then performing R1 will lead to a crash

"uk"   - the default config which support flash-hook (R) only triggering call waiting and 3-way conference.
         R4 and R5 are for attended and unattended call transfer respectively with a timer.
"etsi" - Using R0, R1, R2, to trigger different ways of handling call waiting. R3 for 3-way conference.
         R4 and R5 are for attended and unattended call transfer respectively without a timer.

unicast_rtp_request() was setting the channel variables like this:

pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
    ast_sockaddr_stringify_addr(&local_address));
ast_rtp_instance_get_local_address(instance, &local_address);
pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
    ast_sockaddr_stringify_port(&local_address));

...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being
set before local_address was set.  In fact, the address part of
local_address was set earlier in the function, just not the port.
This was confusing however so ast_rtp_instance_get_local_address()
is now being called before setting UNICASTRTP_LOCAL_ADDRESS.

ASTERISK-30281

Change-Id: I872ac49477100f4eb33891d46efc6ca21ec81aa4

This fixes dahdi_request to properly set the cause
code to CONGESTION instead of BUSY if no channels
were actually available.

Currently, the cause is erroneously set to busy
if the channel itself is found, regardless of its
current state. However, if the channel is not available
(e.g. T1 down, card not operable, etc.), then the
channel itself may not be in a functional state,
in which case CHANUNAVAIL is the correct cause to use.

This adds a simple check to ensure that busy tone
is only returned if a channel is encountered that
has an owner, since that is the only possible way
that a channel could actually be busy.

ASTERISK-30274 #close

Change-Id: Iad5870223c081240c925b19df8d6af136953b994

- The issue about TELCHAN stuck after call hold/unhold
- Asterisk crashes when call transfer is provided in an invalid scenario: Transferor calls to Transferee,
  then Transferor calls to Transfer Target, i.e. trying to let the caller do the transfer.

Fixes a format truncation warning in notify_message.

ASTERISK-30256 #close

Change-Id: I983a423c0214641ca4f8c9dfe0b19c47448fdee1

The service is triggered by R5, i.e. flash hook + 5.

Sequence numbers received in RTP packet need to be forwarded to brcm endpoint since based on this parameter
some of RTP statistics are calculated. It was wrong to use the locally generated sequence numbers.