Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

By default all tellines are enabled, we can disable it by
uci set asterisk.tellineX.enabled=0
When the corresponding telline is disabled all events in
brcm_process_event() are ignored (offhook, onhook etc),
no dialtone played and no call proceeded. For incoming
INVITE there is no ringing on the telline, we respond
with "403 Forbidden" when the telline is disabled.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

MWI feature can be enabled/disabled by the 'mwi_enabled' configuration
parameter in asterisk config. When the NOTIFY message is received by
pjsip with new message the ubus event is sent in asterisk.mwi path.
Apart from that 'mwi_dialtone_state' can be configured for specific
endpoint (sip account) - it will be used as audiable indication when new
message is waiting for the mailbox connected with the endpoint.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

When the NOTIFY event was received by pjsip and the message body was not
properly terminated with '\r\n' the SEGFAULT happened in
parse_simple_message_summary().
Check the proper termination of message body for NOTIFY and fix it if
its wrong.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Adam Borowski authored 3 years ago and Sukru Senli committed 3 years ago

Updated howler audio files target path

Adam Borowski authored 3 years ago and Sukru Senli committed 3 years ago

Added howler tones audio files and Makefile install update
to copy *.cln files into target's /usr/share/asterisk/sounds dir

Adam Borowski authored 3 years ago and Sukru Senli committed 3 years ago

Play howler tone in case user forgot to put handset on hook
as a reminder to end the call properly.

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

Move creating of ubus thread after getting the endpt_id and
do not call ubus_lookup_id() afterwards since it can cause deadlock.
Ubus thread uses the same ctx (ubus context *) as the chan_brcm thread
while UBUS is not thread safe and ctx needs to be used by only one
thread in the same time.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 3 years ago and Sukru Senli committed 3 years ago

There was a situation that during asterisk restart -> chan_brcm.so
module loading, endptmngr was not up and running yet, causing the ubus
lookup for endpt_id failed. It caused chan_brcm.so module not loaded
and asterisk not fully working properly. Add waiting for endptmngr fixes
the issue.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Adam Borowski authored 4 years ago and Sukru Senli committed 3 years ago

Rejecting call ewhen do not disturbed
by indicating line as busy instead of user call rejection

Signed-off-by: Adam Borowski <adam.borowski@sigma.se>

Adam Borowski authored 4 years ago and Sukru Senli committed 3 years ago

Rejecting call ewhen do not disturb is enabled

Signed-off-by: Adam Borowski <adam.borowski@sigma.se>

Adam Borowski authored 4 years ago and Sukru Senli committed 3 years ago

http://downloads.asterisk.org

 being down

if downloading from asterisk fails second try is made
to get the files from fancom mirror

Signed-off-by: Adam Borowski <adam.borowski@sigma.se>

Adam Borowski authored 4 years ago and Sukru Senli committed 3 years ago

http://downloads.asterisk.org

 being down

if downloading from asterisk fails second try is made
to get the files from fancom mirror

Signed-off-by: Adam Borowski <adam.borowski@sigma.se>

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Sign '#' was recognized as the end of feature code hence some of the
codes for getting status were not working, like:
*#43# - check status of call waiting
*#227# - check status of anonymous call rejection

The fix makes '#' not recognized as the end of feature code if the
previous character is '*'

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Since we modified asterisk default implementation of attended call
transfer there was an issue with channels stuck (not destroyed)
afterwards, It was caused due to Ref count for channels used in call
transfer was greater than 2.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Yalu Zhang authored 4 years ago and Sukru Senli committed 3 years ago

(cherry picked from commit 629000dc on branch iopsys-16.3)

Yalu Zhang authored 4 years ago and Sukru Senli committed 3 years ago

Some long verbose or debug log messages like contents of INVITE are truncated in syslog,
/var/log/asterisk/messages and console.

(cherry picked from commit 4764ca4a on branch iopsys-16.3)

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Yalu Zhang authored 4 years ago and Sukru Senli committed 3 years ago

Call ubus_lookup_id directly instead of passively being updated by
the callback ubus_event_new_obj() which doesn't work reliably during
the system startup.

Yalu Zhang authored 4 years ago and Sukru Senli committed 3 years ago

(cherry picked from commit 8e6bae6e)

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Change implementation of adding mediasec headers in REGISTER, INVITE,
REINVITE, OPTIONS events to be dynamically configured based on the
response from the Sip Server.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

In pjsip the attended call transfer has an issue that connection between
transferee and transferor is not finished immediately but after 60s
timeout. Not sure why it is implemented this way (defer termination) but
changing the timeout into 1s makes the attended call transfer work the
same as it was with chan_sip driver.

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

Yalu Zhang authored 5 years ago and Sukru Senli committed 3 years ago

It is caused that PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP is not defined on Broadcom ARM platform.
But recursive mutex is mandatory for Asterisk to work properly. Otherwise there will be dead lock
in some modules, e.g. loader.c.

Grzegorz Sluja authored 4 years ago and Sukru Senli committed 3 years ago

Signed-off-by: Grzegorz Sluja <grzegorz.sluja@iopsys.eu>

If an SSL socket parent/listener was destroyed during the handshake,
depending on timing, it was possible for the handling callback to
attempt access of it after the fact thus causing a crash.

ASTERISK-29415 #close

Change-Id: I105dacdcd130ea7fdd4cf2010ccf35b5eaf1432d

If chan_iax2 received a packet with an unsupported media format, for
example vp9, then it would set the frame's format to NULL. This could
then result in a crash later when an attempt was made to access the
format.

This patch makes it so chan_iax2 now ignores/drops frames received
with unsupported media format types.

ASTERISK-29392 #close

Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1

If a re-INVITE is received after we have sent a BYE request then it
is possible for no channel to be present on the session. If this
occurs we allow PJSIP to produce the offer instead. Since the call
is being hung up if it produces an incorrect offer it doesn't
actually matter. This also ensures that code which produces SDP
does not need to handle if a channel is not present.

ASTERISK-29381

Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042