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

- Call waiting is enabled/disabled now per feature_set. Each line has the feature_set defined and each
  provider (pjsip endpoint) has line selected. From now on call waiting status can be defined in uci
  config and changed by feature code, as a result corresponding feature set or endpoint cw status will
  be changed
- Rename some functions and variables which had misleading names
- Add 5s beep timer indicating incoming call waiting
- Fix 20s timeout when there is already another call in progress
- Support call waiting/3 way call for DECT
- Implement "exceed call count" checking for line/extension/all

On receipt of EVENT_CALL_REJECT, hangup all ast_channel that are requested by the same incoming
call when the call is in RINGING or CALLWAITING state. Then the caller will be released and all
ring signal is stopped.

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

There are two ways to play tones, 1) by platform API; 2) by asterisk.
The ongoing tone will be stopped when a new tone is about to start
if both tones are played by platform API.
But if the current tone is played by platform API and the new tone is about
to be played by asterisk, the existing tone must be stopped explicitly.

Use audio files "activated" and "de-activated" for set/get status of
call waiting feature.

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

After removing 'from_user' config from  pjsip_endpoint config file we
need to use 'contact_user' which is translated to proper |USER| value,
otherwise default 'asterisk' user is used.

Hemlata authored 3 years ago and Sukru Senli committed 3 years ago

- UBUS object asterisk
- UBUS object voice.asterisk
- UCI /etc/config/asterisk

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

Check the asterisk sounds url availability. If it is not available
(sometimes asterisk server is down and we could not build our asterisk
fork) then use the iopsys mirror url instead to build the asterisk sounds.

Markus Gothe authored 3 years ago and Sukru Senli committed 3 years ago

When we originate a call from an extension to another we
get a NULL pointer issue when going on hook.

Enable CONFIG_PACKAGE_asterisk-res-clioriginate and then
use the command below:

asterisk -rx 'channel originate local/#123457@local_extensions
extension 0000@local_extensions'

After going on hook on the FXS port, which is mapped to extension
0000, there will be a segmentation fault because the other channel
doesn't implement the getRtpStats() function pointer.

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

This commit includes revert of:
commit d178f497
"res_pjsip:  Filter out non SIP(S) requests"

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

This solves the issue of Unknown state of a device.

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

After 'from_user' config has been removed from pjsip_endpoint config the
user in FROM header was wrong. Fix it with using session->id.number
instead of connected_id.number (which is wrong in this case).

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

Don't destroy the connection when the peer sub-channel is in dialing or ringback state.

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

There was a possibility that during attended call transfer, deadlock
happened in locking bridge_channel and channel.
bridge_channel_queue_deferred_frames() was waiting for brcm_indicate()
to unlock the channel, while brcm_indicate() which called
ast_bridge_channel_queue_frame() was waiting for
bridge_channel_queue_deferred_frames() to unlock the bridge.

Creating new macro to lock both channel and bridge_channel at the same
time and sched_yield() -> retry if failed and use it in
bridge_channel_queue_deferred_frames() avoiding the deadlock.