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

- 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.

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

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

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

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.

SIPSessionID is from Session-ID header field either in INVITE for incoming calls or in
200 OK for outgoing calls.

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

When the internal call "0000" is proceeded from any of DECT handset, busy tone is heared since we always use
extension_id=0 for the first outgoing call and the call is directed to the same extension_id = 0. In this
case CALL_REJECT is received by asterisk, but connection is not closed due to another channel_state value.
Fix it so that connection is properly closed in this case.

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.