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.

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

For the case if there is more than one contact header in 200 OK to REGISTER and the last Contact has
expires=0, asterisk wrongly set next registration period to default value DEFAULTEXPIRY.

The correct behaviour is to pick the expire value in the Contact which matches the Contact in the
original REGISTER.

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

- 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

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

"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.

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.