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(cherry picked from commit e2599426)

-------------------------------------------------------------------------------
* b2268e0ff1 Add config parameter "allow_tx_hash_in_uri" in pjsip
* 823b936390 Update uci.asterisk.md to add uci option asterisk.tel_options.vmloglevel
-------------------------------------------------------------------------------
commit b2268e0ff18c965497cbc213c9408324e3ab0fe8
Author: George Yang <george.yang@iopsys.eu>
Date: 2024-04-16 14:17:49 +0000

    Add config parameter "allow_tx_hash_in_uri" in pjsip

 docs/api/uci.asterisk.md      | 3 +--
 res/res_pjsip/config_global.c | 7 +++++++
 schemas/uci/asterisk.json     | 7 +++++++
 3 files changed, 15 insertions(+), 2 deletions(-)
-------------------------------------------------------------------------------
commit 823b9363908f5fc1d647416b0c00ec5769b499a0
Author: George Yang <george.yang@iopsys.eu>
Date: 2024-04-10 16:38:31 +0000

    Update uci.asterisk.md to add uci option asterisk.tel_options.vmloglevel

 docs/api/uci.asterisk.md | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)
-------------------------------------------------------------------------------

George Yang (IOPSYS) authored 11 months ago and Yalu Zhang committed 11 months ago

This allows '#' unescaped in URI both within requested line and within headers in transmitted
SIP messages.

-------------------------------------------------------------------------------
* 0600aec Remove the default terminationdigit '#'
* bd521df Support K-break signal during call clearing procedure
-------------------------------------------------------------------------------
commit 0600aec8acc0d0dbc5d88929b3538f323cac3274
Author: George Yang <george.yang@iopsys.eu>
Date: 2024-03-21 15:50:57 +0000

    Remove the default terminationdigit '#'

 src/channels/chan_voicemngr.c | 12 +++++-------
 1 file changed, 5 insertions(+), 7 deletions(-)
-------------------------------------------------------------------------------
commit bd521df5d2150e3adc599e9126a7e98f447a1bd6
Author: Yalu Zhang <yalu.zhang@iopsys.eu>
Date: 2024-03-14 12:14:27 +0100

    Support K-break signal during call clearing procedure

 src/channels/chan_voicemngr.c | 50 ++++++++++++++++++++++++++++++++++++++------------
 src/channels/chan_voicemngr.h |  2 ++
 2 files changed, 40 insertions(+), 12 deletions(-)
-------------------------------------------------------------------------------

Reidar Cederqvist authored 1 year ago and Yalu Zhang committed 1 year ago

This is to avoid cases where the uci option is directly pushed into the asterisk config without
checking the value, otherwise the value can contain a newline followed by a malicious asterisk config option+value.
Note that '\n' is allowed in the value of an option or list.

Workaround for reference count correction 

Channel reference count correction for transfer

Configuration for retries of SIP Registration failover

Fix an issue that codec is not synchronised when switching between two calls

In case multiple server addresses exist, once client received
SIP regisration response 408/503/500 from one address, it will
then move to next address. If all addresses are tried and failed,
the client will schedule a new retry with certain interval.

Following options are added to UCI asterisk sip_service_provider
section for configuring SIP failover.

   - asterisk.sip0.failover_max_retries
     Max number of failover retries.
     By default, '-1', which means unlimited.

   - asterisk.sip0.failover_interval1
     The failover begin with the retry interval.
     By default, 10 seconds.

   - asterisk.sip0.failover_interval2
     After every 3 continuous failed retries (one RETRY trys all server addresses)
     the interval is used to schedule the next retry.
     By default, 60 seconds.

Lwithwma Brahma authored 2 years ago and George Yang committed 1 year ago

The outbound_proxy was added as per the asterisk doc.
https://wiki.asterisk.org/wiki/display/AST/PJSIP+with+Proxies

Change-Id: Id942ae1bae9f5e8242b7381d6e1cfd77601d3229

When applying asterisk server URI, asterisk.sip_service_provider.host
should take precedence over asterisk.sip_service_provider.domain.

TR104 parameters ...SIP.Network.RegistrarServer and .ProxyServer
are all mapped to UCI asterisk.sip_service_provider.host.

Change-Id: I685480308740b9ced922274f6837d0471bdf574c

Fix an un-hold issue during the failed 2nd outgoing call

Fix a bug that unexpected ringing is heard when transferor finishes transfer and hangs up

Fix a bug that RTP/RTCP log messages are shown even if RTP/RTCP debug is disabled

Fix a dead lock and remove some duplicated indications during un-hold

Add missing linking libraries for RDK

DNS SRV resolver: stick to the server address which is used most recently

Cleanup ast_channel references for both hold and unhold SIP channels

 - Add support for single or multiple call mode   
 - Make RTP timestamp always have a proper value 

Change the log lelvel and message to avoid misleading on devices without DECT support

- Relace the calls to ubus_free() with ast_ubus_free_context()
- Remove the definition of ubus_free_context() which has been moved to Asterisk

038e34d Fix the memory leak in blob_buf
679e002 Initialize the pointer to NULL for safety
a2cb58d Fix a memory corruption issue related to ubus_free
f5cb2e8 Add emergency attribute to the output of call_status ubus object