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

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

- 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

Adds a new configuration option, stir_shaken_profile, in pjsip.conf that
can be specified on a per endpoint basis. This option will reference a
stir_shaken_profile that can be configured in stir_shaken.conf. The type
of this option must be 'profile'. The stir_shaken option can be
specified on this object with the same values as before (attest, verify,
on), but it cannot be off since having the profile itself implies wanting
STIR/SHAKEN support. You can also specify an ACL from acl.conf (along
with permit and deny lines in the object itself) that will be used to
limit what interfaces Asterisk will attempt to retrieve information from
when reading the Identity header.

ASTERISK-29476

Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406

Put checks in place to limit how much we will actually download, as well
as a check for the data we receive at the start to ensure it begins with
what we would expect a certificate to begin with.

ASTERISK-29872

Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46

Omit "unsupported column type 'text'" warning in logs while
using text-type column in the PgSQL backend.

ASTERISK-29924 #close

Change-Id: I48061a7d469426859670db07f1ed8af1eb814712

This patch makes the Resource List Subscriptions (RLS) dynamic.
The asterisk updates the current subscriptions to reflect the changes
to the list on the subscriptions refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.

ASTERISK-29906 #close

Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3

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.

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"

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

- asterisk.telephony -> voice.line, "line" -> "id"
- asterisk.sip -> voice.sip.client, "line" -> "uri"
- asterisk.mwi -> voice.mwi

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

Also fix some miscellaneous compiling warnings.

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

There is an issue with copying RTP statistics from snapshot to cdr hence
call log does not contains RTP stats for incoming calls - for outgoing
call it works fine.

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>

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

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

Grzegorz Sluja authored 3 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

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>