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 4 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

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>

If a re-INVITE is received after we have sent a BYE request then it
is possible for no channel to be present on the session. If this
occurs we allow PJSIP to produce the offer instead. Since the call
is being hung up if it produces an incorrect offer it doesn't
actually matter. This also ensures that code which produces SDP
does not need to handle if a channel is not present.

ASTERISK-29381

Change-Id: I673cb88c432f38f69b2e0851d55cc57a62236042

When the MessageSend destination is in the form
PJSIP/<number>@<endpoint> and the endpoint's contact
URI already has a user component, that user component
will now be replaced with <number> when creating the
request URI.

ASTERISK_29404

Change-Id: I80e5910fa25c803d1440da0594a0d6b34b6b4ad5

Set preferred transport when querying the local address to use in
filter_on_tx_messages(). This prevents the module to erroneously select
the wrong transport if more than one transports of the same type (TCP or
TLS) are configured.

ASTERISK-29241

Change-Id: I598e60257a7f92b29efce1fb3e9a2fc06f1439b6

The text description needs to be the last thing on the AST_MODULE_INFO
line to be pulled in properly by menuselect.

Change-Id: I0c913e36fea8b661f42e56920b6c5513ae8fd832

Adds hook flash recognition support
for application/hook-flash.

ASTERISK-29460

Change-Id: I1d060fa89a7cf41244c98f892fff44eb1c9738ea

 * Implemented the new "to" parameter of the MessageSend()
   dialplan application.  This allows a user to specify
   a complete SIP "To" header separate from the Request URI.

 * Completely refactored the get_outbound_endpoint() function
   to actually handle all the destination combinations that
   we advertized as supporting.

 * We now also accept a destination in the same format
   as Dial()...  PJSIP/number@endpoint

 * Added lots of debugging.

ASTERISK-29404
Reported by Brian J. Murrell

Change-Id: I67a485196d9199916468f7f98bfb9a0b993a4cce

STIR/SHAKEN requires a Date header alongside the Identity header, so
that has been added. Still on the outgoing side, we were missing the
dest->tn section of the JSON payload, so that has been added as well.
Moving to the incoming side, URL checking has been added to the public
cert URL to ensure that it starts with http.

https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c

For connection oriented transports PJSIP uses factories to
produce transports. When doing a partial transport reload
we need to also move the factory of the transport over so
that anything referencing the transport (such as an endpoint)
has the factory available.

ASTERISK-29441

Change-Id: Ieae0fb98eab2d9257cad996a1136e5a62d307161

When unsubscribing from an endpoint technology a FRACK
would occur due to incorrect reference counting. This fixes
that issue, along with some other issues.

Fixed a typo in get_subscription when calling ao2_find as it
needed to pass the endpoint ID and not the entire object.

Fixed scenario where a subscription would get returned when
it shouldn't have been when searching based on endpoint
technology.

A doulbe unreference has also been resolved by only explicitly
releasing the reference held by tech_subscriptions.

ASTERISK-28237 #close
Reported by: Lucas Tardioli Silveira

Change-Id: Ia91b15f8e5ea68f850c66889a6325d9575901729

In multidomain environments, it is desirable to create
PJSIP endpoints with the domain info in the endpoint name
in pjsip_endpoint.conf.  This resulted in an error with
registrations, NOTIFY, and OPTIONS packet generation.

This commit will detect if there is an @ in the endpoint
identifier and generate the URI accordingly so NOTIFY and
OPTIONS From headers will generate correctly.

ASTERISK-28393

Change-Id: I96f8d01dfdd5573ba7a28299e46271dd4210b619

RTCP ICE candidates use a base address derived from the RTP
candidate. The port on the base address was not being updated to
the RTCP port.

This change sets the base port to the RTCP port and all is well.

ASTERISK-29433

Change-Id: Ide2d2115b307bfd3c2dfbc4d187515d724519040