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

By default Asterisk reports the PJSIP version in a SOFTWARE attribute
of every STUN packet it sends. This may not be desired in a production
environment, and RFC5389 recommends making the use of the SOFTWARE
attribute a configurable option:

https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2

This patch adds a `stun_software_attribute` yes/no option to make it
possible to omit the SOFTWARE attribute from STUN packets.

ASTERISK-29434

Change-Id: Id3f2b1dd9584536ebb3a1d7e8395fd8b3e46860b

RFC7616 and RFC8760 allow more than one WWW-Authenticate or
Proxy-Authenticate header per realm, each with different digest
algorithms (including new ones like SHA-256 and SHA-512-256).
Thankfully however a UAS can NOT send back multiple Authenticate
headers for the same realm with the same digest algorithm.  The
UAS is also supposed to send the headers in order of preference
with the first one being the most preferred.  We're supposed to
send an Authorization header for the first one we encounter for a
realm that we can support.

The UAS can also send multiple realms, especially when it's a
proxy that has forked the request in which case the proxy will
aggregate all of the Authenticate headers and then send them all
back to the UAC.

It doesn't stop there though... Each realm can require a
different username from the others.  There's also nothing
preventing each digest algorithm from having a unique password
although I'm not sure if that adds any benefit.

So now... For each Authenticate header we encounter, we have to
determine if we support the digest algorithm and, if not, just
skip the header.  We then have to find an auth object that
matches the realm AND the digest algorithm or find a wildcard
object that matches the digest algorithm. If we find one, we add
it to the results vector and read the next Authenticate header.
If the next header is for the same realm AND we already added an
auth object for that realm, we skip the header. Otherwise we
repeat the process for the next header.

In the end, we'll have accumulated a list of credentials we can
pass to pjproject that it can use to add Authentication headers
to a request.

NOTE: Neither we nor pjproject can currently handle digest
algorithms other than MD5.  We don't even have a place for it in
the ast_sip_auth object. For this reason, we just skip processing
any Authenticate header that's not MD5.  When we support the
others, we'll move the check into the loop that searches the
objects.

Changes:

 * Added a new API ast_sip_retrieve_auths_vector() that takes in
   a vector of auth ids (usually supplied on a call to
   ast_sip_create_request_with_auth()) and populates another
   vector with the actual objects.

 * Refactored res_pjsip_outbound_authenticator_digest to handle
   multiple Authenticate headers and set the stage for handling
   additional digest algorithms.

 * Added a pjproject patch that allows them to ignore digest
   algorithms they don't support.  This patch has already been
   merged upstream.

 * Updated documentation for auth objects in the XML and
   in pjsip.conf.sample.

 * Although res_pjsip_authenticator_digest isn't affected
   by this change, some debugging and a testsuite AMI event
   was added to facilitate testing.

Discovered during OpenSIPit 2021.

ASTERISK-29397

Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281

RFC 4235 Section 4.1.6 describes XML elements that should be
sent to subscribed endpoints to identify the local and remote
participants in the dialog.

This patch adds this functionality to PJSIP by iterating through the
ringing channels causing the NOTIFY, and inserts the channel info
into the dialog so that information is properly passed to the endpoint
in dialog-info+xml.

ASTERISK-24601
Patch submitted: Joshua Elson
Modified by: Joseph Nadiv and Sean Bright
Tested by: Joseph Nadiv

Change-Id: I20c5cf5b45f34d7179df6573c5abf863eb72964b

STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.

The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.

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

Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c