Two previous commits, 620d9f47 and 6d980de2, allow to set up a call
without audio, again. That was introduced originally with commit f04d5fb8
but changed and broke over time. The original commit missed one
scenario: A [peer] section in sip.conf, which does not allow audio at
all. In that case, chan_sip rejected the call, although even when the
requester offered no audio. Now, chan_sip does not check whether there
is no audio format but checks whether there is no format in general. In
other words, if there is at least one format to offer, the call succeeds.

However, to prevent calls with no-audio, chan_sip still rejects calls
when both call parties (caller = requester of the call *and* callee =
[peer] section in sip.conf) included audio. In such a case, it is
expected that the call should have audio.

ASTERISK-29280

Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e

If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext.  You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.

* Added logic to iax2_call() and authenticate_reply() to print
  a warning and hanhup the call if encryption is requested and
  there's no secret or auth method.  This prevents the crash.

* Added the ability to specify a default "auth" in the [general]
  section of iax.conf.

ASTERISK-29624
Reported by: N A

Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254

The previous commit 6d980de2 fixed this issue in the core of Asterisk.
With that, each channel technology can be used without audio
theoretically. Practically, the channel-technology driver chan_sip
turned out to have an invalid check preventing that. chan_sip tested
whether there is at least one audio format. However, chan_sip has to
test whether there is at least one format. More cannot be tested while
requesting chan_sip because only the [general] capabilities but not the
[peer] caps are known yet. And the [peer] caps might not be a subset or
show any intersection with the [general] caps. This change here fixes
this.

The original commit f04d5fb8, thirteen years ago, contained a software
bug as it passed ANY audio capability to the channel-technology driver.
Instead, it should have passed NO audio format. Therefore, this
addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8.
Then, Asterisk 10 changed that from ANY to NO, but nobody reported since
then.

ASTERISK-29265

Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f

When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received

This allows applications to perform actions based on the failure
reason.

ASTERISK-29252 #close
Reported-by: Dan Cropp

Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4

This completes the fix for ASTERISK_24543. Only when the call is an
outgoing call, consult and append the configured format capabilities
(p->caps). When all audio formats got rejected the negotiated format
capabilities (p->jointcaps) contain no audio formats for incoming
calls. This is required when there are other accepted media streams.

ASTERISK-29258

Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9

There are a couple of parameters (datalen and data) that do not get set
in chan_pjsip_indicate which could cause an Invalid message to pop up
for things such as fax. This patch adds them to the frame.

Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8

Previously, chan_sip parsed all known media streams in an SDP offer
like video (and text) even when videosupport=no (and textsupport=no).
This wasted processor power. Furthermore, chan_sip accepted SDP offers,
including no audio but just video (or text) streams although
videosupport=no (or textsupport=no). Finally, chan_sip denied the whole
offer instead of individual streams when they had encryption (SDES-sRTP)
unexpectedly enabled.

ASTERISK-29238
ASTERISK-29237
ASTERISK-29222

Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755

session->channel doesn't exist until chan_pjsip creates it, so intead of
setting a channel variable every new incoming call sets one and the same
global variable.

This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
a newly created channel, it also removes a misleading reference to
channel->session used to fetch call pickup configuraion.

ASTERISK-29240

Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755

The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice.
This resulted in extra noise in logs (for example, "is making progress"
and "is ringing" get logged twice by app_dial), as well as in noise in
signalling: one incoming 183 Session Progress results in 2 outgoing 183-s.

This change splits the response handler into 2 functions:
 - one for updating HANGUPCAUSE, which is still called twice,
 - another that does the rest, which is called only once as before.

ASTERISK-28016
Reported-by: Alex Hermann

ASTERISK-28549
Reported-by: Gant Liu

ASTERISK-28185
Reported-by: Julien

Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940

Add channel reference count for PJSIP REFER. The call could be terminated
prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY
occurred several minutes later, it would attempt to access a session which was
no longer valid.  Terminate event subscription if pjsip_xfer_initiate() or
pjsip_xfer_send_request() fails in transfer_refer().

ASTERISK-29201 #close
Reported-by: Dan Cropp

Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435

This was dead code, test code introduced with Asterisk 13. This was
found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
partly fixes, not closes those two issues.

Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940

Fix the usual compile problem when someone adds a new callback to struct
ast_channel_tech.

Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32

In some circumstances it was possible for an INVITE
session to be destroyed while we were still using it.
This occurred due to the reference on the INVITE session
being released internally as a result of its state
changing to DISCONNECTED.

This change adds a reference to the INVITE session
which is released when our own session is destroyed,
ensuring that the INVITE session remains valid for
the lifetime of our session.

ASTERISK-29022

Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9

12 years ago, with ASTERISK_12115 the last four get/uses of socket.port
vanished. However, the struct member itself and all seven set/uses
remained as dead code.

ASTERISK-28798

Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59

ASTERISK-29144

Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62

ASTERISK-29146

Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41

RFC 8760 added new digest-access-authentication schemes. Testing
revealed that chan_sip does not pick MD5 if several schemes are offered
by the User Agent Server (UAS). This change does not implement any of
the new schemes like SHA-256. This change makes sure, MD5 is picked so
UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can
still be used. This should have worked since day one because SIP/2.0
already envisioned several schemes (see RFC 3261 and its augmented BNF
for 'algorithm' which includes 'token' as third alternative; note: if
'algorithm' was not present, MD5 is still assumed even in RFC 7616).

Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd

ASTERISK-29083 #close

Change-Id: I9ea25fba3ba8f63a47886894bd966e0f08d5e003

The module description needs to be on the same line as the
AST_MODULE_INFO or it is not parsed correctly.

Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21

Added to:
 * bridges/bridge_softmix.c
 * channels/chan_pjsip.c
 * include/asterisk/res_pjsip_session.h
 * main/channel.c
 * res/res_pjsip_session.c

There NO functional changes in this commit.

Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3

This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function
is called on a channel prior to answering a warning is issued and the
function returns unsuccessful.

ASTERISK-28878 #close

Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb

The ToHost parameter was not cleared when a peer's host value was
changed to dynamic. This causes invites to be sent to the original host.

ASTERISK-29011 #close

Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c

Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.

No functional changes were made with this commit.

Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c

When using the PSJIP_MEDIA_OFFER dialplan function it was not
overriding an endpoint's configured codecs on refresh unless
they had a shared codec between the two.

This patch makes it so whatever is set using PJSIP_MEDIA_OFFER
is used when creating the SDP for a refresh no matter what.

ASTERISK-28878 #close

Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6

The Streams API becomes the home for the core ACN capabilities.
These include...

 * Parsing and formatting of codec negotation preferences.
 * Resolving pending streams and topologies with those configured
   using configured preferences.
 * Utility functions for creating string representations of
   streams, topologies, and negotiation preferences.

For codec negotiation preferences:
 * Added ast_stream_codec_prefs_parse() which takes a string
   representation of codec negotiation preferences, which
   may come from a pjsip endpoint for example, and populates
   a ast_stream_codec_negotiation_prefs structure.
 * Added ast_stream_codec_prefs_to_str() which does the reverse.
 * Added many functions to parse individual parameter name
   and value strings to their respectrive enum values, and the
   reverse.

For streams:
 * Added ast_stream_create_resolved() which takes a "live" stream
   and resolves it with a configured stream and the negotiation
   preferences to create a new stream.
 * Added ast_stream_to_str() which create a string representation
   of a stream suitable for debug or display purposes.

For topology:
 * Added ast_stream_topology_create_resolved() which takes a "live"
   topology and resolves it, stream by stream, with a configured
   topology stream and the negotiation preferences to create a new
   topology.
 * Added ast_stream_topology_to_str() which create a string
   representation of a topology suitable for debug or display
   purposes.
 * Renamed ast_format_caps_from_topology() to
   ast_stream_topology_get_formats() to be more consistent with
   the existing ast_stream_get_formats().

Additional changes:
 * A new function ast_format_cap_append_names() appends the results
   to the ast_str buffer instead of replacing buffer contents.

Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56

chan_sip handle_response() function, for a 400 response to an INVITE,
calls handle_response_invite() and does not generate ACK.
handle_response_invite() does not recognize 400 response and has no
default response processing for unexpected responses, thus it does not
generate ACK either.
The ACK on response repetition comes from handle_response() mechanism
"We must re-send ACKs to re-transmitted final responses".

According to code history, 400 response specific processing was
introduced with commit
"channels/chan_sip: Add improved support for 4xx error codes"
This commit added support for :
- 400/414/493 in handle_response_subscribe() handle_response_register()
  and handle_response().
- 414/493 only in handle_response_invite().

This fix adds 400 response support in handle_response_invite().

ASTERISK-28957

Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad

A patch made a reference to the PJSIP_SC_NULL enumeration value, which
was added to pjproject 2.8 and above thus making it so Asterisk would
fail to compile with prior versions of pjproject.

This patch removes the reference, and instead initializes the value
to '0'.

ASTERISK-28886 #close

Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7

The change to how setvar works for various channels performed in
ASTERISK~23756 missed some required change in the dahdi channel,
where the variables are actually set while reading configuration.
This change should fix the issue.

ASTERISK-28955

Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274

When fax_gateway_framehook is called and a gateway hasn't already
been started, the framehook gets the t38 state for both the current
channel and the peer.  That call trickles down to the channel
driver which determines the state.  If either channel is hung up
(or in the process of being hung up), the channel driver's tech_pvt
is going to be NULL which, in the case of chan_pjsip, will cause a
segfault.

* Added a hangup check for both the channel and peer channel
  before starting a fax gateway.

* Added a check for NULL tech_pvt to chan_pjsip_queryoption
  so we don't attempt to reference a tech_pvt that's already
  gone.

ASTERISK-28923
Reported by: Yury Kirsanov

Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c

Some places in Asterisk did not treat the formats on a stream
as immutable when they are.

The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.

An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.

ASTERISK-28846

Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe

ASTERISK-28372
Reported by: Anton Satskiy

ASTERISK-24428
Reported by: sstream

Change-Id: I2b7432a9bf3b09dc8515297ff955636db7a6224c

ASTERISK-27195
Reported by: Joshua Roys

Change-Id: I6e72ecb874200dec7a3865c7babaf5ac0d3101de

It is possible to configure a TCP/TLS client without having a TCP/TLS
server. In that case, no error or warning was printed but the headers
Contact and Via in SIP REGISTER were "(null)".

ASTERISK-28798

Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2

If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a
digit begin before media, or rtp has been setup then it's possible the
outgoing channel will hear a constant DTMF tone upon answering.

This happens because when there is no media, or rtp chan_pjsip notifies
the core to initiate inband DTMF. However, upon digit end if media, and
rtp become available then chan_pjsip does not notify the core to stop
inband DTMF. Thus the tone continues playing.

This patch makes it so chan_pjsip only notifies the core to start
inband DTMF in only the required cases. Now if there is no media, or
rtp availabe upon digit begin chan_pjsip does nothing, but tells the
core it handled it.

ASTERISK-28817 #close

Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5

ASTERISK-28803

Change-Id: I15449621b68d0ad4d57b7c337c1167adb15135af

Change-Id: I0910c79196f2b7c7e5ad6f1db95e83800ac737a2

Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).

ASTERISK-28774
Reported-by: Michael Neuhauser

Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1

Fixes the following compile error:

    chan_vpb.cc:2688:26: error: catching polymorphic type
        ‘class std::exception’ by value

Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649

If the SSRC of a received RTP packet differed from the previous SSRC
an SSRC change control frame would be queued ahead of the media
frame. In the case of audio this would result in the format of the
audio frame not being checked, and if it differed or was not allowed
then it could cause the call to drop due to failure to set up a
translation path.

The chan_pjsip module will now no longer assume the first frame
will be the audio frame and instead goes through the complete list
to find it.

ASTERISK-28759

Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec

If you're for some reason out of RTP ports, chan_sip would previously
responde to an INVITE with a 403, which will fail the call.

Now, it returns a 503, allowing the device/proxy to retry the call on a
different machine.

ASTERISK-28718

Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90