When optimistic SRTP was on it was possible for us to still
set up a call without an audio stream if an offer was received
with required SRTP.

This change makes it so this scenario will now fail with a 488
response.

ASTERISK-26575

Change-Id: I7d14187037681f48879bd20319ac79d0877318f3

Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
rtp instance destroy for chan_unistim. Also several fixes
for displayed text translation.

Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc

Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b

Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94

Correct typo of end-pints to end-points
Re-wrap session timer parameter docs to max 80 chars wide; this
eases reading on terminals with lower resolution, commonly the case
for those with visual impairments.

ASTERISK-26573

Change-Id: I22c94459f4bb6b8a2f6713cfd22e87c32f204e6b
Signed-off-by: C.J. Collier <cjcollier@linuxfoundation.org>

This change fixes the SIP resolver such that if an IPv6 transport
is explicitly used it will resolve NAPTR, SRV, and AAAA records.

You can explicitly use one by specifying it on an endpoint.

ASTERISK-26571

Change-Id: I2ed3ce81b43a6a8a937c0ebc1b8ed2da5ac2ef36

ASTERISK-26558

Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e

This reverts commit f073f648.
Multiple testsuite failures were detected after the fact.

Change-Id: I968c380418bf65c7166f6ecff30fe8e247ea6682

This reverts commit 28926d1c.
Multiple testsuite failures were detected after the fact.

Change-Id: I8d4f5ccbb421a351d616254844ae7e5a31053edb

This reverts commit afef1b8e.
Multiple testsuite failures were detected after the fact.

Change-Id: Ib4cb0c0a6475681ce817f71b4050be25640ab67f

This reverts commit 39220230.

Multiple testsuite issues were discovered after the fact.

Change-Id: I848c4196dca2994b1a368087004326ea354cff95

2 of the sanitizers didn't have default values so in systems that
don't support sanitizers menuselect would spit out warnings.  They
were harmless but confusing.  They've now been set to "0".

Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58

sets the variable ABANDONED to TRUE if the call was not answered.

ASTERISK-26558

Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3

res_pjsip_sesssion was hooking into transaction and invite state
changes. One of the reasons for doing so was due to the
PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the
message sending process, and so we should call session supplements to
alter the outgoing message.

In reality, this event was meant to indicate that the message either
a) had already been sent, or
b) required a DNS lookup and would be sent when the DNS query
completed.

In case (a), this meant we were altering an already-sent
request/response for no reason. In case (b), this potentially meant we
could be trying to alter a request/response at the same time that the
DNS resolution completed. In this case, it meant we might be stomping on
memory being used by the thread actually sending the message. This
caused potential crashes and memory corruption.

This patch removes the calls to session supplements from the case where
the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to
alter the message at this point is too late, and it can cause nothing
but harm to try to do it. Because there were no longer any calls to the
handle_outgoing() function, it has been removed.

Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92

This is another case where manual frame deferral can be replaced with
centralized routines instead.

Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e
(cherry picked from commit d149c4b9e07eeb880d8428ad52c6fdb315cc15f5)

Rather than use manual frame deferral, just let the channel API do it
for us.

ASTERISK-26343

Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49
(cherry picked from commit 8ba3e2fc27f9966b8c7ce75c1eca6208613a9315)

AGI recently was modified to defer important frames. This was because
when AGI was used in a connected line interception routine, the
resulting connected line frame would end up getting discarded by the
AGI.

However, this caused bad behavior in other cases. Specifically, during a
transfer, if someone attempted to manually set the Caller ID on a
channel in an AGI, the deferred connected line frame would end up
overwriting what had been manually set in the AGI.

Since the initial issue was specific to interception routines, this
change removes the manual frame deferral from AGI and instead uses the
new frame deferral API in interception routines.

ASTERISK-26343 #close
Reported by Morton Tryfoss

Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208

There are several places in Asterisk that have duplicated logic
for deferring important frames until later.

This commit adds a couple of API calls to facilitate this automatically.

ast_channel_start_defer_frames(): Future reads of deferrable frames on
this channel will be deferred until later.

ast_channel_stop_defer_frames(): Any frames that have been deferred get
requeued onto the channel.

ASTERISK-26343

Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641