If Asterisk sends out and INVITE and receives a challenge with a
different nonce value each time, it will continually send out INVITEs,
even if the call is hung up. The endpoint must be configured for
outbound authentication in order for this to occur. A limit has been set
on outbound INVITEs so that, once reached, Asterisk will stop sending
INVITEs and the transaction will terminate.
ASTERISK-29013
Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7
* Wildcards in #includes are now properly expanded
* Implement operators for Section class to allow sorting
ASTERISK-29142 #close
Change-Id: I9b9cd95f4cbe5c24506b75d17173c5aa1a83e5df
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
Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0
Warning:
say.c:2371:24: error: ‘%d’ directive output may be truncated writing
between 1 and 11 bytes into a region of size 10
[-Werror=format-truncation=]
2371 | snprintf(buf, 10, "%d", num);
say.c:2371:23: note: directive argument in the range [-2147483648, 9]
That's not possible though, as the if() starts out checking for (num < 0),
making this Warning a false positive.
(Also replaced some else<TAB>if with else<SP>if while in the vicinity.)
Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a
This patch initializes a couple of local variables to some default values.
Interestingly, in the 'pj_status_t dlg_status' case the value not being
initialized caused memory to grow, and not be recovered, in the off nominal
path (at least on my machine).
Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7
Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not
come with GMime 3.0. aptitude ignores any missing package. Therefore,
it installs the correct package(s). However, in Ubuntu 18.04 LTS and
Ubuntu 20.04 LTS, both versions are installed alongside although only
one is really needed.
Change-Id: Ic58aa9f2e131d94671f286f17dbd61e1ccbabcb7
Commit 44bb0858cb ("debugging: Add enough
to choke a mule") accidentally removed calls to
ast_sip_message_apply_transport when it was attempting to just add
debugging code.
The kiss of death was saying that there were no functional changes in
the commit comment.
This makes outbound calls that use the 'flow' transport mechanism fail,
since this call is used to relay headers into the outbound INVITE
requests.
ASTERISK-29124 #close
Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a
Note to maintainers: Lua 5.4, Lua 5.3, and Lua 5.2 have not been tested
at runtime with pbx_lua. Until then, use the lowest available version
of Lua, if you enabled the module pbx_lua at all.
Change-Id: Ie5270448b11fcb4e2a53d899e4fe7fea793ce7e0
This changes the outgoing offer call preference
default option to match the behavior of previous
versions of Asterisk.
The additional advanced codec negotiation options
have also been removed from the sample configuration
and marked as reserved for future functionality in
XML documentation.
The codec preference options have also been fixed to
enforce local codec configuration.
ASTERISK-29109
Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2
Added debug logging categories that allow a user to output debug
information based on a specified category. This lets the user limit,
and filter debug output to data relevant to a particular context,
or topic. For instance the following categories are now available for
debug logging purposes:
dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
stun, stun_packet
These debug categories can be enable/disable via an Asterisk CLI command.
While this overrides, and outputs debug data, core system debugging is
not affected by this patch. Statements still output at their appropriate
debug level. As well backwards compatibility has been maintained with
past debug groups that could be enabled using the CLI (e.g. rtpdebug,
stundebug, etc.).
ASTERISK-29054 #close
Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849
(cherry picked from commit 56028426de)
When handling a send_message request to a non-existing endpoint, the response's
body is overriden and not properly freed.
ASTERISK-29108
Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8
In the event that the desired extension already exists,
ast_add_extension2_lockopt() will free the 'data' it is passed before
returning an error, so we should not be freeing it ourselves.
Additionally, there were two places where ast_add_extension2_lockopt()
could return an error without also freeing the 'data' pointer, so we
add that.
ASTERISK-29097 #close
Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae
app_confbridge now has the ability to set the estimated bitrate on an
SFU bridge. To use it, set a bridge profile's remb_behavior to "force"
and set remb_estimated_bitrate to a rate in bits per second. The
remb_estimated_bitrate parameter is ignored if remb_behavior is something
other than "force".
Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a
Stop advertising RFC2833 support on the rtp_engine when DTMF mode is
auto but no tel_event was found inside SDP file.
On an incoming call create_rtp will be called and when session->dtmf is
set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without
looking at the SDP file.
Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND
but continued to advertise RFC2833 support.
This meant the native_rtp bridge would falsely consider the two channels
as compatible. In addition to changing the DTMF mode we now set or
remove the AST_RTP_PROPERTY_DTMF.
The property is checked in ast_rtp_dtmf_compatible and called by
native_rtp_bridge_compatible.
ASTERISK-29051 #close
Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287
Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and
AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice,
resulting in to 181 being generated.
Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab
Only track our sample offset if we are playing a non-announcement file,
otherwise we will skip that number of samples when we start playing the
first MoH file.
ASTERISK-24329 #close
Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc
When constructing a stream name based on the media type
and position the allocated name was not being freed
causing a leak.
Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de
Because we use shared thread-local cURL instances, we need to ensure
that the state of the cURL instance is correct before each invocation.
In the case of custom headers, we were not resetting cURL's internal
HTTP header pointer which could result in a crash if subsequent
requests do not configure custom headers.
ASTERISK-29085 #close
Change-Id: I8b4ab34038156dfba613030a45f10e932d2e992d
The ast_sip_dialog_get_session function returns the session
with reference count increased. This was not taken into
account and was causing sessions to remain around when they
should not be.
ASTERISK-29089
Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8
Sometimes not play MOH on bridge.
ASTERISK-29081
Reported-by: Michal Hajek <michal.hajek@daktela.com>
Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232
ast_trace() was always emitting messages when it's level was set to -1
because it was ignoring scope_level.
Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506
The module description needs to be on the same line as the
AST_MODULE_INFO or it is not parsed correctly.
Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21
When a channel joins a bridge, we do topology change requests on all
existing channels to add the new participant to them. However the
announcer channel will return an error because it doesn't support
topology in the first place. Unfortunately, there doesn't seem to be a
reliable way to tell if the error is expected or not so the error is
ignored for all channels. If the request fails on a "real" channel,
that channel just won't get the new participant's video.
Change-Id: Ic95db4683f27d224c1869fe887795d6b9fdea4f0
Implemention of History-Info capable of interworking with Diversion
Header following RFC7544
ASTERISK-29027 #close
Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6
The recent 491 changes introduced a check to determine if the active
and pending topologies were equal and to suppress the re-invite if they
were. When a re-invite is sent for a COLP-only change, the pending
topology is NULL so that check doesn't happen and the re-invite is
correctly sent. Of course, sending the re-invite sets the pending
topology. If a 491 is received, when we resend the re-invite, the
pending topology is set and since we didn't request a change to the
topology in the first place, pending and active topologies are equal so
the topologies-equal check causes the re-invite to be erroneously
suppressed.
This change checks if the topologies are equal before we run the media
state resolver (which recreates the pending topology) so that when we
do the final topologies-equal check we know if this was a topology
change request. If it wasn't a change request, we don't suppress
the re-invite even though the topologies are equal.
ASTERISK-29014
Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314
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
When both Asterisk and a UA send re-invites at the same time, both
send 491 "Transaction in progress" responses to each other and back
off a specified amount of time before retrying. When Asterisk
prepares to send its re-invite, it sets up the session's pending
media state with the new topology it wants, then sends the
re-invite. Unfortunately, when it received the re-invite from the
UA, it partially processed the media in the re-invite and reset
the pending media state before sending the 491 losing the state it
set in its own re-invite.
Asterisk also was not tracking re-invites received while an existing
re-invite was queued resulting in sending stale SDP with missing
or duplicated streams, or no re-invite at all because we erroneously
determined that a re-invite wasn't needed.
There was also an issue in bridge_softmix where we were using a stream
from the wrong topology to determine if a stream was added. This also
caused us to erroneously determine that a re-invite wasn't needed.
Regardless of how the delayed re-invite was triggered, we need to
reconcile the topology that was active at the time the delayed
request was queued, the pending topology of the queued request,
and the topology currently active on the session. To do this we
need a topology resolver AND we need to make stream named unique
so we can accurately tell what a stream has been added or removed
and if we can re-use a slot in the topology.
Summary of changes:
* bridge_softmix:
* We no longer reset the stream name to "removed" in
remove_all_original_streams(). That was causing multiple streams
to have the same name and wrecked the checks for duplicate streams.
* softmix_bridge_stream_sources_update() was checking the old_stream
to see if it had the softmix prefix and not considering the stream
as "new" if it did. If the stream in that slot has something in it
because another re-invite happened, then that slot in old might
have a softmix stream but the same stream in new might actually
be a new one. Now we check the new_stream's name instead of
the old_stream's.
* stream:
* Instead of using plain media type name ("audio", "video", etc) as
the default stream name, we now append the stream position to it
to make it unique. We need to do this so we can distinguish multiple
streams of the same type from each other.
* When we set a stream's state to REMOVED, we no longer reset its
name to "removed" or destroy its metadata. Again, we need to
do this so we can distinguish multiple streams of the same
type from each other.
* res_pjsip_session:
* Added resolve_refresh_media_states() that takes in 3 media states
and creates an up-to-date pending media state that includes the changes
that might have happened while a delayed session refresh was in the
delayed queue.
* Added is_media_state_valid() that checks the consistency of
a media state and returns a true/false value. A valid state has:
* The same number of stream entries as media session entries.
Some media session entries can be NULL however.
* No duplicate streams.
* A valid stream for each non-NULL media session.
* A stream that matches each media session's stream_num
and media type.
* Updated handle_incoming_sdp() to set the stream name to include the
stream position number in the name to make it unique.
* Updated the ast_sip_session_delayed_request structure to include both
the pending and active media states and updated the associated delay
functions to process them.
* Updated sip_session_refresh() to accept both the pending and active
media states that were in effect when the request was originally queued
and to pass them on should the request need to be delayed again.
* Updated sip_session_refresh() to call resolve_refresh_media_states()
and substitute its results for the pending state passed in.
* Updated sip_session_refresh() with additional debugging.
* Updated session_reinvite_on_rx_request() to simply return PJ_FALSE
to pjproject if a transaction is in progress. This stops us from
creating a partial pending media state that would be invalid later on.
* Updated reschedule_reinvite() to clone both the current pending and
active media states and pass them to delay_request() so the resolver
can tell what the original intention of the re-invite was.
* Added a large unit test for the resolver.
ASTERISK-29014
Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb
Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20).
This is fine for normal cases, but if the hostname is longer than 20, it returns error and then
failed to register the contact address of the peer.
Normally, 20 characters limitation for the hostname is fine, but with the cloud env.
So, increased the size to 255.
ASTERISK-29056
Change-Id: Iac52c8c35030303cfa551bb39f410b33bffc507d
Currently, it was not possible to create bridge with video_mode single.
This made hard to put the bridge in a vidoe_single mode.
So, added video_single option for Bridge creation using the ARI.
This allows create a bridge with video_mode single.
ASTERISK-29055
Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae
There's a race condition with bridging where a bridge can be torn down
causing the bridge_channel's ast_channel to become NULL when it's still
needed. This particular case happened with attended transfers, but the
crash occurred when trying to publish a stasis message. Now, the
bridge_channel is locked, a ref to the ast_channel is obtained, and that
ref is passed down the chain.
Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814
Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again.
Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6
This fixes a bug introduced mistakenly in ASTERISK-25665:
If leave-empty is enabled, a call may sometimes be removed from
a queue without recording it as abandoned.
This causes Asterisk to not generate an abandon event for that
call, and for the queue abandoned counter to be incorrect.
ASTERISK-29043 #close
Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7
If you run ast_coredumper --tarball-coredumps in the same directory
as the actual coredump, tar can fail because the link to the
actual coredump becomes recursive. The resulting tarball will
have everything _except_ the coredump (which is usually what
you need)
There's also an issue that the directory name in the tarball
is the same as the coredump so if you extract the tarball the
directory it creates will overwrite the coredump.
So:
* Made the link to the coredump use the absolute path to the
file instead of a relative one. This prevents the recursive
link and allows tar to add the coredump.
* The tarballed directory is now named <coredump>.output instead
of just <coredump> so if you expand the tarball it won't
overwrite the coredump.
Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea
When fixing issues uncovered by GCC10 a copy of the parker UUID
was removed accidentally. This change restores it so that the
subscription has the data it needs.
ASTERISK-29042
Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a
With the latest Linux, 'ifconfig' is not installed on default anymore.
Furthermore, the output of the current net-tools 'ifconfig' changed.
Therefore, parsing failed. This update uses 'ip addr show' instead.
Finally, the service for the external IP changed.
Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0
