A new transport parameter 'symmetric_transport' has been added.
When a request from a dynamic contact comes in on a transport with
this option set to 'yes', the transport name will be saved and used
for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.
It's saved as a contact uri parameter named 'x-ast-txp' and will
display with the contact uri in CLI, AMI, and ARI output. On the
outgoing request, if a transport wasn't explicitly set on the
endpoint AND the request URI is not a hostname, the saved transport
will be used and the 'x-ast-txp' parameter stripped from the
outgoing packet.
* config_transport was modified to accept and store the new parameter.
* config_transport/transport_apply was updated to store the transport
name in the pjsip_transport->info field using the pjsip_transport->pool
on UDP transports.
* A 'multihomed_on_rx_message' function was added to
pjsip_message_ip_updater that, for incoming requests, retrieves the
transport name from pjsip_transport->info and retrieves the transport.
If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter
containing the transport name is added to the incoming Contact header.
* An 'ast_sip_get_transport_name' function was added to res_pjsip.
It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a
transport name if endpoint->transport is set or if there's an
'x-ast-txp' parameter on the uri and the uri host is an ipv4 or
ipv6 address. Otherwise it returns NULL.
* An 'ast_sip_dlg_set_transport' function was added to res_pjsip
which takes an ast_sip_endpoint, a pjsip_dialog, and an optional
pjsip_tpselector. It calls ast_sip_get_transport_name() and if
a non-NULL is returned, sets the selector and sets the transport
on the dialog. If a selector was passed in, it's updated.
* res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas
were modified to call ast_sip_dlg_set_transport() instead of their
original logic.
* res_pjsip/create_out_of_dialog_request was modified to call
ast_sip_get_transport_name() and pjsip_tx_data_set_transport()
instead of its original logic.
* Existing transport logic was removed from endpt_send_request
since that can only be called after a create_out_of_dialog_request.
* res_pjsip/ast_sip_create_rdata was converted to a wrapper around
a new 'ast_sip_create_rdata_with_contact' function which allows
a contact_uri to be specified in addition to the existing
parameters. (See below)
* res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated
since all it did was transport selection and that is now done in
ast_sip_create_dialog_uac and ast_sip_create_dialog_uas.
* 'contact_uri' was added to subscription_persistence. This was
necessary because although the parsed rdata contact header has the
x-ast-txp parameter added (if appropriate),
subscription_persistence_update stores the raw packet which
doesn't have it. subscription_persistence_recreate was then
updated to call ast_sip_create_rdata_with_contact with the
persisted contact_uri so the recreated subscription has the
correct transport info to send the NOTIFYs.
* res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since
all it did was transport selection and that is now done in
ast_sip_create_dialog_uac.
* pjsip_message_ip_updater/multihomed_on_tx_message was updated
to remove all traces of the x-ast-txp parameter from the
outgoing headers.
NOTE: This change does NOT modify the behavior of permanent
contacts specified on an aor. To do so would require that the
permanent contact's contact uri be updated with the x-ast-txp
parameter and the aor sorcery object updated. If we need to
persue this, we need to think about cloning permanent contacts into
the same store as the dynamic ones on an aor load so they can be
updated without disturbing the originally configured value.
You CAN add the x-ast-txp parameter to a permanent contact's uri
but it would be much simpler to just set endpoint->transport.
Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f
This change removes an assumption that when DTLS is stopped
an RTCP session will be present on the RTP session. This is not
always the case.
ASTERISK-26732
Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611
Dereferencing struct ast_autochan.chan without first calling
ast_autochan_channel_lock() is unsafe because the pointer could change at
any time due to a masquerade. Unfortunately, ast_autochan_channel_lock()
itself uses struct ast_autochan.chan unsafely and can result in a deadlock
if the original channel happens to get destroyed after a masquerade in
addition to the pointer getting changed.
The problem is more likely to happen with v11 and earlier because
masquerades are used to optimize out local channels on those versions.
However, it could still happen on newer versions if the channel is
executing a dialplan application when the channel is transferred or
redirected. In this situation a masquerade still must be used.
* Added a lock to struct ast_autochan to safely be able to use
ast_autochan.chan while trying to get the channel lock in
ast_autochan_channel_lock(). The locking order is the channel lock then
the autochan lock. Locking in the other direction requires deadlock
avoidance.
* Fix unsafe ast_autochan.chan usages in app_mixmonitor.c.
* Fix unsafe ast_autochan.chan usages in app_chanspy.c.
* app_chanspy.c: Removed unused autochan parameter from next_channel().
ASTERISK-26867
Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592
This commit adds support for RFC 5761: Multiplexing RTP Data and Control
Packets on a Single Port. Specifically, it enables the feature when
using chan_pjsip.
A new option, "rtcp_mux" has been added to endpoint configuration in
pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with
whatever it communicates with. Asterisk follows the rules set forth in
RFC 5761 with regards to falling back to standard RTCP behavior if the
far end does not indicate support for rtcp-mux.
The lion's share of the changes in this commit are in
res_rtp_asterisk.c. This is because it was pretty much hard wired to
have an RTP and an RTCP transport. The strategy used here is that when
rtcp-mux is enabled, the current RTCP transport and its trappings (such
as DTLS SSL session) are freed, and the RTCP session instead just
mooches off the RTP session. This leads to a lot of specialized if
statements throughout.
ASTERISK-26732 #close
Reported by Dan Jenkins
Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5
When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered
ASTERISK-26869 #close
Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525
A caller can leave the Queue() application after being bridged with a
member in a few ways:
* Caller or member hangup
* Caller is transferred somewhere else (blind or atx)
* Caller is externally redirected elsewhere
The first 2 scenarios are currently handled by subscribing to stasis
messages, but the 3rd is not explicitly covered. If a caller is
redirected away from the Queue() application, the member who was last
bridged with that caller will remain in an "In use" state until the
caller hangs up.
This patch adds handling of the caller leaving the queue via
redirection. We monitor the caller-member bridge, and if the caller is
the one that leaves, we treat it the same as we would a caller hangup.
ASTERISK-26400 #close
Reported by: Etienne Lessard
Change-Id: Iba160907770de5a6c9efeffc9df5a13e9ea75334
This change ensures that if no header_match option is set on an
identify an error message is not output stating the option is set
to an invalid value.
ASTERISK-26863
Change-Id: I239bc6d2319dd3da24ba96a38d4d6e9b5526d62a
This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.
Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.
ASTERISK-26863 #close
Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79d7)
Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range. e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.
The buffer overwrite is fixed two ways in this patch.
1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens. Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set. Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.
2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.
ASTERISK-26668
Change-Id: I555d97411140e47e0522684062d174fbe32aa84a
When using a non-dynamic peer address, build_peer() invalidates the
peer address structure by setting the address family to unspecified.
However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup()
will not amend the peer address if the cache is still valid, resulting
in peer connectivity failures.
To fix this, we call ast_dnsmgr_refresh() instead.
ASTERISK-26865
Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082
In some scenarios, such as when there may not be a terminal (such as
inside a Docker container), curl will apparently direct the progress bar
to stdout. This can cause extra data to be appended to a file curl'd
down to stdout, resulting in md5 verification failures.
This patch removes the progress bar, and tells curl to download the file
silently.
ASTERISK-26872 #close
Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c
* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.
Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48
This patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.
ASTERISK-26717 #close
Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124
During module loading of func_devstate, Asterisk emits the current
device state of all Custom device states currently stored in the AstDB.
This was erroneously including a new line character ('\n') to the end of
the device state, causing two new lines to be emitted in
DeviceStateChange AMI events.
Note that this only happened for those device state changes that
occurred during startup. Regular device state changes for Custom device
states are handled elsewhere, and did not have the newline.
ASTERISK-26643 #close
Reported by: Roman Bedros
Tested by: Matt Jordan
patches:
ami_devstate.diff uploaded by Roman Bedros (License 6842)
Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93
This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.
Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:
"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."
ASTERISK-25237 #close
Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70
When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.
This change just adds a check that the channel exists on the
session before querying it.
ASTERISK-26857
Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01
Bundled pjproject should now only rebuild if one of the menuselect
"Compiler Flags" options changes.
Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43
* res_musiconhold.c: Ensure the general section is not treated as
a moh class.
ASTERISK-26353 #close
Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d
Set a variable on the channel that indicates which attempt number we
are currently performing to allow for attempt-specific behavior.
ASTERISK-26568 #close
Reported by: Roman Shubovich
Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89
This change adds a PJSIP patch (which has been contributed upstream)
to allow the registration of IPv6 transport types.
Using this the res_pjsip_transport_websocket module now registers
an IPv6 Websocket transport and uses it for the corresponding
traffic.
ASTERISK-26685
Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647
* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.
ASTERISK-24562 #close
Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe
When doing some WebRTC testing, I found that the websocket would
disconnect whenever I attempted to place a call into Asterisk. After
looking into it, I pinpointed the problem to be due to the iostreams
change being merged in.
Under certain circumstances, a call to ast_iostream_read() can return a
negative value. However, in this circumstance, the websocket code was
treating this negative return as if it were a partial read from the
websocket. The expected length would get adjusted by this negative
value, resulting in the expected length being too large.
This patch simply adds an if check to be sure that we are only updating
the expected length of a read when the return from a read is positive.
ASTERISK-26842 #close
Reported by Mark Michelson
Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab
When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.
ASTERISK-26841
Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099
This change adds a few things to facilitate stream topology changing:
1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.
2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.
Tests have also been written which confirm the multistream and
non-multistream behavior.
ASTERISK-26839
Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9
George Joseph
Joshua Colp
zuul
Richard Mudgett
Mark Michelson
Torrey Searle
Sean Bright
Matt Jordan
Richard Begg
Daniel Journo
Jean Aunis