Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.
ASTERISK-26864
Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6
POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.
Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.
Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48
We are currently passing in the capacity of the read buffer instead of the
number of bytes that we actually read off the wire.
Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36
The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.
Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088
We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.
Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0
If local_net is not defined on a transport, transport_state->localnet
will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in
this case, causing the external_media_address, if set, to be skipped.
This patch causes us to only check if we are sending within a network if
local_net is defined.
ASTERISK-26879 #close
Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb
Currently a wildcard address is used for the local RTP socket, which
will not always result in the same address as used by the SIP socket
(e.g. if explicit transport addresses are configured).
Use the transport's host address when binding new local RTP sockets if
available.
ASTERISK-26851
Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a
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
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 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
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
When transfering to a URI without an extension, ensure that the
s extension of the dialplan is entered
ASTERISK-26869 #close
Change-Id: I07403df66cf93f09e00a40ab5b41bfc6f72b1525
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
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)
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
The Binaural Rendering patches in the master branch required
menuselect to be updated with a new support type called 'option'.
This allows binaural rendering to be turned on or off when
bridge_softmix is built. This patch backports the 'option'
functionality to the 13 and 14 branches.
Here's what it looks like in menuselect:
[*] bridge_simple
[*] bridge_softmix
--- Module Options ---
[ ] binaural_rendering_in_bridge_softmix
To create an option for a module, you can create (or update) the
menuselect-tree xml snippet in the directory where the module
resides and add a member element with an 'option' support_level.
Example (abbreviated) from bridges/bridges.xml:
<member name="binaural_rendering_in_bridge_softmix"
displayname="Enable binaural rendering in bridge_softmix"
remove_on_change="bridges/bridge_softmix.o bridges/bridge_softmix.so">
<support_level>option</support_level>
<depend>bridge_softmix</depend>
<depend>fftw3</depend>
<defaultenabled>no</defaultenabled>
</member>
The 'name' will be added or removed from the MENUSELECT_<dir>
make variable following the standard module "missing means yes"
rules.
Example (abbreviated) from bridges/Makefile:
ifeq ($(findstring binaural_rendering,$(MENUSELECT_BRIDGES)),)
bridge_softmix.o: _ASTCFLAGS+=-DBINAURAL_RENDERING
bridge_softmix.so: LIBS+=$(FFTW3_LIB)
endif
Change-Id: I66d23755ed6e81f8d439cad410f2ffa7c30f25ad
Bundled pjproject should now only rebuild if one of the menuselect
"Compiler Flags" options changes.
Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43
