Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list.
ao2_container_alloc is now restricted to modules only and is being
removed from Asterisk 17.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
ASTERISK-28103
Change-Id: I99bee23895baa0a117985d51683f7963b77aa190
PJSIP assumes that these header names are not allocated, does not clone
the name strings when reusing headers.
Block unload of res_pjsip_caller_id until shutdown to ensure static
memory stays valid. It was previously unsafe to unload while any
sessions are active.
Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f
The presence of Record-Route in re-invites is optional, thus it is
important to make sure the dialog doesn't have a routset before
rewriting the contact header.
ASTERISK-28129 #close
Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc
The marker bit set on the voice packet indicates the start
of a new stream and a new time stamp.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet.
If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.
This bug was caught using Cisco ip-phone SPA50X and codec g722.
On SIP session update the SPA50X resets stream indicating it with market bit
and a new timestamp is twice smaller then the previous.
ASTERISK-28162 #close
Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620
When Asterisk's taskprocessors get overloaded we need to reduce the work
load. res_pjsip currently ignores new SIP requests and relies on SIP
retransmissions in the hope that the overload condition will clear soon
enough to handle the retransmitted SIP request.
This change adds the following code after ast_taskprocessor_alert_get()
has returned TRUE:
1- identifies transport type. If non-udp then send a 503 response
2- if transport type is udp/udp6 then ignore, as before.
Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836
The use of a '|' in the "global/debug" synopsis documentation caused the
generated html table on the wiki to add an extra column that included the
text after the pipe.
This patch replaces the pipe with a comma.
ASTERISK-28150
Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719
The current round-robin method does not take the current taskprocessor
load into consideration when distributing requests. Using the least-size
method the request goes to the taskprocessor that is servicing the least
number of active tasks at the current time.
Longer running tasks with the round-robin method can delay processing
tasks.
* Change the algorithm from round-robin to least-size for picking the
PJSIP taskprocessor from the default serializer pool.
Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd
This patch adds new options 'trust_connected_line' and 'send_connected_line'
to the endpoint.
The option 'trust_connected_line' is to control if connected line updates
are accepted from this endpoint.
The option 'send_connected_line' is to control if connected line updates
can be sent to this endpoint.
The default value is 'yes' for both options.
Change-Id: I16af967815efd904597ec2f033337e4333d097cd
The module 'res_pjsip_notify' inefficiently makes a lot of DB requests
on CLI completion on the endpoint.
For example if there are 10k endpoints the module makes 10k requests
of these 10k records.
Even if a part of the endpoint entered
the module makes the same 10k requests and then filtered them by itself.
This patch gathers endpoints container by prefix
and adds all gathered endpoints to completion at once.
ASTERISK-28137 #close
Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b
Add a new global flag to res_pjsip to allow the callerid to be used
as the username in the contact header. This allows chan_pjsip to have
the same behavour as chan_sip
ASTERISK-28087 #close
Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are still available for use but only in modules. Only
ao2_container_alloc remains due to it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
Most were in comments. A couple were in warning messages.
Pointed out by Jonathan H on the Asterisk users mailing list.
Change-Id: I6286939dff5d0a27a2758140570106f1cb351855
This patch sets the callerid_tag to empty string by default.
If the callerid_tag is set to NULL then the tag does not
become part of a connected line update.
For example:
Alice's tag is "Alice".
Bob's tag is empty.
Charlie's tag is "Charlie".
Alice calls Bob and then does attended transfer to Charlie.
When Alice hangs up the CONNECTEDLINE(tag) is "Alice"
on the interception routine on the Charlie's channel, but should be empty.
Ths patch also fix memory leaks if there are more then one options
"callerid", "callerid_tag", "voicemail_extension" and "contact_user"
in the pjsip.conf endpoint definition.
Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4
The return status when there was no change in statsd.conf was incorrect.
This resulted in the wrong status message on the CLI when reloading the
module.
* Fixed cleanup on initial load if initializing statsd failed.
Change-Id: Id24fae75f1a7ff584a444a5680e867d989792481
I think this module is so screwed up that it doesn't work anymore. Even
with these attempts to fix things it still won't gracefully shut down.
The module refs will not go to zero to allow unloading the module.
* Fix module ref counting dealing with the SMDI interface object. There
were several off-nominal paths that unbalanced the module ref count. Also
the destructor freed the ao2 object itself which is bad. Made the
smdi_read thread not hold its own ref to the SMDI interface object so when
all refs go away the destructor will stop the listener thread.
* Fixed the smdi_load() return code of 1 concerning the number of
listeners. The test was inverted.
Change-Id: Ic288db51b58e395d6a2fc3847f77176c16988784
When networks experience disruptions, there can be large gaps of time
between receiving packets. When strictrtp is enabled, this created
issues where a flood of packets could come in and be seen as an attack.
Another option - seqno - has been added to the strictrtp option that
ignores the time interval and goes strictly by sequence number for
validity.
Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71
CLI command 'pjsip show contacts' inefficiently make a lot of DB requests.
For example if there are 10k aors then asterisk requests these 10k records
of aor and then does 10k requests of contact - one request per aor.
Even if use 'like <pattern>' the asterisk requests all aor's and contact's
records and then filters them by itself.
This patch gathers contact's container by
- retrieving all dynamic contacts by regex (filtered by reg_server)
- retrieving all aors with permanent contacts
- finally filters container by regex
ASTERISK-28077 #close
Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618
Fixed an issue that resulted in "Allocation failed" each time an ARI
request was made to start playing MOH on a bridge.
In bridge_moh_create() we were attaching the after bridge callbacks to
chan which is the ;1 channel of the unreal channel pair. We should have
attached them to the ;2 channel which is pushed into the bridge by
ast_unreal_channel_push_to_bridge(). The callbacks are called when the
specific channel leaves the bridging system. Since the ;1 channel is
never put into a bridge the callbacks never get called. The callbacks
then never remove the moh_wrapper from the app_bridges_moh container. As
a result we cannot find the channel associated with the wrapper to start
MOH because it has hungup. This is the reason causing the reported issue.
* Rather than using after bridge callbacks to cleanup, we now have
moh_channel_thread() doing the cleanup when the channel hangs up.
* Fixed moh_channel_thread() accumulating control frames on the stasis
bridge MOH channel until MOH is stopped. Control frames are no longer
accumulated while MOH is playing.
* Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may
or may not return a channel ref. As a result ast_ari_bridges_start_moh()
wouldn't know it may have a channel ref to release.
stasis_app_bridge_moh_channel() will now return a ref with the channel it
returns.
* Eliminated RAII_VAR in bridge_moh_create().
ASTERISK-26094 #close
Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c
This change raises a testsuite event to provide what port
Asterisk has actually allocated for RTP. This ensures that
testsuite tests can remove any assumption of ports and instead
use the actual port in use.
ASTERISK-28070
Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044
On SQL error there is not diagnostic information about this error.
There is only
WARNING res_odbc.c: SQL Execute error -1!
The function ast_odbc_print_errors calls a SQLGetDiagField to get the number
of available diagnostic records, but the SQLGetDiagField returns 0.
However SQLGetDiagRec could return one diagnostic records in this case.
Looking at many example of getting diagnostics error information
I found out that the best way it's to use only SQLGetDiagRec
while it returns SQL_SUCCESS.
Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR
to res_config_odbc.
ASTERISK-28065 #close
Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1
'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults
if they are not present in the updated configuration file.
Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670
The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.
* No longer allocate memory from the stack in a loop to parse the header
values. NOTE: There is a slight API change when using the passed in
strings as is. We now require the passed in strings to no longer have
leading or trailing whitespace. This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.
ASTERISK-28013 #close
Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a
This change brings in PJSIP 2.8, removes all the patches
that were merged upstream, and makes a minor change to
support a breaking change that was done.
ASTERISK-28059
Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189
Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name
store IPv6 addresses without enclosing brackets. This causes some log
output to be confusing because it is difficult to separate the IPv6
address from a port specification.
* Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and
pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6
output.
* When a pj_sockaddr is not available, explicitly wrap IPv6 addresses
in brackets.
* When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure
to also set pjsip_rx_data.pkt_info.src_addr.
Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8
As they're not actively used, they only grow stale. The moduleinfo field itself
is kept in Asterisk 13/15 for ABI compatibility.
ASTERISK-28046 #close
Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc
The bundled version of pjproject has a patch for Solaris compatability
that changes the definition of various socket structures which we need
to account for when compiling against a non-bundled version.
ASTERISK-28049 #close
Change-Id: Ia1ea47c433fc2d915115193ee889a752373925f0
On MOH activation, moh_files_readframe() is called while the current
stream attached to the channel is NULL and it calls ast_moh_files_next()
immediately. However, it won't call ast_moh_files_next() again if sample
reading fails. The failure may occur because res_musiconhold retains the
last sample reading position in the channel data and MOH during the
previous hold/retrieve just reached EOF. Obviously, a bit of bad luck is
required here.
* Restructured moh_files_readframe() to try a second time to start MOH if
there was no stream setup and the saved position was at EOF. Also added
comments describing what is going on for each step.
ASTERISK-28029
Change-Id: I1508cf2c094f8feca22d6f76deaa9fdfa9944860
Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address
needs to be rendered to a string and then parsed into the correct
structure. This also involves a call to getaddrinfo(3). The same is true
for the inverse operation.
Instead, because we know the internal structure of both ast_sockaddr and
pj_sockaddr, we can translate directly between the two without the
need for an intermediate string.
Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761
The T.38 fax gateway state machine can cause the fax gateway
to be started more than once on a channel depending on the
responses of the remote endpoint. This would previously leak
the channel name, channel unique id, and underlying fax engine
state. This change instead makes it so that if the fax gateway
session is already present and not reserved the fax gateway
is not started again.
ASTERISK-27981
Change-Id: I552d95086860cb18f2522ee40ef47b13b6da2e0e
SIP responses over WebSockets when the client is using IPv6 have invalid
Via headers according to RFC 3261. The 'received' header parameter
should not be wrapped in brackets if it is an IPv6 address.
When src_name is populated by the built-in PJSIP transports, the code
uses pj_sockaddr_print() with 'flags' set to 0, meaning that the
brackets are not rendered around IPv6 addresses.
This may be related to ASTERISK~27101.
See also: https://github.com/onsip/SIP.js/pull/594
ASTERISK-28020 #close
Change-Id: I8ea9d289901b837512bee2ca2535e3dc14f04d77
This new option can be passed for ./configure or
./tests/CI/buildAsterisk.sh to prevent download/install of binary
modules.
Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES
will result in binary modules being enabled even if the build target is
incompatible with those modules. This includes CI scripts which enable
categories before disabling specific modules.
If more binary modules are offered in the future this will help avoid
accidentally downloading them if unwanted or incompatible. Adding a
binary module will only require creating a new menuselect entry similar
to the existing ones, it will not be necessary to modify the CI scripts.
Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166
The realtime text timer pops regularly and sends text frames even if
the buffer is empty. This causes a lot of unecessary debug logging.
* Made red_write() test if we need to send a frame before calling
ast_rtp_write()
ASTERISK-28002
Reported by: Emmanuel BUU
Tested by: Emmanuel BUU
Change-Id: Icf81310c3b8080b615a42060afc02ab41f9523dd
The allow_unqualified_fetch option for the sorcery realtime backend
blocked actually fetching all rows when the option is set to warn.
* Made issue a warning and actually do the request when
allow_unqualified_fetch=warn is set.
Change-Id: I74456c80a03a62dce66fc3dc3cb0cf2351ac4312
sip_options_get_endpoint_state_compositor_state leaked a reference to
the first available endpoint state compositor that was found.
Change-Id: Idb6be19f7219b6eed1dfb19c1e740dd40cb3fdc7
This change adds the "party" parameter to the Remote-Party-ID header
which indicates which party the header information is applicable
to. In Asterisk this is determined on whether we are the calling
or called party. This is added to improve interoperability with some
implementations.
ASTERISK-28006
Change-Id: I1eec3e377ffff8633b5c1dd59a05e9533122cfca
If in the initial sdp the caller doesn't include the line
a=rtcp-mux
Then asterisk shoud not include rtcp-mux in the response regardless
of rtcp-mux being enabled on the endpoint
ASTERISK-28007 #close
Change-Id: I58e9b9f40a139afc0da5de41906cc608fb62adc7
* Don't include pjlib.h twice in res_pjsip.h
* Consistently use #include <> form for pjproject includes.
(pjsip.h and pjlib.h)
Change-Id: I3f7b42044840de64edf7e9d7695cb60c45990dc7
This change removes a sorcery lookup for retrieving all
contacts at the end of the registration process by keeping
track of the contacts that are added/updated/deleted.
This ensures at the end of the process the container of
contacts we have is the current state.
Pool usage has also been reduced by allocating one for
usage throughout the handling of a REGISTER and resetting
it to a clean state. This ensures that in most cases
we allocate once and just reuse it.
ASTERISK-28001
Change-Id: I1a78b2d46f9a2045dbbff1a3fd6dba84b612b3cb
This patch adds regular expression support to make the identify section's
match_header option more useful when attempting to match complex headers
like the 'To' or 'From' headers. The 'From' header has variable
components such as the tag parameter that you cannot predict. To specify
a regular expression put slashes around the regular expression in place of
the header value.
[identify-alice]
type=identify
endpoint=alice
match_header=From: /<sip:alice@127\\.0\\.0\\.1>/
* Added regex support to match_header so you could match a 'To' header
among other complex headers.
Fixed reported crashes when trying to match special headers like 'Contact'.
The identify section's match_header method used code that assumed you were
matching a generic header. Any other type of header could cause a crash
if the header structure variant did not match the generic header enough.
* Made use code that will work for any header type instead of code
specific to generic headers.
Other fixes while in the area:
* Made check all headers of the requested name.
* Added some more sanity checks to the configured identify matching
options when applying the configuration.
ASTERISK-27548
Change-Id: I27dfd4ff5e2259b906640e3c330681b76b4ed1f1
The alembic for the PJSIP subscription persistence table has the
"prune_on_boot" field as a boolean. While in Asterisk we are
tolerant of many different definitions of true and false in the
database we only accept "yes" and "no". This change makes the
field treated as a yes/no instead of an integer, thus storing
"yes" and "no" instead of "1" and "0".
Change-Id: Ic8b9211b36babefe78f70def6828a135a6ae7ab6
OpenSSL is an optional external library and should stay optional even when
Developer Mode is configured.
ASTERISK-27990
Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b
If a SIP MESSAGE is triggered for an endpoint that is currently not registered
- and therefore has no valid contact associated - an error message was logged.
Since this is a valid request in a valid use cases this is now changed to a
warning, as discussed with Matt Fredrickson on the asterisk-dev mailing list.
Change-Id: I55eb62d2712818a58c7532119dec288bd98cf0c0
When realtime text packets are to be sent, the text is accumulated in a
buffer and sent regularly by a timer. It can happen that commands such as
a backspace, CR, or LF get merged with regular text. This breaks some
UAs.
The proposed change:
* We test if the current packet contains a command. If so we send the
buffer immediately.
* We test if the buffer contained a command. If so we send the buffer
immediately.
* We accumulate the text (or the command) in the buffer.
ASTERISK-27970
Change-Id: Ifbe993311410fa855cb8aa4a12084db75f413462
A change recently went in which disabled the built-in PJSIP
keepalive. This defaulted to 90 seconds and kept TCP/TLS
connections alive. Disabling this functionality has resulted
in a behavior change of not doing keepalives by default resulting
in TCP/TLS connections dropping for some people.
This change makes our default keepalive interval 90 seconds
to match the previous behavior and preserve it.
ASTERISK-27978
Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6
Keep track if ICE candidates were in the SDP offer & only put them
in the corresponding SDP answer if the offer condaind ICE candidates
ASTERISK-27957 #close
Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92
A problem I've seen countless times is a global or system section
for PJSIP not getting applied. This is inevitably the result of
the "type=" line missing. This change alleviates that problem.
The ability to specify an explicit section name has been
added to res_sorcery_config. If the configured section
name matches this and there are no unknown things configured
the section is taken as being for the given type.
Both the PJSIP "global" and "system" types now support this
so you can just name your section "global" or "system" and it
will be matched and used, even without a "type=" line.
ASTERISK-27972
Change-Id: Ie22723663c1ddd24f869af8c9b4c1b59e2476893
The function pubsub_on_rx_publish_request incorrectly uses
of AST_SCHED_REPLACE_UNREF.
The AST_SCHED_REPLACE_UNREF should unref old '_data'.
Because of this, there may be a double unref
of variable 'publication' when ast_sched_del is unsuccessful
that leads to use after free of the 'publication' in publish_expire.
ASTERISK-27956 #close
Change-Id: Ie0f0cfc7e036953d890b188656010b325a5cdc82
A new option 'suppress_q850_reason_headers' has been added to the
endpoint object. Some devices can't accept multiple Reason headers and
get confused when both 'SIP' and 'Q.850' Reason headers are received.
This option allows the 'Q.850' Reason header to be suppressed.
The default value is 'no'.
ASTERISK-27949
Reported-by: Ross Beer
Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1
When negotiating an incoming T.38 stream the code incorrectly
returned failure instead of a decline for the stream when a
problem occurred or the configuration didn't allow it. This
resulted in SDP offers being rejected with a 488 response
in all cases, even when another valid stream was present.
This change makes it so the stream is now declined. If no
streams are accepted a 488 response is sent while if at least
one stream is accepted all the declined streams are, well,
declined.
ASTERISK-27763
Change-Id: I88bcf793788c412a9839d111a5c736bf6867807c
We were blindly responding with AST_T38_REFUSED when ANY T.38 control
frame came accross the bridge. This causes T.38 Gateway to get confused
and the T.38 session to get in a strange state.
* Made the T.38 framehook only respond to request frames and ignore
response frames.
ASTERISK-27657
ASTERISK-27080
Change-Id: I5fb5967c7d1efb30a7ff375f82887ca82a55b05b
Using the keep_alive_interval option can result in a deadlock between the
pjproject transport manager group lock and the monitored transports ao2
container lock. The pjproject transport manager group lock has to be
superior in the locking order to the monitored transports ao2 container
lock because of pjproject callbacks called when already holding the group
lock. The lock inversion happens when Asterisk attempts to send a keep
alive packet over the reliable transports.
* Made keepalive_transport_thread() iterate over the monitored transports
container rather than use the ao2_callback() method. This avoids holding
the container lock when sending the keep alive packet.
ASTERISK-26686
Change-Id: I5d5392a52e698bbe41a93f7d8e92bf0e61fe3951
The Websocket transport uses the built-in HTTP server. As a result
the TLS configuration is done in http.conf and not in pjsip.conf.
This change adds a warning if this is configured in pjsip.conf and
also clarifies in the sample configuration file.
Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9
If we initiated a T.38 reINVITE, we would crash if we received any other
1xx response message except 100 if it were followed by a 200 response.
* Made ignore any 1xx response so we do not close out the T.38 negotiation
too early. For good measure we'll now accept any 2xx response as
acceptance of the reINVITE T.38 offer.
NOTE: In v13 we don't actually crash but we should deal with the non-final
response messages and not switch to rejected and then to enabled when the
200 comes in.
ASTERISK-27944
Change-Id: I0ca88aae708d091db7335af73f41035a212adff4
Incoming publications need to ensure that the module remains
loaded for the lifetime of them. This is now done by holding
a reference to the module while the publication exists. This
mirrors that of inbound subscriptions.
ASTERISK-27783
Change-Id: Ia98c95a15e11af25728d5fb3e56e12cda0cfc7c0
In addition to text/* content types, incoming_in_dialog_request now
accepts application/* content types.
Also fixed a length issue when copying the body text. It was one
character short.
ASTERISK-27942
Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818
pjproject by default currently will follow media forked during an INVITE
on outbound calls if the To tag is different on a subsequent response as
that on an earlier response. We handle this correctly. There have
been reported cases where the To tag is the same but we still need to
follow the media. The pjproject patch in this commit adds the
capability to sip_inv and also adds the capability to control it at
runtime. The original "different tag" behavior was always controllable
at runtime but we never did anything with it and left it to default to
TRUE.
So, along with the pjproject patch, this commit adds options to both the
system and endpoint objects to control the two behaviors, and a small
logic change to session_inv_on_media_update in res_pjsip_session to
control the behavior at the endpoint level.
The default behavior for "different tags" remains the same at TRUE and
the default for "same tag" is FALSE.
Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6
ASTERISK-27936
Reported-by: Ross Beer
* Fix several instances where we were bumping a ref in the parameter and
then unrefing the object if it failed. The way the AST_VECTOR_APPEND()
and AST_VECTOR_REPLACE() macros are implemented means if it fails the new
value was never evaluated.
Change-Id: I2847872a455b11ea7e5b7ce697c0a455a1d0ac9a
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e
Previously, Asterisk used its script ./configure, to test whether OpenSSL was
built with no-srtp (or was simply too old). However, the header file
<openssl/opensslconf.h> is the preferred way to detect the local configuration
of OpenSSL.
As a positive side-effect the script ./configure does not interleave the
detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of
individual features of OpenSSL anymore.
Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0
When endpoint specific ACL rules block a SIP request they respond with a
403 forbidden. However, if an endpoint is not identified then a 401
unauthorized response is sent. This vulnerability just discloses which
requests hit a defined endpoint. The ACL rules cannot be bypassed to gain
access to the disclosed endpoints.
* Made endpoint specific ACL rules now respond with a 401 unauthorized
which is the same as if an endpoint were not identified. The fix is
accomplished by replacing the found endpoint with the artificial endpoint
which always fails authentication.
ASTERISK-27818
Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32
Furthermore, allow OpenSSL configured with no-dh. Additionally, this change
allows auto-negotiation of the elliptic curve/group for servers, not only with
OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519
(since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect.
ASTERISK-27910
Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537
Currentrly pjsip_options code does not handle the situation when the
qualify options were changed in realtime database.
Only 'module reload res_pjsip' helps.
This patch add a check on contact add/update observers if the contact
qualify options are different than local aor qualify options.
If the qualify options were modified then synchronize
the pjsip_options AOR local state.
ASTERISK-27872
Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62
Certain race conditions between changing bridge types and DTMF can
cause the current FLAG_NEED_MARKER_BIT to send the marker bit before
the actual first packet of native bridging.
This logic keeps track of the ssrc the bridge is currently sending
and will correctly ensure the marker bit is set if SSRC as changed
from the previous sent packet.
ASTERISK-27845
Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b
Currentrly pjsip_options code does not handle the situation when the
AOR qualify options were changed.
Also there is no way to find out what qualify options are using.
This patch add CLI commands to show and synchronize Aor qualify options:
pjsip show qualify endpoint <id>
Show the current qualify options for all Aors on the PJSIP endpoint.
pjsip show qualify aor <id>
Show the PJSIP Aor current qualify options.
pjsip reload qualify endpoint <id>
Synchronize the qualify options for all Aors on the PJSIP endpoint.
pjsip reload qualify aor <id>
Synchronize the PJSIP Aor qualify options.
ASTERISK-27872
Change-Id: I1746d10ef2b7954f2293f2e606cdd7428068c38c
Asterisk uses Reference Counting to track whether a module can be unloaded.
Every consumer who requires a module, increases the reference count. When the
consumer goes, is unloaded itself, it has to decrease the reference count on
all its used/required modules. That way
core stop gracefully
works on the command-line interface (CLI): One module after the other is
unloaded. A recent change broke this for the module res_pjsip.
ASTERISK-27861
Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3
The module (res_)pjsip_transport_management got moved into res_pjsip. It is no
longer an independent/external module with (un)load_module and therefore has to
register just internally with res_pjsip.
ASTERISK-27860
Change-Id: Icd0413be7d2e98b92f51e6d6c353f2570bb4be95
This fixes build warnings found by GCC 8. In some cases format
truncation is intentional so the warning is just suppressed.
ASTERISK-27824 #close
Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84
Previously, only an IP address would be accepted for the capture_address config
setting in hep.conf. This change allows capture_address to be a resolvable
hostname or an IP address.
ASTERISK-27796 #close
Reported-By: Sebastian Gutierrez
Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f
The "ari set debug" code for incoming requests incorrectly assumed
that all requests would contain a body. If one did not exist the
request would be incorrectly rejected. The response that was sent
was also incomplete as an incorrect function was used to construct
the response.
The code has now been changed to no longer require a request to have
a body and the response updated to use the correct function.
ASTERISK-27801
Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8
* Increase maximum number of ciphers from 100 to 256 (or whatever
PJ_SSL_SOCK_MAX_CIPHERS is #define'd to)
* Simplify logic in cipher_name_to_id()
* Make signed/unsigned comparison consistent
Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412
Reported by: Ondřej Holas
Change-Id: Iea620f03915a1b873e79743154255c3148a514e7
The OPTIONS support in PJSIP has organically grown, like many things in
Asterisk. It has been tweaked, changed, and adapted based on situations
run into. Unfortunately this has taken its toll. Configuration file
based objects have poor performance and even dynamic ones aren't that
great.
This change scraps the existing code and starts fresh with new eyes. It
leverages all of the APIs made available such as sorcery observers and
serializers to provide a better implementation.
1. The state of contacts, AORs, and endpoints relevant to the qualify
process is maintained. This state can be updated by external forces (such
as a device registering/unregistering) and also the reload process. This
state also includes the association between endpoints and AORs.
2. AORs are scheduled and not contacts. This reduces the amount of work
spent juggling scheduled items.
3. Manipulation of which AORs are being qualified and the endpoint states
all occur within a serializer to reduce the conflict that can occur with
multiple threads attempting to modify things.
4. Operations regarding an AOR use a serializer specific to that AOR.
5. AORs and endpoint state act as state compositors. They take input
from lower level objects (contacts feed AORs, AORs feed endpoint state)
and determine if a sufficient enough change has occurred to be fed further
up the chain.
6. Realtime is supported by using observers to know when a contact has
been registered. If state does not exist for the associated AOR then it
is retrieved and becomes active as appropriate.
The end result of all of this is best shown with a configuration file of
3000 endpoints each with an AOR that has a static contact. In the old
code it would take over a minute to load and use all 8 of my cores. This
new code takes 2-3 seconds and barely touches the CPU even while dealing
with all of the OPTIONS requests.
ASTERISK-26806
Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082
Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge. res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.
res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame. On a normal
point-to-point call, the frames are forwarded between the two
correctly. bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants. Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.
* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload. A channel
driver can queue a frame of that type when it receives a message
from outside. A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties. If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this. Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.
* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel. This allows the chat client user to set a friendly name
for the chat.
* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).
* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.
* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.
* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.
Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489
This reverts a problem introduced by the fix for ASTERISK_24329.
Now, when an announcement is played while waiting in a queue, music on
hold will not restart from the beginning of the sound file and will
instead pick up where it left off. However, the incorrect behavior in
ASTERISK_24329 is now present again; if an announcement X seconds
long is played when music on hold starts, music on hold will start X
seconds into the file.
ASTERISK-27774 #close
Reported by: lvl
Change-Id: I86b2885ee7063268f9b9747eddb788336ade989b
When a scheduled task is created you can pass in the
AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to
be logged.
Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b
Jenkins2
Corey Farrell
Joshua Colp
George Joseph
Torrey Searle
Alexei Gradinari
Chris-Savinovich
Kevin Harwell
Richard Mudgett
Ben Ford
Moritz Fain
Sean Bright
Walter Doekes
Frederic LE FOLL
neutrino88
Alexander Traud
Florian Floimair
Matthew Fredrickson