Adds support for custom URI and header parameters
in the From header in PJSIP. Parameters can be
both set and read using this function.
ASTERISK-30150 #close
Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428
chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).
This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.
* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)
The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.
The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.
Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.
ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>
Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333
Some SIP devices use an empty extension for PLAR functionality.
Rather than rejecting these empty extensions, we now use the s
extension for such calls to mirror the existing PLAR functionality
in Asterisk (e.g. chan_dahdi).
ASTERISK-30265 #close
Change-Id: I0861a405cd49bbbf532b52f7b47f0e2810832590
This fixes a small typo in the from_domain documentation on the endpoint documentation
ASTERISK-30328 #close
Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21
Adds support for the capture agent name field
of the Homer protocol to Asterisk by allowing
users to specify a name that will be sent to
the HEP server.
ASTERISK-30322 #close
Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b
msg_create_from_file currently does not dispatch emails,
which means that applications using this function, such
as MixMonitor, will not trigger notifications to users
(only AMI events are sent our currently). This is inconsistent
with other ways users can receive voicemail.
This is fixed by adding an option that attempts to send
an email and falling back to just the notifications as
done now if that fails. The existing behavior remains
the default.
ASTERISK-30283 #close
Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7
Adds the If, ElseIf, Else, ExitIf, and EndIf
applications for conditional execution
of a block of dialplan, similar to the While,
EndWhile, and ExitWhile applications. The
appropriate branch is executed at most once
if available and may be broken out of while
inside.
ASTERISK-29497
Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49
Updates the documentation for the 'contact_user' field to point out the
default outbound contact if no contact_user is specified 's'
ASTERISK-30316 #close
Change-Id: I61f24fb9164e4d07e05908a2511805281874c876
The commit that rearchitected media formats,
a2c912e997 (ASTERISK_23114)
introduced a regression by improperly translating code in res_adsi.c.
In particular, the pointer to the frame buffer was initialized
at the top of adsi_careful_send, rather than dynamically updating it
for each frame, as is required.
This resulted in the first frame being repeatedly sent,
rather than advancing through the frames.
This corrupted the transmission of the CAS to the CPE,
which meant that CPE would never respond with the DTMF acknowledgment,
effectively completely breaking ADSI functionality.
This issue is now fixed, and ADSI now works properly again.
ASTERISK-29793 #close
Change-Id: Icdeddf733eda2981c98712d1ac9cddc0db507dbe
When passing a JSON body to the 'create' channel route
it would be converted into Asterisk variables, but never
freed resulting in a memory leak.
This change makes it so that the variables are freed in
all cases.
ASTERISK-30344
Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d
When parsing information from AstDB while loading,
it is possible that certain pointers are never
set, which leads to invalid memory access and
then, fatally, invalid free attempts on this memory.
We now initialize to NULL to prevent this.
ASTERISK-30311 #close
Change-Id: I6120681d04fd2c12a9473f35ce95a1f8e74e3929
ASTERISK_28702 previously attempted to fix an
issue with flash hook hold timing out after
just under 17 minutes, when it should have never
been timing out. It fixed this by changing 999999
to INT_MAX, but it did so in chan_dahdi, which
is the wrong place since ss_thread is now in
sig_analog and the one in chan_dahdi is mostly
dead code.
This fixes this by porting the fix to sig_analog.
ASTERISK-30336 #close
Change-Id: I05eb69cc0b5319d357842a70bd26ef64d145cb15
The XML docs are currently only loaded on
startup with no way to update them during runtime.
This makes it impossible to load modules that
use ACO/Sorcery (which require documentation)
if they are added to the source tree and built while
Asterisk is running (e.g. external modules).
This adds a CLI command to reload the XML docs
during runtime so that documentation can be updated
without a full restart of Asterisk.
ASTERISK-30289 #close
Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07
This file includes some doxygen comments referencing
ast_format_set. This is an obsolete API that was
removed years back, but documentation was not fully
updated to reflect that. These examples are
updated to the current way of doing things
(using the format cache).
ASTERISK-30327 #close
Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7
MixMonitor currently uses the Connected Line as the Caller ID
for voicemails. This is due to the implementation being written
this way for use with Digium phones. However, in general this
is not correct for generic usage in the dialplan, and people
may need the real Caller ID instead. This adds an option to do that.
ASTERISK-30286 #close
Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c
It was possible for a module that registered for transport monitor
events to pass in a pjsip_transport that had already been freed.
This caused pjsip_transport_events to crash when looking up the
monitor for the transport. The fix is a two pronged approach.
1. We now increment the reference count on pjsip_transports when we
create monitors for them, then decrement the count when the
transport is going to be destroyed.
2. There are now APIs to register and unregister monitor callbacks
by "transport key" which is a string concatenation of the remote ip
address and port. This way the module needing to monitor the
transport doesn't have to hold on to the transport object itself to
unregister. It just has to save the transport_key.
* Added the pjsip_transport reference increment and decrement.
* Changed the internal transport monitor container key from the
transport->obj_name (which may not be unique anyway) to the
transport_key.
* Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that
fills a buffer with the transport_key using a passed-in
pjsip_transport.
* Added the following functions:
ast_sip_transport_monitor_register_key
ast_sip_transport_monitor_register_replace_key
ast_sip_transport_monitor_unregister_key
and marked their non-key counterparts as deprecated.
* Updated res_pjsip_pubsub and res_pjsip_outbound_register to use
the new "key" monitor functions.
NOTE: res_pjsip_registrar also uses the transport monitor
functionality but doesn't have a persistent object other than
contact to store a transport key. At this time, it continues to
use the non-key monitor functions.
ASTERISK-30244
Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b
Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.
ASTERISK-30176
Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404
When decoding q.931 encoded calling/called number
now checking for length being less than minimum required.
ASTERISK-30103
Change-Id: I3dcfce0f35eca258dc450f87c92d4d7af402c2e7
This PR contains two relatively separate changes in channel.c and
res_pjsip_session.c which ensure that topology changes are not ignored
in cases where they should be handled.
For channel.c:
The function ast_channel_request_stream_topology_change only triggers a
stream topology request change indication, if the channel's topology
does not equal the requested topology. However, a channel could be in a
state where it is currently "negotiating" a new topology but hasn't
updated it yet, so the topology request change would be lost. Channels
need to be able to handle such situations internally and stream
topology requests should therefore always be passed on.
In the case of chan_pjsip for example, it queues a session refresh
(re-INVITE) if it is currently in the middle of a transaction or has
pending requests (among other reasons).
Now, ast_channel_request_stream_topology_change always indicates a
stream topology request change even if the requested topology equals the
channel's topology.
For res_pjsip_session.c:
The function resolve_refresh_media_states does not process stream state
changes if the delayed active state differs from the current active
state. I.e. if the currently active stream state has changed between the
time the sip session refresh request was queued and the time it is being
processed, the session refresh is ignored. However, res_pjsip_session
contains logic that ensures that session refreshes are queued and
re-queued correctly if a session refresh is currently not possible. So
this check is not necessary and led to some session refreshes being
lost.
Now, a session refresh is done even if the delayed active state differs
from the current active state and it is checked whether the delayed
pending state differs from the current active - because that means a
refresh is necessary.
Further, the unit test of resolve_refresh_media_states was adapted to
reflect the new behavior. I.e. the changes to delayed pending are
prioritized over the changes to current active because we want to
preserve the original intention of the pending state.
ASTERISK-30184
Change-Id: Icd0703295271089057717006730b555b9a1d4e5a
The Answer application currently waits for up to 500ms
for media, even if users specify a different timeout.
This adds an option to not wait for media on the channel
by doing a raw answer instead. The default 500ms threshold
is also documented.
ASTERISK-30308 #close
Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4
Currently, chan_dahdi will wait for at least one
ring before an incoming call can enter the dialplan.
This is generally necessary in order to receive
the Caller ID spill and/or distinctive ringing
detection.
However, if neither of these is required, then there
is nothing gained by waiting for one ring and this
unnecessarily delays call setup. Users can now
use immediate=yes to make FXO channels (FXS signaled)
begin processing dialplan as soon as Asterisk receives
the call.
ASTERISK-30305 #close
Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5
SLAStation currently autoservices the station channel before
creating a thread to actually dial the trunk. This leads
to duplicate servicing of the channel which causes assertions,
deadlocks, crashes, and moreover not the correct behavior.
Removing the autoservice prevents the crash, but if the station
hangs up before the trunk answers, the call hangs since the hangup
was never serviced on the channel.
This is fixed by not autoservicing the channel, but instead
servicing it in the thread dialing the trunk, since it is doing
so synchronously to begin with. Instead of sleeping for 100ms
in a loop, we simply use the channel for timing, and abort
if it disappears.
The same issue also occurs with SLATrunk when a call is answered,
because ast_answer invokes ast_waitfor_nandfds. Thus, we use
ast_raw_answer instead which does not cause any conflict and allows
the call to be answered normally without thread blocking issues.
ASTERISK-29998 #close
Change-Id: Icc237d50354b5910000d2305901e86d2c87bb9d8
Found in res_geolocation, but I believe others may have similar issues,
thus not linking to a specific issue.
Essentially gcc doesn't mark the stack for being non-executable unless
it's compiling the source, this informs ld via gcc to mark the object as
not requiring an executable stack (which a binary blob obviously
doesn't).
ASTERISK-30321
Change-Id: I71bcc2fd1fe0c82a28b3257405d6f2b566fd9bfc
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
A memory leak was present in func_json due to
using ast_json_free, which just calls ast_free,
as opposed to recursively freeing the JSON
object as needed. This is now fixed to use the
right free functions.
ASTERISK-30293 #close
Change-Id: I982324dde841dc9147c8d8ad35c8719daf418b49
Removes the function mkstemp_file and uses
ast_file_mkftemp from file.h instead.
ASTERISK-30295 #close
Change-Id: I7412ec06f88c39ee353bcdb8c976c2fcac546609
The "RECORD FILE" command in res_agi has its own
implementation for actually doing the recording. This
has resulted in it not actually obeying the option
"transmit_silence" when recording.
This change causes it to now send silence if the
option is enabled.
ASTERISK-30314
Change-Id: Ib3a85601ff35d1b904f836691bad8a4b7e957174
The ModuleCheck XML documentation falsely
claims that the module's version number is returned.
This has not been the case since 14, since the version
number is not available anymore, but the documentation
was not changed at the time. It is now updated to
reflect this.
ASTERISK-30285 #close
Change-Id: Idde2d1205a11f2623fa1ddab192faa3dc4081e91
Adds an option that allows MixMonitor to delete
its copy of any recording files before exiting.
This can be handy in conjunction with options
like m, which copy the file elsewhere, and the
original files may no longer be needed.
ASTERISK-30284 #close
Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e
When gosub is executed on channels without a PBX, the context,
extension, and priority are initialized to the channel driver's
default location for that endpoint. As a result, the last Return
will restore this location and the Gosub logs will print out bogus
information about our exit point.
To fix this, on channels that don't have a PBX, the execution
location is left intact on the last return if there are no
further stack frames left. This allows the correct location
to be printed out to the user, rather than the bogus default
context.
ASTERISK-30076 #close
Change-Id: I1d42a99c9aa9e3708d32718863175158a894e414
(cherry picked from commit 12d18b0a40)
Fixed the specification of "outputdir" when calling ast_coredumper
so the txt files are saved in the correct place.
ASTERISK-30282
Change-Id: Ic631cb90c1e4c29d970c982dff45fda5e0eb15b6
unicast_rtp_request() was setting the channel variables like this:
pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
ast_sockaddr_stringify_addr(&local_address));
ast_rtp_instance_get_local_address(instance, &local_address);
pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
ast_sockaddr_stringify_port(&local_address));
...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being
set before local_address was set. In fact, the address part of
local_address was set earlier in the function, just not the port.
This was confusing however so ast_rtp_instance_get_local_address()
is now being called before setting UNICASTRTP_LOCAL_ADDRESS.
ASTERISK-30281
Change-Id: I872ac49477100f4eb33891d46efc6ca21ec81aa4
When a websocket (or potentially any stateful connection) is quickly
created then destroyed, it is possible that the qualify thread will
destroy the transaction before the initialzing thread is finished
with it.
Depending on the timing, this can cause an assertion within pjsip.
To prevent this, ast_send_stateful_response will now create the group
lock and add a reference to it before creating the transaction.
While this should resolve the crash, there is still the potential that
the contact will not be cleaned up properly, see:ASTERISK~29286. As a
result, the contact has to 'time out' before it will be removed.
ASTERISK-28689
Change-Id: Id050fded2247a04d8f0fc5b8a2cf3e5482cb8cee
Current registration code use pjsip_parse_uri to verify outbound_proxy
that is different from the reading this option for the endpoint. This
made value with multiple proxies invalid for registration pjsip settings.
Removing URI validation helps to use registration through multiple proxies.
ASTERISK-30217 #close
Change-Id: I064558e66f04b9f3260c46181812a01349761357
write_openssl_error_to_log has been erroneously
using ast_free instead of free, which will
cause a crash when MALLOC_DEBUG is enabled since
the memory was not allocated by Asterisk's memory
manager. This changes it to use the actual free
function directly to avoid this.
ASTERISK-30278 #close
Change-Id: Iac8b6468b718075809c45d8ad16b101af21a474d
Fix compilation errors caused by using size_t
instead of uintmax_t and non-portable format
specifiers.
ASTERISK-30273 #close
Change-Id: I363e6057ef84d54b88af80d23ad6147eef9216ee
Currently chan_pjsip on receiving a re-INVITE without SDP will only
return the codecs that are previously negotiated and not offering
all enabled codecs.
This causes interoperability issues with different equipment (e.g.
from Cisco) for some of our customers and probably also in other
scenarios involving 3PCC infrastructure.
According to RFC 3261, section 14.2 we SHOULD return all codecs
on a re-INVITE without SDP
The PR proposes a new parameter to configure this behaviour:
all_codecs_on_empty_reinvite. It includes the code, documentation,
alembic migrations, CHANGES file and example configuration additions.
ASTERISK-30193 #close
Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148
Expands the pjsip logger to support the ability to filter
by SIP message method. This can make certain types of SIP debugging
easier by only logging messages of particular method(s).
ASTERISK-30146 #close
Co-authored-by: Sean Bright <sean@seanbright.com>
Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f
The PJSIP notify CLI commands allow for using
"options" configured in pjsip_notify.conf.
This allows these same options to be used in
AMI actions as well.
Additionally, as part of this improvement,
some repetitive common code is refactored.
ASTERISK-30263 #close
Change-Id: Ie4496b322b63b61eaf9672183a959ab99a04b6b5
race condition: ast_dial_join() may not cancel outgoing call, if
function is called just after called party answer and before
application execution (bit is_running_app not yet set).
This fix adds ast_softhangup() calls in addition to existing
pthread_kill() when is_running_app is not set.
ASTERISK-30258
Change-Id: Idbdd5c15122159661aa8e996a42d5800083131e4
This fixes dahdi_request to properly set the cause
code to CONGESTION instead of BUSY if no channels
were actually available.
Currently, the cause is erroneously set to busy
if the channel itself is found, regardless of its
current state. However, if the channel is not available
(e.g. T1 down, card not operable, etc.), then the
channel itself may not be in a functional state,
in which case CHANUNAVAIL is the correct cause to use.
This adds a simple check to ensure that busy tone
is only returned if a channel is encountered that
has an owner, since that is the only possible way
that a channel could actually be busy.
ASTERISK-30274 #close
Change-Id: Iad5870223c081240c925b19df8d6af136953b994
Some logic in say.c for determining if we need
to also add an ampersand for file seperation was faulty,
as non-successful files would increment the count, causing
a leading ampersand to be added improperly.
This is fixed, and a unit test that captures this regression
is also added.
ASTERISK-30248 #close
Change-Id: I02c1d3a11d82fe4ea8b462070cbd1effb5834d2b
pjproject does not provide any mechanism of removing
event packages, which means that once a subscription
handler is registered, it is effectively permanent.
pjproject will assert if the same event package is
ever registered again, so currently unloading and
loading any Asterisk modules that use subscriptions
will cause a crash that is beyond our control.
For that reason, we now prevent users from being
able to unload these modules, to prevent them
from ever being loaded twice.
ASTERISK-30264 #close
Change-Id: I7fdcb1a5e44d38b7ba10c44259fe98f0ae9bc12c
Add enum to allow setting optional direction. If set to only one
direction, only feed matching-direction frames to the associated
slin factory.
This prevents mangling the transcoder on non-mixed frames when the
READ and WRITE frames would have otherwise required it. Also
removes the need to mute or discard the un-wanted frames as they
are no longer added in the first place.
res_stasis_snoop is changed to use this addition to set direction
on audiohook based on spy direction.
If no direction is set, the ast_audiohook_init will init this enum
to BOTH which maintains existing functionality.
ASTERISK-30252
Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb
Adds support for detecting audible ringback tone
to the TONE_DETECT function using the p option.
ASTERISK-30254 #close
Change-Id: Ie2329ff245248768367d26749c285fbe823f6414
Naveen Albert
Michael Kuron
Marcel Wagner
Joshua C. Colp
Ben Ford
George Joseph
Mike Bradeen
Maximilian Fridrich
Jaco Kroon
Igor Goncharovsky
Henning Westerholt
Frederic LE FOLL
Philip Prindeville