Since ASTERISK-26922, this issue affected only those chan_sip which were
* enabled for dual-stack (bindaddr=::), and
* enabled for TCP (tcpenable=yes) and/or TLS (tlsenable=yes), and
* tried to register and/or invite a IPv4-only service,
* via TCP and/or TLS.
Now, ast_tcptls_client_create does not re-bind to [::] anymore.
ASTERISK-27324 #close
Change-Id: I4b242837bdeb1ec7130dc82505c6180a946fd9b5
At some point in time in the history of Corosync (certainly within the
2.x branch), the corosync_cfg_state_track function was removed.
Unfortunately, the cfg library is only linked if this function is
present. Without the cfg library being linked to res_corosync, loading
of res_corosync will fail.
This patch makes it so that detecting corosync's core libraries,
determined by the COROSYNC external library checks, links both the cpg
and cfg libraries with res_corosync.
Change-Id: I674e9e1c8fea11c3bf81154aaa7c1fd43f945465
Do not manually call sip_endpoint_apply_handler from load_all_endpoints.
This is not necessary and causes memory leaks.
Additionally reinitialize persistent->aors when we reuse a persistent
object with a new endpoint.
ASTERISK-27306
Change-Id: I59bbfc8da8a14d5f4af8c5bb1e71f8592ae823eb
Use temporary variable to prevent multiple evaluations of elem argument.
This resolves a memory leak in res_pjproject startup.
ASTERISK-27317 #close
Change-Id: Ib960d7f5576f9e1a3c478ecb48995582a574e06d
pjsip_distributor leaks references to fake_auth when the default realm
has not changed.
ASTERISK-27306
Change-Id: I3fcf103b3680ad2d1d4610dcd6738eeaebf4d202
ast_strings_match uses sscanf and checks for non-zero return to verify a
token was parsed. This is incorrect as sscanf returns EOF (-1) for errors.
ASTERISK-27318 #close
Change-Id: Ifcece92605f58116eff24c5a0a3b0ee08b3c87b1
Currently privacy requests are only granted if the Privacy header
value is exactly "id" (defined in RFC 3325). It ignores any other
possible value (or a combination there of). This patch reverses the
logic from testing for "id" to grant privacy, to testing for "none" and
granting privacy for any other value. "none" must not be used in
combination with any other value (RFC 3323 section 4.2).
ASTERISK-27284 #close
Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56
This provides better information to REF_DEBUG log for troubleshooting
when the system is unable to unload res_pjsip.so during shutdown due to
module references.
ASTERISK-27306
Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612
res_pjsip and res_pjsip_session had circular references, preventing both
modules from shutting down.
* Move session supplement registration to res_pjsip.
* Use create internal functions for use by pjsip_message_filter.c.
ASTERISK-27306
Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b
When we are loading the calendars, we call libical's
icalcomponent_foreach_recurrence method for each VEVENT component that
we have in our calendar.
That method has no knowledge concerning the existence of the other
VEVENT components and will feed our callback with all ocurrences
matching the requested time span.
The occurrences generated by icalcomponent_foreach_recurrence while
expanding a recurring VEVENT's RRULE and RDATE properties can be
superceded by an other VEVENT sharing the same UID.
I use an external iterator (in libical terminology) to avoid messing
with the internal ones from the calling function, and search for
VEVENTS which could supersede the current occurrence.
The event which can invalidate this occurence needs to have:
- the same UID as our recurrent component (comp)
- a RECURRENCE-ID property, which represents the start time of this
occurrence
If one component is found, just clean and return.
ASTERISK-27296 #close
Reported by: Benoît Dereck-Tricot
Change-Id: I8587ae3eaa765af7cb21eda3b6bf84e8a1c87af8
The previous patch for ASTERISK-27216 made it so you wouldn't get any
position or periodic announcements unless you had announce-to-first-user
enabled. The announce-to-first-user feature was added by ASTERISK_21782
as a result of the patch which introduced the redundant announcements that
ASTERISK-27216 removes.
* By noting that the makeannouncement variable is used to suppresses the
first user announcement, we set its initial value to the
announce-to-first-user enable setting.
ASTERISK-27216
Change-Id: Ieaeb7dbea8ae7073086b775fbafe0625b000b10a
Somewhere along the way we lost the ability to debug individual
source files. For modules, this wasn't a big deal but all the
source files in ./main are in the one "core" module so debugging
individual core capabilities was almost impossible.
* Added a test to DEBUG_ATLEAST that also checks __FILE__ instead
of just module name. Any source file will work even if it's in
a module subdirectory.
Change-Id: Icc0af41837f3b1679dec7af21fa32cd1f7469f6e
This change adds 'video_sfu' as a requested bridge type when
creating a bridge. By specifying this a mixing type bridge is
created that exchanges video in an SFU fashion.
Change-Id: I2ada47cf5f3fc176518b647c0b4aa39d55339606
The pjsip_publishc_init() call was referenced with a misplaced
parentheses. As a result, outbound publication messages went out with an
expiration of 1 second.
ASTERISK-27298
Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08
The "res_pjsip: Filter out non SIP(S) requests" commit moved the
filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER
in order to filter out incoming bad uri schemes as early as possible.
Since the change affected outgoing messages as well and the TRANSPORT
layer is the last to be run on outgoing messages, we were overwriting
the setting of external_signaling_address (which is set earlier by
res_pjsip_nat) with an internal address.
* pjsip_message_filter now registers itself as a pjproject module
twice. Once in the TSX layer for the outgoing messages (as it was
originally), then a second time in the TRANSPORT layer for the
incoming messages to catch the invalid uri schemes.
ASTERISK-27295
Reported by: Sean Bright
Change-Id: I2c90190c43370f8a9d1c4693a19fd65840689c8c
The bridge_p2p_rtp_write() has potential reentrancy problems.
* Accessing the bridged RTP members must be done with the instance1 lock
held. The DTMF and asymmetric codec checks must be split to be done with
the correct RTP instance struct locked. i.e., They must be done when
working on the appropriate side of the point to point bridge.
* Forcing the RTP mark bit was referencing the wrong side of the point to
point bridge. The set mark bit is used everywhere else to set the mark
bit when sending not receiving.
The patches for ASTERISK_26745 and ASTERISK_27158 did not take into
account that not everything carried by RTP uses a codec. The telephony
DTMF events are not exchanged with a codec. As a result when
RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is
enabled, the DTMF digits would always get passed to the core even though
the local native RTP bridge is active, and the DTMF digits would go out
using the wrong SSRC id.
* Add protection for non-format payload types like DTMF when updating the
lastrxformat and lasttxformat. Also protect against non-format payload
types when checking for asymmetric codecs.
ASTERISK-27292
Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186
This could have been fixed by subtracting 1 from the final value of
'len' but the way the packet was being constructed was confusing so I
took the opportunity to (I think) make it more clear.
We were sending 1 extra byte at the end of the SDES RTCP packet which
caused Chrome to complain (in its debug log):
Too little data (1 byte) remaining in buffer to parse
RTCP header (4 bytes).
We now send the correct number of bytes.
Change-Id: I9dcf087cdaf97da0374ae0acb7d379746a71e81b
If using a legitimate certificate from a trusted certificate authority,
you don't need to provide CA file.
Change-Id: I8623973b4209b44889243716d7880274caed8a6d
For some scenarios when an outgoing call was made only a subset of the
configured codecs were offered. If the codecs being offered happened to
not have a codec supported by the phone then the call would fail.
For instance Alice and Bob both are configured in Asterisk for g722 and ulaw(
allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's
only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with
Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains
g722 and not both g722 and ulaw, so the call ends.
This patch makes it so all the audio codecs configured on the endpoint always
get sent, and not just a subset. However priority is given to those codecs that
are compatible with the "other side".
ASTERISK-27259 #close
Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696
This patch reverts the change by patch 2263 from old reviewboard.
Note that reverting that 2263-patch still preserves the behaviour that
the commit log of the 2263-patch claimed to add. The reason for this is:
The function wait_for_answer is only called from try_calling which
in turn is only called from the main for loop in queue_exec, and
earlier in that loop we already check the things that's removed by
this patch. There's no need to check those things twice each loop
iteration, and I think the proper place to check it is before each
ringing cycle. By checking it in wait_for_answer, you allow the issue
explained in the jira - that the head caller hears announcements while
the agents' sip phones are actively ringing.
Reported-by: Stefan Engström
Tested-by: Stefan Engström
ASTERISK-27216 #close
Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0
Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic
Edition after accepting the audio request but declining the video one.
Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c
When pruning a request to change the topology of a channel be
more intelligent about the resulting topology that is actually
used for SDP renegotiation.
In a case where a stream has not already been negotiated we
don't need to renegotiate and offer a declined stream. This can
occur if something in Asterisk (such as ConfBridge) requests
to add video to a PJSIP channel that has no video codecs configured.
In this case since the stream did not already exist we can safely
remove the stream from the requested topology, resulting in no
renegotiation occurring.
In a case where a renegotiation is requested with a codec that is
not supported we can reuse the formats of the existing stream if
it exists to ensure that the stream continues to flow, instead of
removing it.
Change-Id: I636540798d55922377318fe619c510fb6ed125fb
