Several modules removal and deprecations occurred in 19.0.0 (initial
19 release), but associated UPGRADE files were not removed from
staging for some reason in the master branch.
This patch removes those files, and also removes a spurious leftover
header, chan_phone.h (associated module removed in 19).
Change-Id: Ib92142c846b45c882d6b2b6caca7225253c83add
The iax2 show netstats command previously didn't contain
enough spacing in the header to properly align the table
header with the table body. This caused column headers
to not align with the values on longer channel names.
Some spacing is added to account for the longest channel
names that display (before truncation occurs) so that
columns are always properly aligned.
ASTERISK-29895 #close
patches:
61205_misaligned2.patch submitted by Birger Harzenetter (license 5870)
Change-Id: I450ce6bb81157b9d6d149007e53b749f237b6d9f
ASTERISK_22025 introduced a regression that shows
the host IP and port as the perceived IP and port
again, as opposed to showing the actual perceived
address. This fixes this by showing the correct
information.
ASTERISK-29048 #close
Change-Id: I0ad3e25bc6b449e83ce72ea5d1a1cdba72aa304a
Change RTP timer behavior for detecting RTP only after two-way
SDP channel establishment. Ignore detecting after receiving 183
with SDP or while direct media is used.
Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout
and rtpholdtimeout options in chan_sip.
ASTERISK-26689 #close
ASTERISK-29929 #close
Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb
gethostbyname() and gethostbyname_r() are deprecated in favor of
getaddrinfo() which we use in the ast_sockaddr family of functions.
ASTERISK-29819 #close
Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7
Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before
initiating an outgoing call will cause Asterisk to crash. This is
because a null field is accessed, resulting in an offset from null and
subsequent memory access violation.
Since RURI is not guaranteed to exist, we now check if the base
pointer is non-null before calculating an offset.
ASTERISK-29772
Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83
Since Doxygen 1.8.16, a special comment block is required. Otherwise
(pure C comment), the group command is ignored. Additionally, several
unbalanced group commands were fixed.
ASTERISK-29732
Change-Id: I4687857b9d56e6f44fd440b73af156691660202e
Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.
This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.
ASTERISK-29777 #close
Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b
Fixes compiler warning caused by a truncated copy of the ANI2 into a
buffer of size 10. This could prevent the null terminator from being
copied if the copy value exceeds the size of the buffer. This increases
the buffer size to 101 to ensure there is no way for truncation to occur.
ASTERISK-29702 #close
Change-Id: Ief9052212952840fa44de6463b8699fdb3e163d0
Historically, the dial syntax for IAX2 has held that
an outkey (used only for RSA authenticated calls)
and a secret (used only for plain text and MD5 authenticated
calls, historically) were mutually exclusive, and thus
the same position in the dial string was used for both
values.
Now that encryption is possible with RSA authentication,
this poses a limitation, since encryption requires a
secret and RSA authentication requires an outkey. Thus,
the dial syntax is extended so that both a secret and
an outkey can be specified.
The new extended syntax is backwards compatible with the
old syntax. However, a secret can now be specified after
the outkey, or the outkey can be specified after the secret.
This makes it possible to spawn an encrypted RSA authenticated
call without a corresponding peer being predefined in iax.conf.
ASTERISK-29707 #close
Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e
test_voicemail_api: Use empty char* for empty_msg_ids.
chan_skinny: Fix size of calledParty to be maximum extension.
menuselect: Change Makefile to stop deprecated warnings. Added comments
test_linkedlist: 'bogus' variable was manually allocated from a macro
and the test fails if this happens but the compiler couldn't 'see' this
and returns a warning. memset to all 0's after allocation.
chan_ooh323: Fixed various indentation issues that triggered misleading
indentation warnings.
ASTERISK-29682
Reported by: George Joseph
Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe
Adds support for encryption to RSA-authenticated
calls. Also prevents crashes if an RSA IAX2 call
is initiated to a switch requiring encryption
but no secret is provided.
ASTERISK-20219
Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40
Adds parsing of ANI II digits (Originating
Line Information) to PJSIP, on par with
what currently exists in chan_sip.
ASTERISK-29472
Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847
Adds an information element for ANI2 so that
Originating Line Information can be transmitted
over IAX2 channels.
ASTERISK-29605 #close
Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2
This changeset is intended to address compatibility issues encountered
when interfacing Asterisk to electromechanical telephone switches that
implement ANI-B, ANI-C, or ANI-D.
In particular the behaviours that this impacts include:
- FGC-CAMA did not work at all when using MF signaling. Modified the
switch case block to send calls to the correct part of the
signaling-handling state machine.
- For FGC-CAMA operation, the delay between called number ST and
second wink for ANI spill has been made configurable; previously
all calls were made to wait for one full second.
- After the ANI spill, previous behavior was to require a 'ST' tone
to advance the call. This has been changed to allow 'STP' 'ST2P'
or 'ST3P' as well, for compatibility with ANI-D.
- Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.
- For calls with an ANI failure, No. 1 Crossbar switches will send
forward a single-digit failure code, with no calling number digits
and no ST pulse to terminate the spill. I've made the ANI timeout
configurable so to reduce dead air time on calls with ANI fail.
- ANI info digits configurable. Modern digital switches will send 2
digits, but ANI-B sends only a single info digit. This caused the
ANI reported by Asterisk to be misaligned.
- Changed a confusing log message to be more informative.
ASTERISK-29518
Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256
app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.
ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29558
ASTERISK-29567
ASTERISK-29572
Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131
If chan_iax2 received a packet with an unsupported media format, for
example vp9, then it would set the frame's format to NULL. This could
then result in a crash later when an attempt was made to access the
format.
This patch makes it so chan_iax2 now ignores/drops frames received
with unsupported media format types.
ASTERISK-29392 #close
Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1
Some ATAs send hook flash events as application/hook-flash, rather than a DTMF
event. Now, we also recognize hook-flash as a flash event.
ASTERISK-29370
Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725
Enhancements:
* The MessageSend dialplan application now takes an optional
third argument that can set the message's "To" field on
outgoing messages. It's an alternative to using the
MESSAGE(to) dialplan function.
NOTE: No channel driver currently implements this field. A
follow-on commit for res_pjsip_messaging will implement it for
the chan_pjsip channel driver.
* To prevent confusion with the first argument, currently named
"to", it's been renamed to "destination". Its function,
creating the request URI, hasn't changed.
* The documentation for MessageSend was updated to be
more clear about the parameters and how they interact
the MESSAGE() dialplan function.
* With the rename of MessageSend's first parameter, and the fact
that message.c references <info> elements in chan_sip.c,
res_pjsip_messaging.c and res_xmpp, they each needed
documentation updates to use MessageDestinationInfo instead of
MessageToInfo.
* appdocsxml.dtd was updated to include a missing element
declaration for "dataType". This was showing up as an error
in Eclipse's dtd editor.
* Despite the changes in this commit, there should be
no impact to current users of MessageSend.
Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a
This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.
ASTERISK-29335
Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90
The system header strings was included mistakenly with commit 3de0204.
That header is included via asterisk.h and there via the compat.h.
Change-Id: I3dc49060e275295f785670c87cc65fd3c3abd24a
Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.
This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.
Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34
Two previous commits, 620d9f4 and 6d980de, allow to set up a call
without audio, again. That was introduced originally with commit f04d5fb
but changed and broke over time. The original commit missed one
scenario: A [peer] section in sip.conf, which does not allow audio at
all. In that case, chan_sip rejected the call, although even when the
requester offered no audio. Now, chan_sip does not check whether there
is no audio format but checks whether there is no format in general. In
other words, if there is at least one format to offer, the call succeeds.
However, to prevent calls with no-audio, chan_sip still rejects calls
when both call parties (caller = requester of the call *and* callee =
[peer] section in sip.conf) included audio. In such a case, it is
expected that the call should have audio.
ASTERISK-29280
Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e
If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext. You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.
* Added logic to iax2_call() and authenticate_reply() to print
a warning and hanhup the call if encryption is requested and
there's no secret or auth method. This prevents the crash.
* Added the ability to specify a default "auth" in the [general]
section of iax.conf.
ASTERISK-29624
Reported by: N A
Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254
The previous commit 6d980de fixed this issue in the core of Asterisk.
With that, each channel technology can be used without audio
theoretically. Practically, the channel-technology driver chan_sip
turned out to have an invalid check preventing that. chan_sip tested
whether there is at least one audio format. However, chan_sip has to
test whether there is at least one format. More cannot be tested while
requesting chan_sip because only the [general] capabilities but not the
[peer] caps are known yet. And the [peer] caps might not be a subset or
show any intersection with the [general] caps. This change here fixes
this.
The original commit f04d5fb, thirteen years ago, contained a software
bug as it passed ANY audio capability to the channel-technology driver.
Instead, it should have passed NO audio format. Therefore, this
addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8.
Then, Asterisk 10 changed that from ANY to NO, but nobody reported since
then.
ASTERISK-29265
Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f
When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received
This allows applications to perform actions based on the failure
reason.
ASTERISK-29252 #close
Reported-by: Dan Cropp
Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4
This completes the fix for ASTERISK_24543. Only when the call is an
outgoing call, consult and append the configured format capabilities
(p->caps). When all audio formats got rejected the negotiated format
capabilities (p->jointcaps) contain no audio formats for incoming
calls. This is required when there are other accepted media streams.
ASTERISK-29258
Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9
There are a couple of parameters (datalen and data) that do not get set
in chan_pjsip_indicate which could cause an Invalid message to pop up
for things such as fax. This patch adds them to the frame.
Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8
Previously, chan_sip parsed all known media streams in an SDP offer
like video (and text) even when videosupport=no (and textsupport=no).
This wasted processor power. Furthermore, chan_sip accepted SDP offers,
including no audio but just video (or text) streams although
videosupport=no (or textsupport=no). Finally, chan_sip denied the whole
offer instead of individual streams when they had encryption (SDES-sRTP)
unexpectedly enabled.
ASTERISK-29238
ASTERISK-29237
ASTERISK-29222
Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755
session->channel doesn't exist until chan_pjsip creates it, so intead of
setting a channel variable every new incoming call sets one and the same
global variable.
This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
a newly created channel, it also removes a misleading reference to
channel->session used to fetch call pickup configuraion.
ASTERISK-29240
Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755
The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice.
This resulted in extra noise in logs (for example, "is making progress"
and "is ringing" get logged twice by app_dial), as well as in noise in
signalling: one incoming 183 Session Progress results in 2 outgoing 183-s.
This change splits the response handler into 2 functions:
- one for updating HANGUPCAUSE, which is still called twice,
- another that does the rest, which is called only once as before.
ASTERISK-28016
Reported-by: Alex Hermann
ASTERISK-28549
Reported-by: Gant Liu
ASTERISK-28185
Reported-by: Julien
Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940
Add channel reference count for PJSIP REFER. The call could be terminated
prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY
occurred several minutes later, it would attempt to access a session which was
no longer valid. Terminate event subscription if pjsip_xfer_initiate() or
pjsip_xfer_send_request() fails in transfer_refer().
ASTERISK-29201 #close
Reported-by: Dan Cropp
Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435
This was dead code, test code introduced with Asterisk 13. This was
found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
partly fixes, not closes those two issues.
Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940
In some circumstances it was possible for an INVITE
session to be destroyed while we were still using it.
This occurred due to the reference on the INVITE session
being released internally as a result of its state
changing to DISCONNECTED.
This change adds a reference to the INVITE session
which is released when our own session is destroyed,
ensuring that the INVITE session remains valid for
the lifetime of our session.
ASTERISK-29022
Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9
12 years ago, with ASTERISK_12115 the last four get/uses of socket.port
vanished. However, the struct member itself and all seven set/uses
remained as dead code.
ASTERISK-28798
Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59
RFC 8760 added new digest-access-authentication schemes. Testing
revealed that chan_sip does not pick MD5 if several schemes are offered
by the User Agent Server (UAS). This change does not implement any of
the new schemes like SHA-256. This change makes sure, MD5 is picked so
UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can
still be used. This should have worked since day one because SIP/2.0
already envisioned several schemes (see RFC 3261 and its augmented BNF
for 'algorithm' which includes 'token' as third alternative; note: if
'algorithm' was not present, MD5 is still assumed even in RFC 7616).
Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd
The module description needs to be on the same line as the
AST_MODULE_INFO or it is not parsed correctly.
Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21
Added to:
* bridges/bridge_softmix.c
* channels/chan_pjsip.c
* include/asterisk/res_pjsip_session.h
* main/channel.c
* res/res_pjsip_session.c
There NO functional changes in this commit.
Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3
This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function
is called on a channel prior to answering a warning is issued and the
function returns unsuccessful.
ASTERISK-28878 #close
Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb
The ToHost parameter was not cleared when a peer's host value was
changed to dynamic. This causes invites to be sent to the original host.
ASTERISK-29011 #close
Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c
Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.
No functional changes were made with this commit.
Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c
When using the PSJIP_MEDIA_OFFER dialplan function it was not
overriding an endpoint's configured codecs on refresh unless
they had a shared codec between the two.
This patch makes it so whatever is set using PJSIP_MEDIA_OFFER
is used when creating the SDP for a refresh no matter what.
ASTERISK-28878 #close
Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6
The Streams API becomes the home for the core ACN capabilities.
These include...
* Parsing and formatting of codec negotation preferences.
* Resolving pending streams and topologies with those configured
using configured preferences.
* Utility functions for creating string representations of
streams, topologies, and negotiation preferences.
For codec negotiation preferences:
* Added ast_stream_codec_prefs_parse() which takes a string
representation of codec negotiation preferences, which
may come from a pjsip endpoint for example, and populates
a ast_stream_codec_negotiation_prefs structure.
* Added ast_stream_codec_prefs_to_str() which does the reverse.
* Added many functions to parse individual parameter name
and value strings to their respectrive enum values, and the
reverse.
For streams:
* Added ast_stream_create_resolved() which takes a "live" stream
and resolves it with a configured stream and the negotiation
preferences to create a new stream.
* Added ast_stream_to_str() which create a string representation
of a stream suitable for debug or display purposes.
For topology:
* Added ast_stream_topology_create_resolved() which takes a "live"
topology and resolves it, stream by stream, with a configured
topology stream and the negotiation preferences to create a new
topology.
* Added ast_stream_topology_to_str() which create a string
representation of a topology suitable for debug or display
purposes.
* Renamed ast_format_caps_from_topology() to
ast_stream_topology_get_formats() to be more consistent with
the existing ast_stream_get_formats().
Additional changes:
* A new function ast_format_cap_append_names() appends the results
to the ast_str buffer instead of replacing buffer contents.
Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56
chan_sip handle_response() function, for a 400 response to an INVITE,
calls handle_response_invite() and does not generate ACK.
handle_response_invite() does not recognize 400 response and has no
default response processing for unexpected responses, thus it does not
generate ACK either.
The ACK on response repetition comes from handle_response() mechanism
"We must re-send ACKs to re-transmitted final responses".
According to code history, 400 response specific processing was
introduced with commit
"channels/chan_sip: Add improved support for 4xx error codes"
This commit added support for :
- 400/414/493 in handle_response_subscribe() handle_response_register()
and handle_response().
- 414/493 only in handle_response_invite().
This fix adds 400 response support in handle_response_invite().
ASTERISK-28957
Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad
A patch made a reference to the PJSIP_SC_NULL enumeration value, which
was added to pjproject 2.8 and above thus making it so Asterisk would
fail to compile with prior versions of pjproject.
This patch removes the reference, and instead initializes the value
to '0'.
ASTERISK-28886 #close
Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7
The change to how setvar works for various channels performed in
ASTERISK~23756 missed some required change in the dahdi channel,
where the variables are actually set while reading configuration.
This change should fix the issue.
ASTERISK-28955
Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274
When fax_gateway_framehook is called and a gateway hasn't already
been started, the framehook gets the t38 state for both the current
channel and the peer. That call trickles down to the channel
driver which determines the state. If either channel is hung up
(or in the process of being hung up), the channel driver's tech_pvt
is going to be NULL which, in the case of chan_pjsip, will cause a
segfault.
* Added a hangup check for both the channel and peer channel
before starting a fax gateway.
* Added a check for NULL tech_pvt to chan_pjsip_queryoption
so we don't attempt to reference a tech_pvt that's already
gone.
ASTERISK-28923
Reported by: Yury Kirsanov
Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c
Some places in Asterisk did not treat the formats on a stream
as immutable when they are.
The ast_stream_get_formats function is now const to enforce this
and parts of Asterisk have been updated to take this into account.
Some violations of this were also fixed along the way.
An additional minor tweak is that streams are now allocated with
an empty format capabilities structure removing the need in various
places to check that one is present on the stream.
ASTERISK-28846
Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe
It is possible to configure a TCP/TLS client without having a TCP/TLS
server. In that case, no error or warning was printed but the headers
Contact and Via in SIP REGISTER were "(null)".
ASTERISK-28798
Change-Id: I387ca5cb6a65f1eb675a29c5e41df8ec6c242ab2
If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a
digit begin before media, or rtp has been setup then it's possible the
outgoing channel will hear a constant DTMF tone upon answering.
This happens because when there is no media, or rtp chan_pjsip notifies
the core to initiate inband DTMF. However, upon digit end if media, and
rtp become available then chan_pjsip does not notify the core to stop
inband DTMF. Thus the tone continues playing.
This patch makes it so chan_pjsip only notifies the core to start
inband DTMF in only the required cases. Now if there is no media, or
rtp availabe upon digit begin chan_pjsip does nothing, but tells the
core it handled it.
ASTERISK-28817 #close
Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5
Do not hang up a PJSIP channel on RTP timeout if that channel is in
a direct-media bridge. Also reset the time of the last received RTP packet when
direct-media ends (wait full rtp_timeout period before checking first time after
audio came back to Asterisk).
ASTERISK-28774
Reported-by: Michael Neuhauser
Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1
Fixes the following compile error:
chan_vpb.cc:2688:26: error: catching polymorphic type
‘class std::exception’ by value
Change-Id: Ic87bc357d72427d77626735c83200fd278a7a649
If the SSRC of a received RTP packet differed from the previous SSRC
an SSRC change control frame would be queued ahead of the media
frame. In the case of audio this would result in the format of the
audio frame not being checked, and if it differed or was not allowed
then it could cause the call to drop due to failure to set up a
translation path.
The chan_pjsip module will now no longer assume the first frame
will be the audio frame and instead goes through the complete list
to find it.
ASTERISK-28759
Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec
If you're for some reason out of RTP ports, chan_sip would previously
responde to an INVITE with a 403, which will fail the call.
Now, it returns a 503, allowing the device/proxy to retry the call on a
different machine.
ASTERISK-28718
Change-Id: I968dcf6c1e30ecddcce397dcda36db727c83ca90
Fixes no-audio issues when the media source is changed and
strictrtp is enabled (default).
If the peer media source changes, the SDP session version also changes.
If it is lower than the one we had stored, chan_sip would ignore it.
This changeset keeps track of the remote media origin identifier,
comparing that as well. If it changes, the session version needn't be
higher for us to accept the SDP.
Common scenario where this would've caused problems: a separate media
gateway that informs the caller about premium rates before handing off
the call to the final destination.
(An alternative fix would be to set ignoresdpversion=yes on the peer.)
ASTERISK-28686
Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee
If chan_pjsip receives an RTP packet whose payload differs from the
channel's native format, and asymmetric_rtp_codec is disabled (the
default), Asterisk will switch the channel's native format to match
that of the incoming packet without regard to the negotiated payloads.
We now check that the received frame is in a format we have negotiated
before switching payloads which results in these packets being dropped
instead of causing the session to terminate.
ASTERISK-28139 #close
Reported by: Paul Brooks
Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3
The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX
to match behavior of "no timeout" defined in comment.
ASTERISK-28702 #close
Change-Id: I4ea015986e061374385dba247b272f7aac60bf11
lws2sws() does not stop trying to handle header continuation lines
even after all headers have been found. This is problematic if the
first character of a SIP message body is a space or tab character, so
we update to recognize the end of the message header.
ASTERISK-28693 #close
Reported by: Frank Matano
Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df
This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.
A description of the protocol can be found on the above referenced
GitHub page. A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.
ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.
ASTERISK-28484 #close
Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
The change to add setting hangupsource to sig_pri_queue_hangup()
made in https://gerrit.asterisk.org/c/asterisk/+/12857 casued
deadlocks when a hangup request was received from the core at the
same time a hanguprequest was received from the remote end via the
D channel.
Although the PRI's channel private structure was being unlocked
before setting the hangupsource, the PRI's own lock was still being
held during the process. If channel actions were also coming from
the core, a deadlock on the PRI could result. This deadlock could
then escalate to the entire DAHDI subsystem via DAHDI's global
interface list lock, especially if someone used the PRI CLI commands.
Fix:
* We now unlock the PRI as well as the PRI's channel private
structure before setting the hangupsource, then relock both
afterwards.
ASTERISK-28605
Reported by: Dirk Wendland
Change-Id: Id74aaa5d4e3746063dbe9deed188eb65193cb9c9
chan_sip.c:3782 __sip_xmit: sip_xmit of 0x7f1478069230 (len 600) to
213.150.203.60:1492 returned -2: Interrupted system call
returned -2 implies this wasn't actually an OS error, so errno makes no
sense either. Internal error was already logged higher up, and -2
generally means that either there isn't a valid connection available, or
the pipe notification failed, and that is already correctly logged.
ASTERISK-28651 #close
Change-Id: I46eb82924beeff9dfd86fa6c7eb87d2651b950f2
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
The SIP transaction state was reset when emitting an UPDATE or a re-INVITE
related to a COLP, preventing RTP packets to be emitted.
ASTERISK-28647
Change-Id: Ie7a30fa7a97f711e7ba6cc17f221a0993d48bd8b
During capabilities selection (joint capabilities of us and peer,
configured capability for this peer, or general configured
capabilities), if sip_new() does not keep framing information,
then directmedia activation will fail for any framing different
from default framing.
ASTERISK-28637
Change-Id: I99257502788653c2816fc991cac7946453082466
If the name of a peer is known and a SIP request is sent using that
peer's name, the address of the peer will change even if the request
fails the authentication challenge. This means that an endpoint can
be altered and even rendered unusuable, even if it was in a working
state previously. This can only occur when the nat option is set to the
default, or auto_force_rport.
This change checks the result of authentication first to ensure it is
successful before setting the address and the nat option.
ASTERISK-28589 #close
Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df
Upon a short PRI disconnection, libpri may maintain Q.921 layer 'up' and
may thus not send PRI_EVENT_DCHAN_DOWN / PRI_EVENT_DCHAN_UP events.
If pri_event_alarm() clears DCHAN_UP status bit upon alarm detection
and no Q.921 reconnection sequence occurs, chan_dahdi will keep
seeing span status "Down" at the end of alarm.
This patch modifies pri_event_alarm() in order to keep DCHAN_UP bit
unchanged. libpri will send a PRI_EVENT_DCHAN_DOWN event if it detects
a disconnection of Q.921 layer and this will clear DCHAN_UP if required.
ASTERISK-28615
Change-Id: Ibe27df4971fd4c82cc6850020bce4a8b2692c996
This patch fixes several issues reported by the lgtm code analysis tool:
https://lgtm.com/projects/g/asterisk/asterisk
Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:
* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards
Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25
* Pass caller information to frame allocation functions.
* Disable caching as it interfers with MALLOC_DEBUG reporting.
* Stop using ast_calloc_cache.
Change-Id: Id343cd80a3db941d2daefde2a060750fea8cd260
During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.
ASTERISK-28578
Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f
Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis
ASTERISK-28542 #close
Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8
Kevin Harwell
Naveen Albert
Boris P. Korzun
Mark Petersen
Sean Bright
Alexander Traud
Josh Soref
Mike Bradeen
Sarah Autumn
Joshua C. Colp
Naveen Albert
George Joseph
Holger Hans Peter Freyther
Dan Cropp
Ben Ford
Ivan Poddubnyi
Richard Mudgett
Kevin Harwell
Dennis Buteyn
Frederic LE FOLL
Guido Falsi
Michael Neuhauser
Paulo Vicentini
Walter Doekes
Friendly Automation
Andrew Siplas
Seán C McCord
Jaco Kroon
Jean Aunis
Joshua Colp
Corey Farrell
Salah Ahmed
lvl
Torrey Searle