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
This change adds H.265/HEVC as a known codec and creates a cached
"h265" media format for use.
Note that RFC 7798 section 7.2 also describes additional SDP
parameters. Handling of these is not yet supported.
ASTERISK-28512
Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2
On FreeBSD using the clang/llvm compiler build fails to build due
to the switch statement argument being a non integer type expression.
Switch to an if/else if/else construct to sidestep the issue.
ASTERISK-28536 #close
Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8
When fax detection occurs on an outbound PJSIP channel the
redirect operation will result in a masquerade occurring and
the underlying channel on the session changing. The code
incorrectly relocked the new channel instead of the old
channel when returning. This resulted in the new channel
being locked indefinitely. The code now always acts on the
expected channel.
ASTERISK-28538
Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3
When modifying an already defined variable in some channel drivers they
add a new variable with the same name to the list, but that value is
never used, only the first one found.
Introduce ast_variable_list_replace() and use it where appropriate.
ASTERISK-23756 #close
Patches:
setvar-multiplie.patch submitted by Michael Goryainov
Change-Id: Ie1897a96c82b8945e752733612ee963686f32839
When the remote ISDN party ends an ISDN call on a PRI link
(DISCONNECT), CHANNEL(hangupsource) information is not available.
chan_dahdi already contains an ast_set_hangupsource() in
__dahdi_exception() function but it seems that ISDN message processing
does not use this part of code.
Two other channel modules associate ast_queue_hangup() and
ast_set_hangupsource() functions calls:
- chan_pjsip in chan_pjsip_session_end() function,
- chan_sip in sip_queue_hangup_cause() function.
chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and
set_hangup_source_and_cause().
Thus, I propose to add ast_set_hangupsource() beside
ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and
chan_sip already do.
ASTERISK-28525
Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c
The links in the deprecation notice were the shortened
variety but it makes better sense to show the unshortened
links as they're more descriptive.
I.E.
wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
rather than
wiki.asterisk.org/wiki/x/tAHOAQ
Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9
On reading information about initial client packet unistim use dirty
implementation of destination ip address retrieval. This fix uses
CMSG_*(..) to get ip address and make clang compile without warning.
ASTERISK-25592 #close
Reported-by: Alexander Traud
Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1
Current implementation of ast_channel_tech send_digit_begin hook uses
same function for tone playback as key press handler. This cause every
incoming dtmf send back to asterisk. In case of two unistim phones
connected to each other, it'll cause indefinite DTMF loop. Fix add
separate function for dtmf tone phone play.
Change-Id: I5795db468df552f0c89c7576b6b3858b26c4eab4
This patch fixes one-way oudio that users expirienced on
big-endian architechtires. RTP port number bytes was stored
in improper order and phone sent RTP to wrong RTP port.
Reported-by: Andrey Ionov
Change-Id: I9a9ca7f26e31a67bbbceff12923baa10dfb8a3be
The UnicastRTP channel driver provided by chan_rtp now accepts
"<hostname>:<port>" as an alternative to "<ip_address>:<port>"
in the destination. The first AAAA (preferred) or A record resolved
will be used as the destination. The lookup is synchronous so beware
of possible dialplan delays if you specify a hostname.
Change-Id: Ie6f95b983a8792bf0dacc64c7953a41032dba677
There were still a few places in the code that could overflow when "packing"
a json object with a value outside the base type integer's range. For instance:
unsigned int value = INT_MAX + 1
ast_json_pack("{s: i}", value);
would result in a negative number being "packed". In those situations this patch
alters those values to a ast_json_int_t, which widens the value up to a long or
long long.
ASTERISK-28480
Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1
Otherwise, OpenR2 threads go crazy and consume almost all CPU resources
Change-Id: I10a41f617613fe7399c5bdced5c64a2751173f28
Signed-off-by: Oron Peled <oron.peled@xorcom.com>
Details:
- The memcpy() call copied part of "dahdi_conf" and not "dahdi_conf.mfcr2"
- As a result, the memcmp() in dahdi_r2_get_link() always fails
- This cause dahdi_r2_get_link() to create new link for every channel
(instead of a new link for every ~30 channels)
- With the fix, far less links are generated -- so we use far less threads
Change-Id: I7259dd6272f5e46e8a6c7f5bf3e8c2ec01b8c132
Signed-off-by: Oron Peled <oron.peled@xorcom.com>
The chan_sip module performs a T.38 re-invite using a single media
stream of udptl, and expects the SDP answer to be the same.
If an SDP answer is received instead that contains an additional
media stream with no joint codec a crash will occur as the code
assumes that at least one joint codec will exist in this
scenario.
This change removes this assumption.
ASTERISK-28465
Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
The MWI core recently got some new API calls that make tracking MWI state
lifetime more reliable. This patch updates those modules that subscribe to
specific MWI topics to use the new API. Specifically, these modules now
subscribe to both MWI topics and MWI state.
ASTERISK-28442
Change-Id: I32bef880b647246823dbccdf44a98d384fcabfbd
Fixes a crash in chan_dahdi occurring on 32-bit systems. A previous
patch introduced a variable of type unassigned long long which is 64-bits.
Casting it as 'ast_json_int_t' along with JSON type 'I' makes it work
with 32-bit systems.
ASTERISK-28457
Change-Id: I9cef6b5f2d826fc5c93f2f6a1c997c4e3e6c93fe
Previously, when a Transfer (REFER) was performed, chan_pjsip would set
the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not
reflect a successful/unsuccessful transfer the way chan_sip did.
Added a callback module to process the refer subscription information.
Now depends on res_pjsip_pubsub so call transfer progress can be monitored
and reported
ASTERISK-26968 #close
Reported-by: Dan Cropp
Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc
Fixed format-truncation issues in chan_dahdi.c and
sig_analog.c. Since they're related to fields provided
by dahdi-tools we can't change the buffer sizes so we're just
checking the return from snprintf and printing an errior if we
overflow.
Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5
We have seen some rare case of segmentation fault in hangup function
and we could notice that channel pointer was NULL. Debug log shows
that there is a 200 OK answer and SIP timeout at the same time. It
looks that while the SIP session was being destroyed due to timeout
call hangup due to answer event lead to race condition and channel
is being destroyed from two different places. The check ensures we
check it not to be NULL before freeing it.
ASTERISK-25371
Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778
After some definitions have been moved to asterisk/mwi.h the files
channels/chan_dahdi.h channels/sig_pri.c are missing this new
include.
ASTERISK-28427 #close
Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91
The caller endpoint hears dead silence if a callee replies 180 (without SDP)
and the caller already received 183 (with SDP).
It happens because Asterisk sends 180 (WITH SDP) to the caller,
there are not incoming RTP packets from the callee
and Asterisk does not generate inband ringing,
so there are not any outgoing RTP packets to the caller.
This patch replaces 180 by 183 if SDP negotiation has completed,
as if the caller endpoint is configured with "inband_progress=yes".
In this case Asterisk will generate inband ringing untill Asterisk receive
incoming RTP packets from the callee.
ASTERISK-27994 #close
Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73
Various fixes for issues caught by gcc 9. Mostly snprintf
trying to copy to a buffer potentially too small.
ASTERISK-28412
Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0
When the dtmf_mode on an endpoint is configured as "auto_info"
Asterisk will produce an inband DTMF tone alongside an INFO
message when sending DTMF.
ASTERISK-28371
Change-Id: I1380b82f006e110a1b83fbb50c9873edd13a5d9a
The compiler complained about a couple of variables that weren't
initialized but were being used. Initializing them to NULL resolves the
warnings/errors.
ASTERISK-28362 #close
Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b
The next usage of PJSIP_PARSE_URI will crash asterisk
${PJSIP_PARSE_URI(tel:+1234567890,host)}
or
${PJSIP_PARSE_URI(192.168.1.1:5060,host)}
The function pjsip_parse_uri successfully parses then, but returns
struct pjsip_other_uri *.
This patch restricts parsing only SIP/SIPS URIs.
Change-Id: I16f255c2b86a80a67e9f9604b94b129a381dd25e
Passing negative intervals to the scheduler rips a hole in the
space-time continuum.
ASTERISK-25792 #close
Reported by: Paul Sandys
Change-Id: Ie706f21cee05f76ffb6f7d89e9c867930ee7bcd7
Add logical group at DAHDIChannel event
and create "dahdi_group" at CHANNEL function.
ASTERISK-28317
Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a
chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP. This new flag allows chan_pjsip to have the same
behavior as chan_sip.
ASTERISK-28322 #close
Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a
strtok() uses a static buffer, making it not thread safe.
Also add a #define to cause a compile failure if strtok is used.
Change-Id: Icce265153e1e65adafa8849334438ab6d190e541
Currently, the pjsip show channelstats cli does not show channel's
stats after hits the invalid channel info. This makes hard to use
this cli. Changed to keep iterate after hits the invalid channel
info.
ASTERISK-28292
Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c
When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.
This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.
The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.
Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.
ASTERISK-28119
Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423
Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.
As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.
The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.
ast_channel_snapshot_get_latest() still returns the latest snapshot.
The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.
ASTERISK-28102
Change-Id: I9334febff60a82d7c39703e49059fa3a68825786
Replace usage of ao2_container_alloc with ao2_container_alloc_hash or
ao2_container_alloc_list. Remove ao2_container_alloc macro.
Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088
New dialplan function PJSIP_PARSE_URI added to parse an URI and return
a specified part of the URI.
This is useful when need to get part of the URI instead of cutting it
using a CUT function.
For example to get 'user' part of Remote URI
${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)}
ASTERISK-28144 #close
Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a
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
When a call pickup is performed using and invite with replaces header
the ast_do_pickup method is attempted and a PICKUP stasis message is sent.
ASTERISK-28081 #close
Reported-by: Luit van Drongelen
Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a
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
This officially deprecates chan_sip in Asterisk 17+. A warning is
printed upon startup or module load to tell users that they should
consider migrating. chan_sip is still built by default but the default
modules.conf skips loading it at startup.
Very important to note we are not scheduling a time where chan_sip will
be removed. The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.
Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93
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 also removed. Only ao2_container_alloc remains due to
it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
With tls and udp enabled asterisk generates a warning about sending
message via udp instead of tls.
sip notify command via cli works as expected and without warning.
asterisk has to set the connection information accordingly to connection
and not on presumption
ASTERISK-28057 #close
Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e
Fixes random asterisk crash on start or reload with TLS phones.
ASTERISK-28034 #close
Reported-by: David Hajek
Change-Id: I2a859f97dc80c348e2fa56e918214ee29521c4ac
Also remove function peer_ipcmp_cb since it's not used (according to
rmudgett).
Prior to b2c4e8660a (ASTERISK_27457)
insecure=port was the defacto standard. That commit also prevented
insecure=port from being applied for sip/tcp or sip/tls.
Into consideration there are three sets of behaviour:
1. "previous" - before the above commit.
2. "current" - post above commit, pre this one.
3. "new" - post this commit.
The problem that the above commit tried to address was guests over TCP.
It succeeded in doing that but broke transport!=udp with host!=dynamic.
This commit attempts to restore sane behaviour with respect to
transport!=udp for host!=dynamic whilst still retaining the guest users
over tcp.
It should be noted that when looking for a peer, two passes are made, the
first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer,
thus looking for full matches (IP + Port), the second pass sets
SIP_INSECURE_PORT, thus expecting matches on IP only where the matched
peer allows for that (in the author's opinion: UDP with insecure=port,
or any TCP based, non-dynamic host).
In previous behaviour there was special handling for transport=tcp|tls
whereby a peer would match during the first pass if the utilized
transport was TCP|TLS (and the peer allowed that specific transport).
This behaviour was wrong, or dubious at best. Consider two dynamic tcp
peers, both registering from the same IP (NAT), in this case either peer
could match for connections from an IP. It's also this behaviour that
prevented SIP guests over tcp.
The above referenced commit removed this behaviour, but kept applying
the SIP_INSECURE_PORT only to WS|WSS|UDP. Since WS and WSS is also TCP
based, the logic here should fall into the TCP category.
This patch updates things such that the previously non-explicit (TCP
behaviour) transport test gets performed explicitly (ie, matched peer
must allow for the used transport), as well as the indeterministic
source-port nature of the TCP protocol is taken into account. The new
match algorithm now looks like:
1. As per previous behaviour, IP address is matched first.
2. Explicit filter with respect to transport protocol, previous
behaviour was semi-implied in the test for TCP pure IP match - this now
made explicit.
3. During first pass (without SIP_INSECURE_PORT), always match on port.
4. If doing UDP, match if matched against peer also has
SIP_INSECURE_PORT, else don't match.
5. Match if not a dynamic host (for non-UDP protocols)
6. Don't match if this is WS|WSS, or we can't trust the Contact address
(presumably due to NAT)
7. Match (we have a valid Contact thus if the IP matches we have no
choice, this will likely only apply to non-NAT).
To logic-test this we need a few different scenarios. Towards this end,
I work with a set number of peers defined in sip.conf:
[peer1]
host=1.1.1.1
transport=tcp
[peer2]
host=1.1.1.1
transport=udp
[peer3]
host=1.1.1.1
port=5061
insecure=port
transport=udp
[peer4]
host=1.1.1.2
transport=udp,tcp
[peer5]
host=dynamic
transport=udp,tcp
Test cases for UDP:
1 - incoming UDP request from 1.1.1.1:
- previous:
- pass 1:
* peer1 or peer2 if from port 5060 (indeterminate, depends on peer
ordering)
* peer3 if from port 5061
* peer5 if registered from 1.1.1.1 and source port matches
- pass 2:
* peer3
- current: as per previous.
- new:
- pass 1:
* peer2 if from port 5060
* peer3 if from port 5061
* peer5 if registered from 1.1.1.1 and source port matches
- pass 2:
* peer3
2 - incoming UDP request from 1.1.1.2:
- previous:
- pass 1:
* peer5 if registered from 1.1.1.2 and port matches
* peer4 if source port is 5060
- pass 2:
* no match (guest)
- current: as previous.
- new as previous (with the variation that if peer5 didn't have udp as
allowed transport it would not match peer5 whereas previous
and current code could).
3 - incoming UDP request from anywhere else:
- previous:
- pass 1:
* peer5 if registered from that address and source port matches.
- pass 2:
* peer5 if insecure=port is additionally set.
* no match (guest)
- current - as per previous
- new - as per previous
Test cases for TCP based transports:
4 - incoming TCP request from 1.1.1.1
- previous:
- pass 1 (indeterministic, depends on ordering of peers in memory):
* peer1; or
* peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or
* peer2 if the source port happens to be 5060; or
* peer3 if the source port happens to be 5061.
- pass 2: cannot happen since pass 1 will always find a peer.
- current:
- pass 1:
* peer1 or peer2 if from source port 5060
* peer3 if from source port 5060
* peer5 if registered as 1.1.1.1 and source port matches
- pass 2:
* no match (guest)
- new:
- pass 1:
* peer 1 if from port 5060
* peer 5 if registered and source port matches
- pass 2:
* peer 1
5 - incoming TCP request from 1.1.1.2
- previous (indeterminate, depends on ordering):
- pass 1:
* peer4; or
* peer5 if peer5 registered from 1.1.1.2
- pass 2: cannot happen since pass 1 will always find a peer.
- current:
- pass 1:
* peer4 if source port is 5060
* peer5 if peer5 registered as 1.1.1.2 and source port matches
- pass 2:
* no match (guest).
- new:
- pass 1:
* peer4 if source port is 5060
* peer5 if peer5 registered as 1.1.1.2 and source port matches
- pass 2:
* peer4
6 - incoming TCP request from anywhere else:
- previous:
- pass 1:
* peer5 if registered from that address
- pass 2: cannot happen since pass 1 will always find a peer.
- current:
- pass 1:
* peer5 if registered from that address and port matches.
- pass 2:
* no match (guest)
- new: as per current.
It should be noted the test cases don't make explicit mention of TLS, WS
or WSS. WS and WSS previously followed UDP semantics, they will now
enforce source port matching. TLS follow TCP semantics.
The previous commit specifically tried to address test-case 6, but broke
test-cases 4 and 5 in the process.
ASTERISK-27881 #close
Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2
If asterisk offer an endpoint with SRTP and that endpoint respond
with non srtp, in that case channel(rtp,secure,audio) reply wrong
status.
Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key:
Currently this flag has being set redundantly. In either case identical
or different remote_key this flag has being set. So if we
don't set it while we receive identical remote_key or non SRTP SDP
response then we can take decision of srtp use by using that flag.
ASTERISK-27999
Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7
Changing any Menuselect option in the `Compiler Flags` section causes a
full rebuild of the Asterisk source tree. Every enabled option causes
a #define to be added to buildopts.h, thus breaking ccache caching for
every source file that includes "asterisk.h". In most cases each option
only applies to one or two files. Now we only define those options for
the specific sources which use them, this causes much better cache
matching when working with multiple builds. For example testing code
with an without MALLOC_DEBUG will now use just over half the ccache
size, only main/astmm.o will have two builds cached instead of every
file.
Reorder main/Makefile so _ASTCFLAGS set on specific object files are all
together, sorted by filename. Stop adding -DMALLOC_DEBUG to CFLAGS of
bundled pjproject, this define is no longer used by any header so only
serves to break cache.
The only code change is a slight adjustment to how main/astmm.c is
initialized. Initialization functions always exist so main/asterisk.c
can call them unconditionally. Additionally rename the astmm
initialization functions so they are not exported.
Change-Id: Ie2085237a964f6e1e6fff55ed046e2afff83c027
I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.
ASTERISK-27965
Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7
The iax2 module is not handling timeout and EINTR case properly. Mainly when
there is an interupt to the kernel thread. In case of ast_io_wait recieves a
signal, or timeout it can be an error or return 0 which eventually escapes the
thread loop, so that it cant recieve any data. This then causes the modules
receive queue to build up on the kernel and stop any communications via iax in
asterisk.
The proposed patch is for the iax module, so that timeout and EINTR does not
exit the thread.
ASTERISK-27705
Reported-by: Kirsty Tyerman
Change-Id: Ib4c32562f69335869adc1783608e940c3535fbfb
chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant
it was not updating HANGUPCAUSE for 4XX responses. If the remote end
sent a "180 Ringing", then a "486 Busy", the hangup cause was left at
"180 Normal Clearing".
* Removed chan_pjsip_incoming_response from the original session
supplement (which was handling only "AFTER MEDIA") and added it to a
new session supplement which accepts both "BEFORE_MEDIA" and
"AFTER_MEDIA".
* Also cleaned up some cleanup code in load module.
ASTERISK-27902
Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a
This function originally was used in chan_sip to enable some simplifying
assumptions and eventually was copy and pasted into res_pjsip_logger and
res_hep. Since it's replicated in three places, it's probably best to
move it into the public netsock2 API for these modules to use.
Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04
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
Analog phones dial overlap dialing and it is chan_dahdi's job to read the
numbers. It has three timeout constants that this commit converts to
channel-level configuration options:
* firstdigit_timeout: Default time (ms) to detect first digit
* interdigit_timeout: Default time (ms) to detect following digits
* matchdigit_timeout: Default time (ms) to wait in case of ambiguous
match. This happens when the dialed digits match a number in the current
context but are also the prefix of another number.
Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213
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
clang 6.0 warned about this. Beside that, this change removes the used variable
'desc'.
ASTERISK-27808
Change-Id: Ia26bdcc0a562c058151814511cfcf70ecafa595b
ast_sip_push_task_synchronous() did not necessarily execute the passed in
task under the specified serializer. If the current thread is any
registered pjsip thread then it would execute the task immediately instead
of under the specified serializer. Reentrancy issues could result if the
task does not execute with the right serializer.
The original reason ast_sip_push_task_synchronous() checked to see if the
current thread was a registered pjsip thread was because of a deadlock
with masquerades and the channel technology's fixup callback
(ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356)
involving call pickups avoided the original deadlock situation entirely.
The PJSIP channel technology's fixup callback no longer needed to call
ast_sip_push_task_synchronous().
However, there are a few places where this unexpected behavior is still
required to avoid deadlocks. The pjsip monitor thread executes callbacks
that do calls to ast_sip_push_task_synchronous() that would deadlock if
the task were actually pushed to the specified serializer. I ran into one
dealing with the pubsub subscriptions where an ao2 destructor called
ast_sip_push_task_synchronous().
* Split ast_sip_push_task_synchronous() into
ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer().
ast_sip_push_task_wait_servant() has the old behavior of
ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has
the new behavior where the task is always executed by the specified
serializer or a picked serializer if one is not passed in. Both functions
behave the same if the current thread is not a SIP servant.
* Redirected ast_sip_push_task_synchronous() to
ast_sip_push_task_wait_servant() to preserve API for released branches.
ASTERISK_26806
Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
There is a problem when an INVITE-with-Replaces transfer targets a channel
in a ConfBridge. The transfer will unconditionally swap out the
ConfBridge channel. Unfortunately, the ConfBridge state will not be aware
of this change. Unexpected behavior will happen as a result since
ConfBridge channels currently can only be replaced by a masquerade and not
normal bridge channel moves.
* We just need to pretend that the channel isn't in a bridge (like other
transfer methods already do) so the transfer channel will masquerade into
the ConfBridge channel.
Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82
This change allows chan_pjsip to be given an AST_FRAME_RTCP
containing REMB feedback and pass it to res_rtp_asterisk.
Once res_rtp_asterisk receives the frame a REMB RTCP feedback
packet is constructed with the appropriate contents and sent
to the remote endpoint.
ASTERISK-27776
Change-Id: Ic53f821c1560d8924907ad82c4d9c0bc322b38cd
Given the below call scenario:
A -> Ast1 -> B
C <- Ast2 <- B
1) A calls B through Ast1
2) B calls C through Ast2
3) B transfers A to C
When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to
send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of
the channel between Ast1 and Ast2.
Channel ref leaks are easily seen in the CLI "core show channels" output.
The leaked channels appear in the output but you can do nothing with them
and they never go away unless you restart Asterisk.
* Properly account for the channel refs when imparting a channel into a
bridge when handling an INVITE with replaces in handle_invite_replaces().
The ast_bridge_impart() function steals a channel ref but the code didn't
account for how many refs were held by the code at the time and which ref
was stolen.
* Eliminated RAII_VAR in handle_invite_replaces().
ASTERISK-27740
Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4
This change extends the existing AST_FRAME_RTCP frame type to be
able to contain additional RTCP message types, such as feedback
messages. The payload type is contained in the subclass which allows
knowing what is in the frame itself.
The RTCP feedback message type is now handled and REMB[1] messages
are raised with their containing information.
This also fixes a bug where all feedback messages were triggering
video updates instead of just FIR and FUR.
Finally RTCP frames are now passed up through the Asterisk core to
what is handling the channel, mapped appropriately in the case of
bridging, and written to an outgoing stream. Since RTCP frames are
on a per-stream basis this is only done on multistream capable
channels.
[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
ASTERISK-27758
ASTERISK-26366
Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e
In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some
libraries do not specify all their dependencies and require additional shared
libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a
library is specified there, it must exist on the platform, because ./configure
tries to compile/link/execute a small app using those statements. For example,
the library libdl.so is Linux specific and does not exist on BSD-like platforms.
Furthermore, no supported platform/version was found, which still (ever?)
requires those additional libraries. Therefore, they were simply removed.
Finally, this change adds the error code ESTRPIPE to the channel driver
chan_alsa for those platforms which lack it, again for example NetBSD.
ASTERISK-27720
Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b
* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl
These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.
Some of these modules are still initialized or shutdown from outside the
module loader. logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).
Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f
Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
The "ptime" SDP parameter received in a SIP response was not honoured.
Moreover, in the abscence of this "ptime" parameter, locally configured
framing was lost during response processing.
This patch systematically stores the framing information in the
ast_rtp_codecs structure, taking it from the response or from the
configuration as appropriate.
ASTERISK-27674
Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c
When constructing a dialog-info+xml NOTIFY message a ringing channel
is found if the state is ringing and further information is placed into
the message. Due to the migration to the Stasis message bus this did
not always work as expected.
This change raises a second ringing event in such a way to guarantee
that the event is received by chan_sip and another lookup is done to
find the ringing channel.
ASTERISK-24488
Change-Id: I547a458fc59721c918cb48be060cbfc3c88bcf9c
Check if initreq data string exists before using it when processing a
CANCEL request.
ASTERISK-27666
Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97
It seems that the ALSA backend of PortAudio doesn't know how to both
read and write at the same time by adding a per-device mutex.
FIXME: currently only a draft version. Need to either auto-detect
we work with the ALSA backend or add an extra configuration option
to use this mutex.
ASTERISK-27426 #close
Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
This patch fix chan_unistim hold functions to correctly support
hold function in different states possible in case of multiple lines
established on the phone
ASTERISK-26596 #close
Change-Id: Ib1e04e482e7c8939607a42d7fddacc07e26e14d4
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
Per RFC 5245, the foundation specified with an ICE candidate can be up
to 32 characters but we are only allowing for 31.
ASTERISK-27498 #close
Reported by: Michele Prà
Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf
This is the old ASTOBJ macro's which are no longer used except by the
deprecated netsock.c. Move it to the chan_iax2 include folder so it
does not get used elsewhere.
Change-Id: I7e4ae96678b36b9f41d3cae14b167f110eb5d349
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
In change_redirecting_information variables we use ast_strlen_zero to
see if a value should be saved. In the case where the value is not NULL
but is a zero length string we leaked.
handle_response_subscribe leaked a reference to the ccss monitor
instance.
Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f
Some variables are set and never changed, making them constant. This
means that code in the 'false' block of the conditional is unreachable.
In chan_skinny and res_config_ldap I used preprocessor directive `#if 0`
as I'm unsure if the unreachable code could be enabled in the future.
Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059
chan_console supports multiple devices but the CLI only works on a
single device. 'console set active' selects this device.
Sadly that CLI picks the wrong command-line parameter and will only
work for a device called 'active'.
ASTERISK-27490 #close
Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d
This moves netsock.c / netsock.h to the chan_iax2 module. netsock.h has
been marked deprecated since 13.0.0, chan_iax2 is the only remaining
user.
Change-Id: I28c6578043bac18de5ea608e136acec4f83d5dd3
Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and
disable_multi_domain=no results in a misleading empty endpoint name
message. The message should say the endpoint was not found.
* Added missing endpoint not found message.
* Added more information to the empty endpoint name msgs if available.
* Eliminated RAII_VAR in request().
Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4
Log a message to security events when an INVITE is received to an
invalid extension.
ASTERISK-25869 #close
Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f
Remove nearly all use of regex from ACO users. Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
callers use simple prefix based regex. I haven't decided the best
way to fix this in both 13/15 and master.
Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
Stripping the DNID in a SIP dial string can result in attempting to call
the argument parsing macros on an empty string, causing a crash.
ASTERISK-26131 #close
Reported by: Dwayne Hubbard
Patches:
dw-asterisk-master-dnid-crash.patch (license #6257) patch
uploaded by Dwayne Hubbard
Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
Previously, peers connected via TCP (or TLS) were matched by ignoring their
source port. One cannot say anything when protocol:IP:port match, yes (see
<http://stackoverflow.com/q/3329641>). However, when the ports do not match, the
peers do not match as well.
This change allows two peers connected to an Asterisk server via TCP (or TLS)
behind a NAT (= same source IP address) to be differentiated via their port as
well.
ASTERISK-27457
Reported by: Stephane Chazelas
Change-Id: Id190428bf1d931f2dbfd4b293f53ff8f20d98efa
chan_skinny creates a new thread for each new session. In trying
to be a good cleanup citizen, the threads are joinable and the
unload_module function does a pthread_cancel() and a pthread_join()
on any sessions that are active at that time. This has an
unintended side effect though. Since you can call pthread_join on a
thread that's already terminated, pthreads keeps the thread's
storage around until you explicitly call pthread_join (or
pthread_detach()). Since only the module_unload function was
calling pthread_join, and even then only on the ones active at the
tme, the storage for every thread/session ever created sticks
around until asterisk exits.
* A thread can detach itself so the session_destroy() function
now calls pthread_detach() just before it frees the session
memory allocation. The module_unload function still takes care
of the ones that are still active should the module be unloaded.
ASTERISK-27452
Reported by: Juan Sacco
Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd
(cherry picked from commit 8f5dff543e)
The media frame cache gets in the way of finding use after free errors of
media frames. Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.
* Added the "cache_media_frames" option to asterisk.conf. Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.
To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.
Sample asterisk.conf setting:
[options]
cache_media_frames=no
ASTERISK-27413
Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00
This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.
Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.
ASTERISK-27395
Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4
When chan_sip receives a SUBSCRIBE request with no "Expires" header it
processes the request as an unsubscribe. This is incorrect, per RFC3264
when the "Expires" header is missing a default expiry should be used.
ASTERISK-18140
Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5
When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'. This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.
ASTERISK-23462 #close
Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318
The sys/sysmacros.h include file does not exist in BSD systems and
is not required to build this module there.
Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section
already exist I moved that include line inside it's #else branch.
ASTERISK-27343 #close
Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1
chan_vpb was trying to use sizeof(*p->play_dtmf), where
p->play_dtmf is defined as char[16], to get the length of the array
but since p->play_dtmf is an actual array, sizeof(*p->play_dtmf)
returns the size of the first array element, which is 1. gcc7
validly complains because the context in which it's used could
cause an out-of-bounds condition.
Change-Id: If9c4bfdb6b02fa72d39e0c09bf88900663c000ba
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
Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.
This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.
ASTERISK-27277
Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07
chan_pjsip_indicate was missing a case for the recently added
AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an
error and causing the call to be hung up instead of just ignoring
it.
ASTERISK-27260
Reported by: Daniel Heckl
Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80
Provide a way to get the contents of the the Request URI from the initial SIP
INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}")
ASTERISK-27278
Reported by: David J. Pryke
Tested by: David J. Pryke
Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e
Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.
ASTERISK-21399 #close
Reported by: Tzafrir Cohen
Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3
In handle_request_invite, when processing a pickup, a call
is made to get_sip_pvt_from_replaces to locate the pvt for
the subscription. The pvt is assumed to be valid when zero
is returned indicating no error, and is dereferenced which
can cause a crash if it was not found.
This change checks the not found case and returns -1 which
allows the calling code to fail appropriately.
ASTERISK-27217 #close
Reported-by: Bryan Walters
Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612
If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE
to both parties to set up media path directly between the endpoints.
In this reINVITE msg SDP origin line (o=) contains IP address of endpoint
instead of IP of asterisk. This behavior violates RFC3264, sec 8:
"When issuing an offer that modifies the session,
the "o=" line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP."
This patch assures IP address of Asterisk is always sent in
SDP origin line.
ASTERISK-17540
Reported by: saghul
Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e
When rtp_keepalive is on for a PJSIP endpoint dialing to another
Asterisk instance also using PJSIP, Asterisk will continue to print
warning messages about not being able to send frames of a certain
type. This suppresses that warning message.
Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67
Create local_tag and remote_tag in CHANNEL info to get tag from From and
To headers of a SIP dialog.
ASTERISK-27220
Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524
Kevin Harwell
George Joseph
Frederic LE FOLL
Guido Falsi
Joshua C. Colp
Alexander Traud
Michael Neuhauser
Sean Bright
Paulo Vicentini
Walter Doekes
Friendly Automation
Andrew Siplas
Seán C McCord
Jaco Kroon
Jean Aunis
Joshua Colp
Ben Ford
Corey Farrell
Salah Ahmed
lvl
Torrey Searle
Florian Floimair
Igor Goncharovsky
Leonid Fainshtein
Tzafrir Cohen
Francesco Castellano
Chris-Savinovich
Dan Cropp
agupta
Alexei Gradinari
Salah Ahmed
cirillor
sungtae kim
Giuseppe Sucameli
Jasper Hafkenscheid
pk16208
hajekd
Richard Mudgett
ktyerman
Matthew Fredrickson
Oron Peled
Yasuhiko Kamata
Daniel Tryba
David J. Pryke
Scott Griepentrog
Vitezslav Novy
Andre Nazario