When in a conference bridge it may be necessary to have
text messages disabled for specific participants or for
all. This change adds a configuration option, "text_messaging",
which can be used to enable or disable this on the
user profile. By default existing behavior is preserved
as it defaults to "yes".
ASTERISK-28841
Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13
When the receive buffer was flushed by a received packet while it
already contained a packet with the same sequence number, Asterisk
never left the while loop which tried to order the packets.
This change makes it so if the packet is in the receive buffer it
is retrieved and freed allowing the buffer to empty.
ASTERISK-28827
Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3
When the ast_data_buffer_put rejects to add a packet, for example because
the buffer already contains a packet with the same sequence number, the
payload will never be freed, resulting in a memory leak.
The data buffer will now return an error if this situation occurs
allowing the caller to free the payload. The res_rtp_asterisk module
has also been updated to do this.
ASTERISK-28826
Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1
By using pjproject to give us a list of candidates, and then filtering,
if the host has >32 addresses configured, then it is possible that we
end up filtering out all 32 of those, and ending up with no candidates
at all. Instead, get getifaddrs (which pjsip is using underlying
anyway) to retrieve all local addresses, and iterate those, adding the
first 32 addresses not excluded by the ICE ACL.
In our setup at any point in time We've got between 6 and 328 addresses
on any given system. The lower limit is the lower limit but the upper
limit is growing on a near daily basis currently.
Change-Id: I109eaffc3e2b432f00bf958e3caa0f38cacb4edb
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
The Python/C API is used only if the Test Framework was enabled in Asterisk
'make menuselect'. The Test Framework is available only if the Developer Mode
was enabled in Asterisk './configure --enable-dev-mode'. And that Python/C API
is used only if the PJProject was found and not disabled in Asterisk; the user
did not go for './configure --without-pjproject'.
Furthermore, because version 2 of that Python/C API is required (currently) and
because some platforms do not offer a generic version 2, the script searches
for 2.7 explicitly as well.
To avoid version mismatch between the Python/C API and the Python environment,
the script searches for the latter in the same versions, in the same the order
as well. Because this Python/C API is just for (some) Asterisk contributors,
the script also goes for the Python 3 environment as a last resort for all
other Asterisk users. This allows 'make full' even on minimal installations of
Ubuntu 18.04 LTS and newer.
Because the Python/C API is Asterisk contributor specific, the Python packages
are removed from the script './contrib/scripts/install_prereq' as this script
is intended for Asterisk users. Asterisk contributors have to install much more
packages in any case, like:
sudo apt install autoconf automake git git-review python2.7-dev
ASTERISK-28824
ASTERISK-27717
Change-Id: Id46d357e18869f64dcc217b8fdba821b63eeb876
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
The following fields return an error when read from dialplan:
- exten
- context
- userfield
- channame
ASTERISK-28796 #close
Change-Id: Ieacaac629490f8710fdacc9de80ed5916c5f6ee2
This reverts commit a3a2fbaec6.
Reason for revert: There is a lot of code that relies on the broken
behavior that this fixes.
Change-Id: I410c395a0168acbdaf89e616e3cb5e1312d190cb
When an AOR is modified endpoints are updated that reference
the AOR so they can start receiving updates and reflect the
correct state. If this is the case then we shouldn't change
the endpoint to be offline if it does not reference the AOR
but instead only when the endpoint is completely updated for
all its AORs.
ASTERISK-28056
patches:
pjsip_options-aor.diff submitted by jhord (license 6978)
Change-Id: I3ee00023be2393113cd4e056599f23f3499ef164
If a frame handling routine returns a list of frames (vs. a single frame)
those frames are never passed to a tech's write_stream handler even if one is
available. For instance, if a codec translation occurred and that codec
returned multiple frames then those particular frames were always only sent
to the tech's "write" handler. If that tech (pjsip for example) was stream
capable then those frames were essentially ignored. Thus resulting in bad
audio.
This patch makes it so the "write_stream" handler is appropriately called
for all cases, and for all frames if available.
ASTERISK-28795 #close
Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d
The dial application had 80 characters of destination length
limitation. But this limitation causes unexpected dial string
cut if the dial string is long.
Removed unnecessary limited buffer to support longer dial
destination.
ASTERISK-27946
Change-Id: I72c8f0319a4b47e8180817a66a7e9bde063cb330
RFC5621 requires any content type with a Content-Disposition
with handling=required to be rejected with a 415 response
ASTERISK-28782 #close
Change-Id: Iad969df75936730254b95c1a8bc3b48497070bb4
named_acl.c (which is really a named_ha) now uses ast_ha_output.
I've also updated main/manager.c to output the actual ACL on "manager
show user <username>" if one is set. If this works then we can add
similar to other modules as required.
Change-Id: I0ec9876a90dddd379c80ec078d48e3ee6991eb0f
When an outgoing channel is created a list of formats may
optionally be provided which is used as a request that the
formats be used if possible. If an endpoint is not configured
for any of the formats we ignore this request and use what is
configured. This has the side effect of also including other
stream types (such as video) that were not present in the
requested formats.
This change makes it so that the intention of the request is
preserved - that is if only an audio format is requested then
even if there is no joint audio format between the request and
the configuration we will still only place an audio stream in
the outgoing call.
ASTERISK-28787
Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f
This patch makes it so ast_coredumper now outputs the following information to
a *-info.txt file when processing a core file:
asterisk version and "built by" string
BUILD_OPTS
system start, and last reloaded date/time
taskprocessor list
equivalent of "bridge show all"
equivalent of "core show channels verbose"
Also a slight modification was made when trying to obtain the pid(s) of a
running Asterisk. If it fails to retrieve any it now reports an error.
Change-Id: I54f35c19ab69b8f8dc78cc933c3fb7c99cef346b
This fixes ast_addressfamily_to_sockaddrsize to reference the
provided argument, and ast_sockaddr_from_sockaddr to not use the name of
a structure as argument.
Change-Id: Ibf5db469c47c3b4214edf8456326086174e8edd7
If a negative (error) return is received from dundi_lookup_internal,
this is not handled correctly when assigning the result to the buffer.
As such, use a signed integer in the assignment and do a proper
comparison.
ASTERISK-21205
Change-Id: I5214ebb6491e2bd14f90c7d3ce229da86888f739
When examining a stream to determine hold/unhold information we
only care about the default audio stream. Other streams aren't
used for hold/unhold.
ASTERISK-28784
Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4
When the Asterisk receives 200 OK with invalid SDP,
the Asterisk/PJPROJECT terminating the session.
But if the channel was in the Bridge, Asterisk tries send
the Re-Invite before terminating the session.
And when the Asterisk sending the Re-Invite, it doesn't check
the SDP is NULL or not. This crashes the Asterisk.
Fixed it to close the session correctly if the UAS sends the
200 OK with wrong SDP.
ASTERISK-28743
Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da
This patch has been included in Gentoo distribution for at least since
asterisk 1.8, but there are references in the logs going back as far as
1.0.0 - not sure if this is still required in any way, it does apply,
and it doesn't (as far as we can determine) cause build failures.
Change-Id: I46d8845e30200205e80580680bf060aa3012ba54
We (Gentoo distribution) reckon that in the case of multiple versions of
gmime installed we should prefer the newest one.
Change-Id: Idf7be613230232eb1d573d93c4a5a8297f4ecd2d
The state of the default audio stream is used for hold/unhold so we
restrict it to sendrecv as the core does not handle when it changes as
a result of hold/unhold.
This restriction does not apply to other media types though so we now
only restrict it to audio. This allows the other default streams to
store their state at all values, and not just sendrecv and removed.
ASTERISK-28783
Change-Id: I139740f38cea7f7d92a876ec2631ef50681f6625
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
