When fixing issues uncovered by GCC10 a copy of the parker UUID
was removed accidentally. This change restores it so that the
subscription has the data it needs.
ASTERISK-29042
Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a
The name of the voicemail context was overwriting the name of the
subscribed mailbox. Fix by simplifying how we create the MWI
subscription.
ASTERISK-29029 #close
Change-Id: Ie8a7db6a0b68f3995b0846bbb733a21909ba44e5
When the ExtensionState AMI action is executed on a pattern matched
hint it can end up adding a new hint if one does not already exist.
This results in a locking order of contexts -> hints -> contexts.
If at the same time a reload is occurring and adding its own hint
it will have a locking order of hints -> contexts.
This results in a deadlock as one thread wants a lock on contexts
that the other has, and the other thread wants a lock on hints
that the other has.
This change enforces a hints -> contexts locking order by explicitly
locking hints in the places where a hint is added when queried for.
This matches the order seen through normal adding of hints.
ASTERISK-29046
Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504
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
Currently, if the bridge has created by the ARI, the video_mode
parameter was
not shown in the BridgeCreated event correctly.
Fixed it and added video_mode shown in the 'bridge show <bridge id>'
cli.
ASTERISK-28987
Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295
The MoH class internal file vector is potentially being manipulated by
multiple threads at the same time without sufficient locking. Switch to
a reference counted list and operate on copies where necessary.
ASTERISK-28927 #close
Change-Id: I479c5dcf88db670956e8cac177b5826c986b0217
The assumed behavior of realloc() - that it was effectively a free() if
its second argument was 0 - is Linux specific behavior and is not
guaranteed by either POSIX or the C specification.
Instead, if we want to resize a vector to 0, do it explicitly.
Change-Id: Ife31d4b510ebab41cb5477fdc7ea4e3138ca8b4f
PJSIP, UDP transport with external_media_address and session timers
enabled. Connected to SIP server that is not in local net. Asterisk
initiated the connection and is refreshing the session after 150s
(timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has
a malformed IP address in its SDP (garbage string). This only happens
when the SDP is modified by the nat-code to replace the local IP address
with the configured external_media_address.
Analysis: the code to modify the SDP (in
res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?)
in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses
the tdata->pool to allocate the replacement string. But the same
pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also
used for the 2nd refresh-INVITE (because it is stored in pjmedia's
pjmedia_sdp_neg structure). The problem is, that at that moment, the
tdata->pool that holds the stringified external_media_address from the
1. refresh-INVITE has long been reused for something else.
Fix by Sauw Ming of pjproject (see
https://github.com/pjsip/pjproject/pull/2476): the local, potentially
modified pjmedia_sdp_stream is cloned in
pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the
clone is stored, thereby detaching from the tdata->pool (which is only
released *after* process_answer())
ASTERISK-28973
Reported-by: Michael Neuhauser
Change-Id: I272ac22436076596e06aa51b9fa23fd1c7734a0e
Statically configured contacts on an AOR don't have an expiration
time so when adding them to the resulting 200 OK if an endpoint
registers ensure they are marked as such.
ASTERISK-28995
Change-Id: I9f0e45eb2ccdedc9a0df5358634a19ccab0ad596
There are various places in Asterisk - specifically in regards to
database integration - where having some kind of UTF-8 validation would
be beneficial. This patch adds:
* Functions to validate that a given string contains only valid UTF-8
sequences.
* A function to copy a string (similar to ast_copy_string) stopping when
an invalid UTF-8 sequence is encountered.
* A UTF-8 validator that allows for progressive validation.
All of this is based on the excellent UTF-8 decoder by Björn Höhrmann.
More information is available here:
https://bjoern.hoehrmann.de/utf-8/decoder/dfa/
The API was written in such a way that should allow us to replace the
implementation later should we determine that we need something more
comprehensive.
Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9
If an ACL is misconfigured in the realtime database (for instance, the
"rule" is blank) and Asterisk attempts to read the ACL, Asterisk will
crash.
ASTERISK-28978 #close
Change-Id: Ic1536c4df856231bfd2da00128f7822224d77610
I noticed this while looking at another issue and brought
it up with Teluu. It was possible for an uninitialized timer
to be cancelled, resulting in the invalid timer id of 0
being placed into the timer heap causing issues.
This change is a backport from the pjproject repository
preventing this from happening.
Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7
We read beyond the end of the buffer when copying the string out of the
buffer when we used ast_copy_string() because the original string was
not null terminated. Instead switch to ast_strndup() which does not
exhibit the same behavior.
ASTERISK-28975 #close
Change-Id: Ib4a75cffeb1eb8cf01136ef30306bd623e531a2a
Reminder: 13 doesn't support scope tracing but we're keeping
the stub defines consistent across branches.
Change-Id: If00733a4d5a4e829cbb4c934bcc2f4c1a80dab81
Currently when the pjsip making an outgoing request, it keep adding the
rport parameter in a request message as a default.
This causes unexpected rport handle at the other end.
Added option for disable this behaviour in the pjsip.conf.
This is a system option, but working as a gloabl option.
ASTERISK-28959
Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc
Do not return error if the client received ping frame
while looking for a string and just wait for another frame.
ASTERISK-28958 #close
Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922
When using the PSJIP_MEDIA_OFFER dialplan function it was not
overriding an endpoint's configured codecs 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 definition no matter what.
ASTERISK-28878 #close
Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6
This patch allows a user of AMI to now specify the type of message
content contained within by setting the 'Content-Type' parameter.
Note, the AMI version has been bumped for this change.
ASTERISK-28945 #close
Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb
Although not implemented in the 13 branch, we need to keep the
defines consistent across all branches.
Change-Id: Id88ad71ca36818a35f1f14a5acb450e240366301
The outbound proxy for an AOR was not being applied to
any statically configured Contacts. This resulted in the
OPTIONS requests being sent to the wrong target.
This change sets the outbound proxy on statically configured
contacts once the AOR configuration is done being
applied.
ASTERISK-28965
Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0
Given a scenario where a module has a dependency on both
an external library and a module if the external library was
available and the module was not an infinite loop would
occur. This happened due to the code changing the dependecy
status to no failure on each dependency checking loop
iteration, resulting in the code thinking that it had
gone from no failure to failure each time triggering another
dependency check.
This change makes it so that the old dependency status is
preserved throughout the dependency checking allowing it to
determine that after the first iteration the dependency
status does not transition from no failure to failure.
ASTERISK-28930
Change-Id: Iea06d45d9fd6d8bfd068882a0bb7e23a53ec3e84
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
1) Fix memory-leaks
Added code to release ast_events extracted from corosync and stasis messages
2) Clean stasis cache when a member of the corosync cluster leaves the group
Added code to remove from the stasis cache of the members remained on the
group all the messages with the EID of the left member.
If the device states of the left member remain in the stasis cache of other
members, they will not be updated anymore and high priority cached values,
like BUSY, will take precedence over current device states.
3) Stop corosync event propagation when node is not joined to the group
Updated dispatch_thread_handler code to detect when asterisk is not joined
to the corosync group and added some condition in publish_event_to_corosync
code to send corosync messages only when joined.
When a node is not joined its corosync daemon can't send messages:
the cpg_mcast_joined function append new messages to the FIFO buffer until
it's full and then it blocks indefinitely.
In this scenario if the stasis_message_cb callback, registered by
res_corosync to handle stasis messages, try to send a corosync messages,
the thread of the stasis thread-pool will be blocked until the node join
the corosync cluster.
ASTERISK-28888
Reported by: Università di Bologna - CESIA VoIP
Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2
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
The AMI action and CLI command did not take into account the properties
of full backend caching. This resulted in an expired object remaining
removed until a full backend update occurred, instead of having the
object updated when needed.
This change makes it so that the AMI action and CLI command for object
expire will now fail instead of putting the cache into an undesired
state. If full backend caching is enabled then only operations
which act on the entire cache are available.
ASTERISK-28942
Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4
Before this changeset, it was possible that a queue member (agent) was
called even though they just got out of a call, and wrapuptime seconds
hadn't passed yet.
This could happen if a member ended a call _between_ a new call attempt
and asterisk trying that particular member for a new call.
In that case, Asterisk would check the hangup time of the
call-before-the-last-call instead of the hangup time of the-last-call.
ASTERISK-28952
Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3
Because ring_entry() is not called, outgoing->chan is not touched here
either.
ASTERISK-28950
ASTERISK-28644
Change-Id: I564613715dfaf45af868251eb75a451f512af90f
This patch makes the usual necessary changes when upgrading to a new
version pjproject. For instance, version number bump, patches removed
from third-party, new *.md5 file added, etc..
This patch also includes a change to the Asterisk pjproject Makefile to
explicitly create the 'source/pjsip-apps/lib' directory. This directory
is no longer there by default so needs to be added so the Asterisk
malloc debug can be built.
This patch also includes some minor changes to Asterisk that were a result
of the upgrade. Specifically, there was a backward incompatibility change
made in 2.10 that modified the "expires header" variable field from a
signed to an unsigned value. This potentially effects comparison. Namely,
those check for a value less than zero. This patch modified a few locations
in the Asterisk code that may have been affected.
Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to
check a minimum version of pjproject at compile time.
ASTERISK-28899 #close
Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81
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
This patch fixes a few compile warnings/errors that now occur when using gcc
10+.
Also, the Makefile.rules check to turn off partial inlining in gcc versions
greater or equal to 8.2.1 had a bug where it only it only checked against
versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures
any version above the specified version is correctly compared.
Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9
You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri,
without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S).
ASTERISK-28936
Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a
In a particular fax gateway scenario whereby it would
have to translate using the read translation path on a
channel the frame being translated would be consumed.
When the frame is in the write path it is not permitted
to free the frame as the caller expects it to continue
to exist.
This change makes it so that the frame is only consumed
on the read path where it is acceptable to free it.
ASTERISK-28900
Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d
When writing tx messages to pcap files, Asterisk is using the wrong
pointer resulting in lots of wasted space. This patch fixes it to use
the correct pointer.
ASTERISK-28932 #close
Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23
Scope tracing isn't supported in Asterisk 13 due to changes made
to logging between 13 and 16 but since the scope tracing macros
may be present in the 16, 17 and master branches, those macros
are defined here as NOOPs so cherry-picking changes downward
to 13 can still be seamless.
Change-Id: I0390ce5651374d8f3e06d7620050acb22c5440a2
The destructor for RTP deallocated transport resources
before terminating the ICE support. This could result
in a crash as the thread handling ICE would access already
freed parts of the RTP data.
This change re-orders the destruction so that ICE is
stopped before destroying things.
ASTERISK-28885
Change-Id: Ie71add549f12b6cdea7e9dbf976d1bd1d2fc0bdc
files
fwrite() does return the number of elements written and not the
number of bytes. However asterisk is currently comparing the return
value to the size of the written element what means that asterisk logs
five WARNING messages on every packet written to the pcap file.
This patch changes the code to check for the correct value, which will
always be 1.
ASTERISK-28921 #close
Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2
When replacing the user portion of the Contact URI the code
was using the ephemeral pool instead of the tdata pool. This
could cause the Contact user value to become invalid after a
period of time.
The code will now use the tdata pool which persists for the
lifetime of the message instead.
ASTERISK-28794
Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5
While asterisk is filtering out the x-ast-orig-host parameter from the
contact on response messages, it is not filtering it out from the
request URI and the to header on SIP requests (for example INVITE).
ASTERISK-28884 #close
Change-Id: Id032b33098a1befea9b243ca994184baecccc59e
When a configuration file in Asterisk is loaded
information about it is stored such that on a
reload it is not reloaded if nothing has changed.
This can be problematic when an error exists in
a configuration file in PJSIP since the error
will be output at start and not subsequently on
reload if the file is unchanged.
This change makes it so that if an error is
encountered when res_sorcery_config is loading
a configuration file a reload will always read
in the configuration file, allowing the error
to be seen easier.
Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed
When receiving audio from a channel we determine if it
is talking or silence based on a threshold value. If
this threshold is met we always mix the audio into the
conference bridge. If this threshold is not met we also
mix the audio into the conference bridge UNLESS the
drop silence option is enabled.
The code that removed the audio from the mixed frame
assumed that it was always not present if it did not
meet the threshold to be considered talking. This is
incorrect. If it has been stated that the audio was
mixed into the mixed frame then it has been mixed into
the mixed frame. By not removing audio that was
considered non-talking it was possible for a channel
to receive a slight echo of audio of itself at times.
This change ensures that the audio is always removed
from the mixed frame going back to the channel so it
no longer receives the slight echo.
ASTERISK-28898
Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb
