This patch fixes 2 original issues and more that those 2 exposed.
* When we send a NOTIFY, and the client either doesn't respond or
responds with a non OK, pjproject only calls our
pubsub_on_evsub_state callback, no others. Since
pubsub_on_evsub_state (which does the sub_tree cleanup) does not
expect to be called back without the other callbacks being called
first, it just returns leaving the sub_tree orphaned. Now
pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE
which is what pjproject will set to tell us that it was the
transaction that timed out or failed and not the subscription
itself timing our or being terminated by the client. If is
TSX_STATE, pubsub_on_evsub_state now does the proper cleanup
regardless of the state of the subscription.
* When a client renews a subscription, we don't update the
persisted subscription with the new expires timestamp. This causes
subscription_persistence_recreate to prune the subscription if/when
asterisk restarts. Now, pubsub_on_rx_refresh calls
subscription_persistence_update to apply the new expires timestamp.
This exposed other issues however...
* When creating a dialog from rdata (which sub_persistence_recreate
does from the packet buffer) there must NOT be a tag on the To
header (which there will be when a client refreshes a
subscription). If there is one, pjsip_dlg_create_uas will fail.
To address this, subscription_persistence_update now accepts a flag
that indicates that the original packet buffer must not be updated.
New subscribes don't set the flag and renews do. This makes sure
that when the rdata is recreated on asterisk startup, it's done
from the original subscribe packet which won't have the tag on To.
* When creating a dialog from rdata, we were setting the dialog's
remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq.
When the client tried to resubscribe after a restart with the
correct cseq, we'd reject the request with an Invalid CSeq error.
* The acts of creating a dialog and evsub by themselves when
recreating a subscription does NOT restart pjproject's subscription
timer. The result was that even if we did correctly recreate the
subscription, we never removed it if the client happened to go away
or send a non-OK response to a NOTIFY. However, there is no
pjproject function exposed to just set the timer on an evsub that
wasn't created by an incoming subscribe request. To address this,
we create our own timer using ast_sip_schedule_task. This timer is
used only for re-establishing subscriptions after a restart.
An earlier approach was to add support for setting pjproject's
timer (via a pjproject patch) and while that patch is still included
here, we don't use that call at the moment.
While addressing these issues, additional debugging was added and
some existing messages made more useful. A few formatting changes
were also made to 'pjsip show scheduled tasks' to make displaying
the subscription timers a little more friendly.
ASTERISK-26696
ASTERISK-26756
Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e
pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND,
which is a compile-time constant. Instead of hard-coding 16 when we
enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can
potentially collect more interfaces if the compile time options are
changed.
Tangentially related to ASTERISK~24464
Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd
OpenSSL 1.1 requires no explicit initialization. The hacks in the
library are not needed. They also happen to fail running Asterisk.
ASTERISK-26109 #close
Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100
Original patch by John Covert, slight modifications by me.
ASTERISK-17428 #close
Reported by: John Covert
Patches:
app_voicemail.c.patch (license #5512) patch uploaded by
John Covert
Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6
OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous
version-specific methods (such as TLSv1_client_method(). Other than
being simpler to use and more correct (gain support for TLS newer that
TLS1, in our case), the older ones produce a deprecation warning that
fails the build in dev-mode.
ASTERISK-26109 #close
Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07
Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect
the openssl 1.1 API.
ASTERISK-26109 #close
Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458
When attempting to use VoiceMailPlayMsg with a realtime data backend
the message is located, but never retrieved. This patch adds the
required RETRIEVE and DISPOSE calls that will fetch the message from
the database (and IMAP storage as well for that matter).
Also, removed extraneous make_file call.
ASTERISK-26723 #close
Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c
When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.
This patch adds the 'u' option to Record() to override that behavior.
ASTERISK-18286 #close
Reported by: var
Patches:
app_record-1.8.7.1.diff (license #6184) patch uploaded by var
Change-Id: Ia1cd163483235efe2db05e52f39054288553b957
* app_minivm: Use built-in completion facilities to complete optional
arguments.
* app_voicemail: Use built-in completion facilities to complete
optional arguments.
* app_confbridge: Add missing colons after 'Usage' text.
* chan_alsa: Use built-in completion facilities to complete optional
arguments.
* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'
* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'
* func_odbc: Correct completions for 'odbc read' and 'odbc write'
* main/asterisk: Correct and extend completions for 'core show file
version.'
* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.
* main/bridge: Correct completions for 'bridge kick.'
* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.
* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'
* main/pbx_app: Remove redundant completions for 'core show
applications.'
* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'
* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.
Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com),
the user part is stripped down as it would be a trunk with a specified user,
and only the host part is called as a PJSIP endpoint and can't be found.
This is not correct in the case of a multidomain SIP account, so the stripping
after the @ sign is done only if the whole endpoint (in multidomain case
1000@test.com) can't be found.
ASTERISK-26248
Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6
The ast_waitfor_nandfds operation will manipulate the flags
of channels passed in. This was previously done without
the channel lock being held. This could result in incorrect
values existing for the flags if another thread manipulated
the flags at the same time.
This change locks the channel during flag manipulation.
ASTERISK-26788
Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed
The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE
but we have no authenticator registered to create the challenge.
Change-Id: I62368180d774b497411b80fbaabd0c80841f8512
In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c49. This patch
restores that behavior.
This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.
ASTERISK-26115 #close
Reported by: Nasir Iqbal
Change-Id: I8afbd06725e99610e02adb529137d4800c05345d
We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.
* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging. Made hold the channel lock after the called
party answers while updating the caller channel staging.
* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.
* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.
* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.
Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a
The entry for 'identify' was incorrectly placed in the
res_pjsip section when it should be in
res_pjsip_endpoint_identifier_ip.
ASTERISK-26785 #close
Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a
This reverts commit 6492e91392.
The change in question was intended to prevent the need to reload in
order to update qualifies on contacts when an AOR changes. However, this
ended up causing a deadlock instead.
Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e
When performing an SRV lookup using the ast_srv_lookup function it
did not properly handle the situation where 0 records are returned.
If this happened it would wrongly assume that at least one record
was present.
This change fixes the code so it will exit early if an error occurs
or if 0 records are returned.
ASTERISK-26772
patches:
srv_lookup.patch submitted by nappsoft (license 6822)
Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351
The adding and removing of device state subscriptions did not protect
fully against simultaneous manipulation. In particular the subscribe
case allowed a small window where two subscriptions could be added for
the same device state instead of just one.
This change makes the code hold the subscriptions lock for the entirety
of each operation to ensure that two are not occurring at the same time.
ASTERISK-26770
Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b
If an audiohook is placed on a channel that does not require transcoding,
muting that hook will cause the underlying frames to be muted as well.
The original patch is from David Woolley but I have modified slightly.
ASTERISK-21094 #close
Reported by: David Woolley
Patches:
ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded
by David Woolley
Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed
Prior to this change, qualifies would only update in the following
cases:
* A reload of res_pjsip.so was issued.
* A dynamic contact was re-registered after its AOR's qualify_frequency
had been changed
This does not work well if you are using realtime for your AORs. You can
update your database to have a new qualify_frequency, but the permanent
contacts on that AOR will not have their qualifies updated. And the
dynamic contacts on that AOR will not have their qualifies updated until
the next registration, which could be a long time.
This change seeks to fix this problem by making it so that whenever AOR
configuration is applied, the contacts pertaining to that AOR have their
qualifies updated.
Additions from this patch:
* AOR sorcery objects now have an apply handler that calls into a newly
added function in the OPTIONS code. This causes all contacts
associated with that AOR to re-schedule qualifies.
* When it is time to qualify a contact, the OPTIONS code checks to see
if the AOR can still be retrieved. If not, then qualification is
canceled on the contact.
Alterations from this patch:
* The registrar code no longer updates contact's qualify_frequence and
qualify_timeout. There is no point to this since those values already
get updated when the AOR changes.
* Reloading res_pjsip.so no longer calls the OPTIONS initialization
function. Reloading res_pjsip.so results in re-loading AORs, which
results in re-scheduling qualifies.
Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121
Using the timerfd timing module can cause channel freezing, lingering, or
deadlock issues. The problem is because this is the only timing module
that uses an associated alert-pipe. When the alert-pipe becomes
unbalanced with respect to the number of frames in the read queue bad
things can happen. If the alert-pipe has fewer alerts queued than the
read queue then nothing might wake up the thread to handle received frames
from the channel driver. For local channels this is the only way to wake
up the thread to handle received frames. Being unbalanced in the other
direction is less of an issue as it will cause unnecessary reads into the
channel driver.
ASTERISK-26716 is an example of this deadlock which was indirectly fixed
by the change that found the need for this patch.
* In channel.c:__ast_queue_frame(): Adding frame lists to the read queue
did not add the same number of alerts to the alert-pipe. Correspondingly,
when there is an exceptionally long queue event, any removed frames did
not also remove the corresponding number of alerts from the alert-pipe.
ASTERISK-26632 #close
Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6
A dialplan intercept routine is equivalent to an interrupt routine. As
such, the routine must be done quickly and you do not have access to the
media stream. These restrictions are necessary because the media stream
is the responsibility of some other code and interfering with or delaying
that processing is bad. A possible future dialplan processing
architecture change may allow the interception routine to run in a
different thread from the main thread handling the media and remove the
execution time restriction.
* Made res_agi.c:run_agi() running an AGI in an interception routine run
in DeadAGI mode. No touchy channel frames.
ASTERISK-25951
ASTERISK-26343
ASTERISK-26716
Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43
There are several issues with deferring frames that are caused by the
refactoring.
1) The code deferring frames mishandles adding a deferred frame to the
deferred queue. As a result the deferred queue can only be one frame
long.
2) Deferrable frames can come directly from the channel driver as well as
the read queue. These frames need to be added to the deferred queue.
3) Whoever is deferring frames is really only doing the __ast_read() to
collect deferred frames and doesn't care about the returned frames except
to detect a hangup event. When frame deferral is completed we must make
the normal frame processing see the hangup as a frame anyway. As such,
there is no need to have varying hangup frame deferral methods. We also
need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real.
That fake hangup is to cause the PBX thread to break out of loops to go
execute a new dialplan location.
4) To properly deal with deferrable frames from the channel driver as
pointed out by (2) above, means that it is possible to process a dialplan
interception routine while frames are deferred because of the
AST_CONTROL_READ_ACTION control frame. Deferring frames is not
implemented as a re-entrant operation so you could have the unsupported
case of two sections of code thinking they have control of the media
stream.
A worse problem is because of the bad implementation of the AMI PlayDTMF
action. It can cause two threads to be deferring frames on the same
channel at the same time. (ASTERISK_25940)
* Rather than fix all these problems simply revert the API refactoring as
there is going to be only autoservice and safe_sleep deferring frames
anyway.
ASTERISK-26343
ASTERISK-26716 #close
Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496
There are some error cases in PJSIP when sending a request that will
result in the callback for the request being invoked. The code did not
handle this case and assumed on every error case that the callback was not
invoked.
The code has been changed to check whether the callback has been invoked
and if so to absorb the error and treat it as a success.
ASTERISK-26679
ASTERISK-26699
Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91
Before Asterisk 13, signed linear was converted into network byte order by a
smoother before being sent over the network. We restore this behavior by
forcing the creation of a smoother when slinear is in use and setting the
appropriate flags so that the byte order conversion is always done.
ASTERISK-24858 #close
Reported-by: Frankie Chin
Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9
