Different clients react differently to being told that a blind transfer
has failed. Some will simply send a BYE and be done with it. Others will
attempt to reinvite themselves back onto the call.
In the latter case, we were creating a new channel and then leaving it to
sit forever doing nothing. With this code change, that new channel will
not be created and the dialog with the transferring channel will be cleaned
up properly.
ASTERISK-24624 #close
Reported by Zane Conkle
Review: https://reviewboard.asterisk.org/r/4339
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When FAX was developed, apparently the faxregistry.container used to be a
linked list that was converted to an ao2 container. Some of the
replacement ao2 container operations still had explicit lock/unlocks
around them.
Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the
channel even though the routine did not lock the channel and other code
paths in the routine do not unlock the channel.
Review: https://reviewboard.asterisk.org/r/4340/
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Due to the split of outbound registration state from configuration it is possible during
a reload for a "pjsip show registrations" CLI command to be executed which gets an older
snapshot of the configuration. This configuration may include outbound registrations which
have been removed due to a reload operation occurring at the same time. The code for
printing the outbound registration did not take this into account but now it does.
AST-1506 #close
Review: https://reviewboard.asterisk.org/r/4338/
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The res_pjsip module was previously unloadable. With this patch it can now
be unloaded.
This patch is based off the original patch on the issue (listed below) by Corey
Farrell with a few modifications. Namely, removed a few changes not required to
make the module unloadable and also fixed a bug that would cause asterisk to
crash on unloading.
This patch is the first step (should hopefully be followed by another/others at
some point) in allowing res_pjsip and the modules that depend on it to be
unloadable. At this time, res_pjsip and some of the modules that depend on
res_pjsip cannot be unloaded without causing problems of some sort.
The goal of this patch is to get res_pjsip and only res_pjsip to be able to
unload successfully and/or shutdown without incident (crashes, leaks, etc...).
Other dependent modules may still cause problems on unload.
Basically made sure, with the patch applied, that res_pjsip (with no other
dependent modules loaded) could be succesfully unloaded and Asterisk could
shutdown without any leaks or crashes that pertained directly to res_pjsip.
ASTERISK-24485 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4311/
patches:
pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909)
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The code was missing the case for explicitly destroying an outbound publication
when Asterisk had never actually published anything. The result was that Asterisk
would hang for a while on a graceful shutdown.
With this change, the case is taken into account, and on a graceful shutdown, these
publications are destroyed without the need to actually send a PUBLISH request.
ASTERISK-24655 #close
Reported by Kevin Harwell
Review: https://reviewboard.asterisk.org/r/4325
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* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
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If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't
survive. If you do a 'core (shutdown|restart) now' or asterisk terminates for
some reason, they do. Here's why...
When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to
subscribers for each subscription. This not only tells the subscribers that the
dialog/state machine is done, it also frees the last reference to the
subscription tree which causes the persistent subscription to get deleted from
astdb. When asterisk restarts, nothing's left. Just preventing the delete from
astdb doesn't work because we already told the subscriber to terminate the
dialog so we can't restart it even if it was still in astdb. Everything works
OK if asterisk terminates unexpectedly because we never send the 'terminated'
message so on restart, the subscription is still in astdb and the subscriber is
none the wiser.
This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for
persistent connections.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4318/
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Every time a registration started, sip_outbound_registration_response_cb bumps
the ref count on client_state then pushes a handle_registration_response task.
handle_registration_response never unreffed it though. So every time a
registration goes out, the ref count goes up by one.
This patch adds the unreffs to handle_registration_response.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4303/
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There are 2 issues with reloading registrations...
1. The 'can_reuse_registration' test wasn't considering the intervals or
expiration in its determination of whether a registration changed or not so if
you changed any of the intervals or the expiration and reloaded, the object
would get reloaded but the actual timers wouldn't change.
can_reuse_registration now does a sorcery diff on the old and new objects
instead of discretely testing certain fields. Now if you change expiration for
instance, and reload, the timer is updated and re-registration will occur on the
new value.
2. If you mung up your password on an outbound registration you get a permanent
failure. If you fix the password (on the outbound_auth object) and reload,
nothing tells outbound_registration to try again because the registration itself
didn't change. This patch adds an observer on the "auth" object type and if any
auth changes, existing registration states are searched and those in a
REJECTED_PERMANENT state are retried.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4304/
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When the AMI Redirect action is used with a channel bridged inside
Stasis() and not running a pbx, the channel is hung up instead of
proceeding to the desired location in dialplan. This change allows
such channels to be Redirected properly by detecting the operation
used by Redirect (ASYNCGOTO) and using the code already established
for functionality of the ARI channel continue operation.
ASTERISK-24591 #close
Review: https://reviewboard.asterisk.org/r/4271/
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With this patch, the following two ARI commands
POST /channels
POST /channels/{id}/continue
Accept a new parameter, label, that can be used to continue to or originate
to a priority label in the dialplan.
Because this is adding a new parameter to ARI commands, the API version of
ARI has been bumped from 1.6.0 to 1.7.0.
This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks!
ASTERISK-24412 #close
Reported by Nir Simionovich
Review: https://reviewboard.asterisk.org/r/4285
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This fix has two parts:
* Corrected an error message to properly state that external_replaces is an extension. The
error message also prints what dialplan context the external_replaces extension was being
looked for in.
* Corrected the printing of the Replaces: header in an INVITE request. We were duplicating
"Replaces: " in the header.
ASTERISK-24376 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/4296
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When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi,
if a contact hasn't registered yet, res_pjsip_mwi spits out a warning.
This is a perfectly normal situation though and doesn't require something
as serious as a warning. It's also self correcting. The device will start
getting mwi as soon as it registers.
This patch changes the warning to a notice.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4314/
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The current behavior of 'pjsip send unregister' is to send the unregister
(REGISTER with 0 exp) but let the next scheduled register proceed normally.
I don't think that's a good idea. If you unregister, it should stay
unregistered until you decide to start registrations again. So this patch
just adds a cancel_registration call to the current unregister_task to
cancel the timer.
Of course, now you need a way to start registration again so I've added
a 'pjsip send register' command that unregisters and cancels any existing
registration (the same as send unregister), then sends an immediate
registration and starts the timer back up again.
Both changes also ripple to AMI. There's a new PJSIPRegister command.
There's no harm in calling either command repeatedly. They don't care
about the actual state.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4301/
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A native rtp bridge was being chosen (it shouldn't have been) when using two
pjsip channels with incompatible DTMF modes. This patch sets the rtp instance
property, AST_RTP_PROPERTY_DTMF, for the appropriate DTMF mode(s) for pjsip.
It was not being set before, meaning all DTMF modes for pjsip were being treated
as compatible, thus native bridging would be chosen as the bridge type when it
shouldn't have been.
ASTERISK-24459 #close
Reported by: Yaniv Simhi
Review: https://reviewboard.asterisk.org/r/4265/
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Prior to this patch, Asterisk would always respond to 401 responses to
registration attempts by trying to provide a registration with authentication
credentials. Even if subsequent attempts were rejected with 401 responses,
Asterisk would continue this behavior. If authentication credentials were
incorrect, this could continue forever.
With this patch, we keep track of whether we have attempted authentication
on an outbound registration attempt. If we already have, we don not try
again until the next attempt. This prevents the infinite loop scenario.
Review: https://reviewboard.asterisk.org/r/4273
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In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:
-out += sprintf(out, "%%%02X", (unsigned char) *ptr);
+out += sprintf(out, "%%%02X", (unsigned) *ptr);
That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.
This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)
Review: https://reviewboard.asterisk.org/r/4263/
ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
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If a remote endpoint reinvites to T.38 immediately the state machine
will go into a peer reinvite state. If a T.38 capable application
(such as ReceiveFax) queries it will receive this state. Normally
the application will then indicate so that the channel driver will
queue up the T.38 offer previously received. Once it receives this
offer the application will act normally and negotiate.
The res_pjsip_t38 module incorrectly partially squashed this indication.
This would cause the application to think the request had failed when
in reality it had actually worked.
This change makes it so that no T.38 control frames (or indications)
are squashed.
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res_pjsip_config_wizard
------------------
* This is a new module that adds streamlined configuration capability for
chan_pjsip. It's targetted at users who have lots of basic configuration
scenarios like 'phone' or 'agent' or 'trunk'. Additional information
can be found in the sample configuration file at
config/samples/pjsip_wizard.conf.sample.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4190/
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The Verbose message displayed when a file is played back via 'stream file'
was formatted differently than other playbacks:
* It didn't include the channel name
* It didn't include the channel language
It does, however, include the playback offset as well as any escape digits.
That information was kept; however, this patch updates the formatting to more
closely match the Verbose messages displayed when a file is played back by
'control stream file', Playback, ControlPlayback, or any other file playback
operation.
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Given the scenario where a PJSIP channel is in a native RTP bridge with direct
media and the channel is then hung up the code will currently re-INVITE the channel
back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
this greatly.
This change makes it so that if a re-INVITE transaction is in progress the BYE
is queued to occur after the completion of the transaction (be it through normal
means or a timeout).
Review: https://reviewboard.asterisk.org/r/4248/
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In the past the SDP negotiation within res_pjsip_session was made more tolerant of
certain situations. The only case where SDP negotiation will fail is when a major
error occurs during negotiation. Receiving an already declined media stream is
not considered a major error.
When producing the local SDP the logic took this into account so on the initial INVITE
the declined media stream did not cause an SDP negotiation failure. Unfortunately
the logic for handling media streams with a handler did not mirror this logic and
considered an already declined media stream an error and thus failed the SDP
negotiation.
This change makes the logic between both situations match so only under major
errors will the SDP negotiation fail.
ASTERISK-24607 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4254/
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When endpoints with direct_media enabled, behind a firewall (Asterisk on a
separate network) and were bridged sometimes Asterisk would send the ip
address of the firewall in the sdp to one of the phones in the reinvite
resulting in one way audio. When sending the reinvite Asterisk will retrieve
the media address from the associated rtp instance, but if frames were being
read this can be overwritten with another address (in this case the
firewall's). This patch ensures that Asterisk uses the original device
address when using direct media.
ASTERISK-24563
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4216/
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When using a non-default sorcery wizard (in this instance realtime) for outbound
publishes Asterisk will crash after a stack overflow occurs due to the code
infinitely recursing. The fix entails removing the outbound publish state
dependency from the outbound publish sorcery object and instead keeping an in
memory container that can be used to lookup the state when needed.
ASTERISK-24514 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4178/
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The reviewboard description does a fine job of summarizing this, so here it is:
A reporter discovered that Asterisk would crash when attempting to retransmit
a reinvite that had previously received a 491 response. The crash occurred
because a pjsip_tx_data structure was being saved for reuse, but its reference
count was not being increased. The result was that the pjsip_tx_data was being
freed before we were actually done with it. When we attempted to re-use the
structure when re-sending the reinvite, Asterisk would crash.
The fix implemented here is not to try holding onto the pjsip_tx_data at all.
Instead, when we reschedule sending the reinvite, we create a brand new
pjsip_tx_data and send that instead. Because of this change, there is no need
for an ast_sip_session_delayed_request structure to have a pjsip_tx_data on
it any more. So any code referencing its use has been removed.
When this initial fix was introduced, I encountered a second crash when
processing a subsequent 200 OK on a rescheduled reinvite. The reason was
that when rescheduling the reinvite, we gave the wrong location for a
response callback. This has been fixed in this patch as well.
ASTERISK-24556 #close
Reported by Abhay Gupta
Review: https://reviewboard.asterisk.org/r/4233
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This corrects several bugs that currently exist in the stasis
application code.
* After a masquerade, the resulting channels have channel topics that
do not match their uniqueids
** Masquerades now swap channel topics appropriately
* StasisStart and StasisEnd messages are leaked to observer
applications due to being published on channel topics
** StasisStart and StasisEnd publishing is now properly restricted
to controlling apps via app topics
* Race conditions exist where StasisStart and StasisEnd messages due to
a masquerade may be received out of order due to being published on
different topics
** These messages are now published directly on the app topic so this
is now a non-issue
* StasisEnds are sometimes missing when sent due to masquerades and
bridge swaps into and out of Stasis()
** This was due to StasisEnd processing adjusting message-sent flags
after Stasis() had already exited and Stasis() had been re-entered
** This was corrected by adjusting these flags prior to sending the
message while the initial Stasis() application was still shutting
down
Review: https://reviewboard.asterisk.org/r/4213/
ASTERISK-24537 #close
Reported by: Matt DiMeo
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When repeatedly starting/stopping a Monitor on a channel, the accumulated
in/out sample counts are never reset to 0. This can cause inadvertent jumps
in the recordings, as the code in the channel core will determine incorrectly
that a jump in the recorded file position should occur. Setting the sample
counts to 0 simply reflects the initial state a Monitor should be in when it
is started, as this is the initial count that would be on the channels at that
time.
ASTERISK-24573 #close
Reported by: Nuno Borges
patches:
24573.patch uploaded by Nuno Borges (License 6116)
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There are two methods within res_pjsip_refer for keeping track of the state of a transfer.
The first is a framehook which looks at frames passing by to determine the state. The second
subscribes to know when the channel joins a bridge. In the case when the channel joins the
bridge the framehook is *NOT* removed and this prevents the native RTP bridging technology
from getting used.
This change gets the channel and if it still exists remove the framehook.
Review: https://reviewboard.asterisk.org/r/4218/
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Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime
was tossing database fields that didn't have an exact match to a sorcery
registered field. This broke the ability to use regexes as field names which
manifested itself as a failure of res_pjsip_phoneprov_provider which uses
this capability. It also broke handling of fields that start with '@' in
realtime but I don't think anyone noticed.
This patch does the following...
* Modifies ast_sorcery_fields_register to pre-compile the name regex.
* Modifies ast_sorcery_is_object_field_registered to test the regex if it
exists instead of doing an exact strcmp.
* Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work
with realtime.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4185/
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The implementation of INVITE with Replaces in res_pjsip_refer did not expect them to
occur in-dialog. As a result it would incorrectly attempt to hang up a channel it
thought was under its control. In reality the channel would be under the control of
another thread. When the other thread accessed the channel it would be accessing freed
memory and could crash.
This change makes res_pjsip_refer not act on an in-dialog INVITE with Replaces.
ASTERISK-24528 #close
Reported by: Joshua Colp
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This change enforces the requirements in PJSIP for session timer configuration. The minimum
expiration period must be 90 seconds or higher and the normal expiration period can not
be lower than the minimum expiration period. If either of these were done the code would
assert at session setup time.
ASTERISK-24336 #close
Reported by: Leon Rowland
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From reviewboard:
"During blind transfer testing, it was noticed that tests were failing
occasionally because the ARI blind transfer event was not being sent.
After investigating, I detected a race condition in the blind transfer
code. When blind transferring a single channel, the actual transfer
operation (i.e. removing the transferee from the bridge and directing
them to the proper dialplan location) is queued onto the transferee
bridge channel. After queuing the transfer operation, the blind transfer
Stasis message is published. At the time of publication, snapshots of
the channels and bridge involved are created. The ARI subscriber to the
blind transfer Stasis message then attempts to determine if the bridge
or any of the involved channels are subscribed to by ARI applications.
If so, then the blind transfer message is sent to the applications. The
way that the ARI blind transfer message handler works is to first see
if the transferer channel is subscribed to. If not, then iterate over
all the channel IDs in the bridge snapshot and determine if any of
those are subscribed to. In the test we were running, the lone
transferee channel was subscribed to, so an ARI event should have been
sent to our application. Occasionally, though, the bridge snapshot did
not have any channels IDs on it at all. Why?
The problem is that since the blind transfer operation is handled by a
separate thread, it is possible that the transfer will have completed and
the channels removed from the bridge before we publish the blind transfer
Stasis message. Since the blind transfer has completed, the bridge on
which the transfer occurred no longer has any channels on it, so the
resulting bridge snapshot has no channels on it. Through investigation of
the code, I found that attended transfers can have this issue too for the
case where a transferee is transferred to an application."
The fix employed here is to decouple the creation of snapshots for the transfer
messages from the publication of the transfer messages. This way, snapshots
can be created to reflect what they are at the time of the transfer operation.
Review: https://reviewboard.asterisk.org/r/4135
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When using a non-default sorcery wizard (in this instance realtime) for
outbound registrations and after adding in an appropriate call to
ast_sorcery_apply_config() (since it is missing) Asterisk will crash after
a stack overflow occurs due to the code infinitely recursing. The fix entails
removing the outbound registration state dependency from the outbound
registration sorcery object and instead keeping an in memory container that
can be used to lookup the state when needed.
ASTERISK-24514
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4164/
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This change corrects message ordering in cases where a channel-related
message can be received after a Stasis/ARI application has received the
StasisEnd message. The StasisEnd message was being passed to
applications directly without waiting for the channel topic to empty.
As a result of this fix, other bugs were also identified and fixed:
* StasisStart messages were also being sent directly to apps and are
now routed through the stasis message bus properly
* Masquerade monitor datastores were being removed at the incorrect
time in some cases and were causing StasisEnd messages to not be sent
* General refactoring where necessary for the above
* Unsubscription on StasisEnd timing changes to prevent additional
messages from following the StasisEnd when they shouldn't
A channel sanitization function pointer was added to reduce processing
and AO2 lookups.
Review: https://reviewboard.asterisk.org/r/4163/
ASTERISK-24501 #close
Reported by: Matt Jordan
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When handling incoming messages we determine if it is associated with
a dialog. If so we use that to determine what serializer and endpoint
to use for the message. Previously this would pass the endpoint to the
endpoint lookup module to actually place the endpoint completely on the
message. For in-dialog responses, however, this did not occur as
dialog processing took over and the endpoint lookup did not occur.
This change just places the endpoint in the expected spot immediately
instead of relying on the endpoint lookup module. In-dialog responses
thus have the expected endpoint.
AST-1459 #close
Review: https://reviewboard.asterisk.org/r/4146/
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Testing has shown repeatedly that PJSIP's default behavior of switching
automatically to TCP for large messages can cause issues. The most common
issues are that devices that we are communicating with do not handle the
switch to TCP gracefully, thus causing situations such as broken calls or
broken subscriptions. Now, in order to have this behavior happen, you must
opt into it. The sample file has been updated to warn that enabling the
TCP switch behavior may cause issues for you, so use at your own risk.
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Virtual line support establishes a relationship between messages
related to an outbound registration and a local endpoint. This is
accomplished by attaching a parameter to the Contact of the outbound
registration and looking for it on any received requests. If the
parameter exists and can be matched to an outbound registration
the configured endpoint is associated with the request.
Review: https://reviewboard.asterisk.org/r/2964/
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When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In
some circumstances (on some networks), this can cause some issues with
messages not getting sent to the correct destination - and can also cause
connections to get dropped due to quirks in pjproject deciding to
terminate TCP connections with no messages.
While fixing the routing/messaging issues is important, having a
configuration option in Asterisk that tells pjproject to not switch over
to TCP would be useful. That way, if some glitch is discovered on some
other network/site, we can at least disable the behavior until a fix is
put into place.
AFS-197 #close
Review: https://reviewboard.asterisk.org/r/4137/
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This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Review: https://reviewboard.asterisk.org/r/4103/
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When the res_stasis module is unloaded, it will dispose of the apps_registry
container. This is a problem if an ARI operation is in flight that attempts
to use the registry, as the shutdown occurs in a separate thread. This patch
adds some sanity checks to the various routines that access the registry which
cause the operations to fail if the apps_registry does not exist.
Crash caught by the Asterisk Test Suite.
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Currently, it is possible for some subscriptions to get into a NULL state. When
this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a
device is subscribed for extension state then the associated subscription state
object can't be located. The code then attempts to dereference a NULL object.
Added a NULL check to avoid the problem.
Reported by: John Bigelow
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In res_phoneprov, unloading the module first destroys the http_routes
container, followed by the users. However, users may have a route in
the http_routes container; the validity of this container is not checked
in the users destructor. Hence, we hit an assert as the container has already
been set to NULL.
This patch does two things:
(1) It adds a sanity check in the user destructor (because why not)
(2) It switches the order of destruction, so that users are disposed of prior
to the HTTP routes they may hold a reference to.
Note that this crash was caught by the Test Suite (go go testing!)
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There are two aspects to the vulnerability:
(1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use
TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the
TCP/TLS core, which should be done as an improvement at a latter date.
(2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified,
will default to the OpenSSL SSLv23_method. This method allows for all
ecnryption methods, including SSLv2/SSLv3. A MITM can exploit this by
forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE.
This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration,
and explicitly disables SSLv2/SSLv3 if using SSLv23_method.
For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is
explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or
SSLv3.
Much thanks to abelbeck for reporting the vulnerability and providing a patch
for the res_jabber/res_xmpp modules.
Review: https://reviewboard.asterisk.org/r/4096/
ASTERISK-24425 #close
Reported by: abelbeck
Tested by: abelbeck, opsmonitor, gtjoseph
patches:
asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903)
asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903)
AST-2014-011-1.8.diff uploaded by mjordan (License 6283)
AST-2014-011-11.diff uploaded by mjordan (License 6283)
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This patch for r425924 introduced a bug, wherein sending an INVITE request
with no SDP would cause Asterisk to not send an SDP Offer in the 200
OK. The current structure of res_pjsip_sdp_rtp is a bit hard to deal with
to fix this, as create_outgoing_sdp has no knowledge of whether or not it is
creating an SDP as a new Offer or an Answer. This is something of an oversight
in the callback definition, as the caller of it does have this information.
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The usage of the local override_prefs variable in create_outgoing_sdp_stream
was previously to track an override format preference set by PJSIP_MEDIA_OFFER.
Now, however, that function simply sets the joint capabilities structure,
session->req_caps. During the media format rework, the override_prefs was
instead used to check if there were any formats in session->req_caps.
However, this usage isn't useful in create_outgoing_sdp_stream.
session->req_caps contains the negotiated formats for *all* streams, not just
the current one being created. Thus, so long as any stream of any type has
provided a format, override_prefs will be non-zero. Hence, its usage in
checking whether or not we should look at the formats on the endpoint or
the joint capabilities is generally useless.
There's only two things useful to check:
(1) Does the endpoint have a format for the media type?
(2) Did we negotiate a format for the media type?
If either of those is a 'no', then we must kill the media stream.
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When an inbound SDP offer is received, Asterisk currently makes a few
incorrection assumptions:
(1) If the offer contains more than a single audio/video stream, Asterisk will
reject the entire stream with a 488. This is an overly strict response;
generally, Asterisk should accept the media streams that it can accept and
decline the others.
(2) If the offer contains a declined media stream, Asterisk will attempt to
process it anyway. This can result in attempting to match format
capabilities on a declined media stream, leading to a 488. Asterisk should
simply ignore declined media streams.
(3) Asterisk will currently attempt to handle offers with AVPF with
use_avpf=No/AVP with use_avpf=Yes. This mismatch results in invalid SDP
answers being sent in response. If there is a mismatch between the media
type being offered and the configuration, Asterisk must reject the offer
with a 488.
This patch does the following:
* Asterisk will accept SDP offers with at least one media stream that it can
use. Some WARNING messages have been dropped to NOTICEs as a result.
* Asterisk will not accept an offer with a media type that doesn't match its
configuration.
* Asterisk will ignore declined media streams properly.
#SIPit31
Review: https://reviewboard.asterisk.org/r/4063/
ASTERISK-24122 #close
Reported by: James Van Vleet
ASTERISK-24381 #close
Reported by: Matt Jordan
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This change adds a module which is configurable using the keep_alive_interval setting in the
global section that will send a CRLF keep alive to all active connection-oriented transports at
the provided interval. This is useful because it can help keep connections open through NATs.
This functionality also exists within PJSIP but can not be controlled at runtime and requires
recompiling it.
Review: https://reviewboard.asterisk.org/r/4084/
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