This patch covers two problems:
1) Currently, when a call is transferred into a parking lot from a bridge
(using either the blind transfer or one touch parking mechanisms), the
application fails to be set to "Park" in the resulting CDR record for
the parked channel. This is due to the ParkedCall message arriving before
the BridgeEnter for the channel entering the parking bridge. The ParkedCall
message isn't handled as the CDR for the channel has already been finalized
(due to the channel having left its two party bridge), and the BridgeEnter -
which creates the new CDR - doesn't have the parking information. This patch
modifies the behavior so that reception of a ParkedCall message will - if
not handled by a CDR chain - cause a new CDR to be created and put into the
Parking state.
2) It fixes a FRACK that occurred when a channel is originated into a parking
space. The DialedPending state - which occurs for both Dialed and Originated
channels - assumed that it couldn't handle the parking transitions due to it
having a Party B; however, Originated channels don't have a Party B. As such,
the existing CDR needs to transition into the parking state - this patch does
that.
Review: https://reviewboard.asterisk.org/r/2877/
(closes issue ASTERISK-22482)
Reported by: Richard Mudgett
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app_queue currently attempts to handle Local channel optimizations in an effort
to provide accurate information in Stasis messages (and their corresponding
AMI events) as well as the Queue log. Sometimes, however, things don't go as
planned.
Consider the following scenario:
SIP/foo <-> L;1 <-> L;2 <-> SIP/agent
SIP/agent answers, triggering a Local channel optimization. app_queue will
normally do the following:
* Listen for the Local optimization events and update our agent accordingly
to SIP/agent in the queue log and messages
* When we get a hangup, publish the AgentComplete event based on our
information (SIP/foo and SIP/agent)
However, as with all things that depend on sanity from something as capricious
as Local channels, things can go wrong:
(1) SIP/agent immediately hangs up upon answering. This triggers a race
condition between termination messages coming from SIP/agent and the
ongoing Local channel optimization messages. (Note that this can also
occur with SIP/foo)
(2) In a race condition, Asterisk can (rarely) deliver the hangup messages
prior to the Local channel optimization.
In that case, the messages *may* arrive to app_queue in the following order:
* Hangup SIP/Agent
* Hangup SIP/foo
* Optimize L;1/L;2
* Hangup L;2
* Hangup L;1
When app_queue receives the hangup of the agent or the caller, it will attempt
to publish the AgentComplete event. However, it now has a problem - it thinks
its agent is the ;1 side of the Local channel, as it never received the
optimization event. At the same time, that channel is already gone. This
results in getting NULL from the Stasis cache. What's more, we can't really
wait for the optimization message, as we are currently handling the hangup
of the channel that the optimization event would tell us to use.
This patch modifies the behavior in app_queue such that, since we still have a
lot of pertinent queue information (interface, queue name, etc.), we now raise
the event with what information we know. The channels involved now may or may
not be present. Users will still at least get the "AgentComplete" event, which
"completes" the known Agent information.
Review: https://reviewboard.asterisk.org/r/2878/
(closes issue ASTERISK-22507)
Reported by: Richard Mudgett
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In r399887, a minor performance improvement was introduced by not allocating
the manager variable struct if it wasn't used. Unfortunately, when directly
accessing an ast_channel struct, manager assumed that the struct was always
allocated. Since this was no longer the case, things got a bit crashy.
This fixes that problem by simply bypassing appending variables if the manager
channel variable struct isn't there.
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There was a collision of mod_data use on the transaction between using a nat
hook and an session response callback. During state change it was assumed
what was in the mod_data was nothing or the response callback. However, it
was possible for it to also contain a nat hook thus resulting in a bad cast
and a crash.
Added the ability to store multiple data elements in mod_data via a hash table.
In this instance, mod_data now stores a hash table of the two values that can
be retrieved using an associated string key.
(closes issue ASTERISK-22394)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2843/
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When Asterisk receives a 200 OK in response to an invite, that peer should have
sent an SDP at some point by then. If the channel has never received an SDP,
media won't have been set and the remote address won't be known. Endpoints in
general should not be doing this. This patch makes it so that Asterisk will
simply hang up a call if it sends a 200 OK at this point. So far this odd
behavior for endpoints has only been observed in tests which involved manually
created SIP transactions in SIPp.
(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/
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These refleaks were causing bridged calls not to close their RTP ports. Thus
a call would leave open 4 ports (RTP for party A, RTCP for party A, RTP for party
B, and RTCP for party B). This led to an eventual depletion of available RTP
ports.
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While handling a registration request a race condition could occur if/when two+
clients registered at the same time. This happened when one request obtained a
copy of the current contacts for an AOR and another request did the same before
the first request updated. Thus the second would update and overwrite the first
(or vice-versa depending on which actually updated first). In the case of it
being the same contact two "add" events would be raised.
pjsip registration handling is now serialized to alleviate this issue.
(closes issue AST-1213)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2860/
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The PRI and SS7 link control threads are not stopped correctly when the
chan_dahdi.so module is unloaded. The link control threads pri_dchannel()
and ss7_linkset() are not awakened from a poll() to cancel the thread.
* Added a SIGURG signal after requesting the thread cancel to break the
link control thread poll() immediately.
For SS7 it was slightly worse, the link poll() timeout would always be
whatever was the last libss7 scheduled event time used. If no libss7
scheduled event was pending, the thread could run more often than
necessary.
* Set nextms to 60 seconds for the ss7_linkset() poll() if there is no
other libss7 scheduled event.
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1st Issue
When a realtime peer sends an un-REGISTER request, Asterisk
un-registers the peer but the database table record still has regseconds and
fullcontact for the peer. This results in calls attempting to be routed to the
peer which is no longer registered. The expected behavior is to get
busy/congested when attempting to call an un-registered peer through the
dialplan.
What was discovered is that we are clearing out the peer's registration in the
database in parse_register_contact() when calling expire_register() but then
upon returning from parse_register_contact(), update_peer() is run which stores
back in the database table regseconds and fullcontact.
2nd Issue
The reporter pointed out that the 200 ok being returned by Asterisk
after un-registering a peer contains a Contact header with ;expires= and the
Expires header is not set to 0. This is actually a regression.
Tests were created for this second issue (ASTERISK-22548). The tests have been
reviewed and a Ship It! was received on those tests.
This patch does the following:
* Do not ignore the Expires header value even when it is set to 0. The patch
sets the pvt->expiry earlier on in the function so that it is set properly and
used.
* If pvt->expiry is 0, do not call update_peer since that means the peer has
already been un-registered and there is no need to update the database record
again since nothing has changed.
(closes issue ASTERISK-22428)
Reported by: Ben Smithurst
Tested by: Ben Smithurst, Michael L. Young
Patches:
asterisk-22428-rt-peer-update-and-expires-header.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2869/
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Re-using some of Mark Michelson's text from an E-mail discussion for:
* Modifying synopsis for both options
* Adding description to both options
* Changing name of "external_media_address" for Endpoint configuration to "media_address" in anticipation of the option name being changed. (As it is not really specific to external destinations)
(issue ASTERISK-22405)
(closes issue ASTERISK-22405)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2850/
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If the channel variable MONITOR_EXEC is set, app_queue will pass the specified
execution parameters to the MixMonitor application when a queue is recorded.
If that channel variable is not set, the buffer that holds the escaped value
was not being initialized to NULL, and so would be passed to the MixMonitor
application with garbage. Hilarity ensued as app_mixmonitor attempted to
execute gobeldy-gook.
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There is a large performance price currently in the CDR engine. We currently
perform two ao2_callback calls on a container that has an entry for every
channel in the system. This is done to create matching pairs between channels
in a bridge.
As such, the portion of the CDR logic that this patch deals with is how we
make pairings when a channel enters a mixing bridge. In general, when a
channel enters such a bridge, we need to do two things:
(1) Figure out if anyone in the bridge can be this channel's Party B.
(2) Make pairings with every other channel in the bridge that is not already
our Party B.
This is a two step process. In the first step, we look through everyone in the
bridge and see if they can be our Party B (single_state_process_bridge_enter).
If they can - yay! We mark our CDR as having gotten a Party B. If not, we keep
searching. If we don't find one, we wait until someone joins who can be our
Party B.
Step 2 is where we changed the logic
(handle_bridge_pairings and bridge_candidate_process). Previously, we would
first find candidates - those channels in the bridge with us - from the
active_cdrs_by_channel container. Because a channel could be a candidate if it
was Party B to an item in the container, the code implemented multiple
ao2_container callbacks to get all the candidates. We also had to store them
in another container with some other meta information. This was rather complex
and costly, particularly if you have 300 Local channels (600 channels!) going
at once.
Luckily, none of it is needed: when a channel enters a bridge (which is when
we're figuring all this stuff out), the bridge snapshot tells us the unique
IDs of everyone already in the bridge. All we need to do is:
For all channels in the bridge:
If the channel is us or our Party B that we got in step 1, skip it
Compare us and the candidate to figure out who is Party A (based on some
specific rules)
If we are Party A:
Make a new CDR for us, append it to our chain, and set the candidate as
Party B
If they are Party A:
If they don't have a Party B:
Make a new CDR for them, append us to their chain, and us as Party B
Otherwise:
Copy us over as Party B on their existing CDR.
This patch does that.
Because we now use channel unique IDs to find the candidates during bridging,
active_cdrs_by_channel now looks up things using uniqueid instead of channel
name. This makes the more complex code simpler; it does, however, have the
drawback that dialplan applications and functions will be slightly slower as
they have to iterate through the container looking for the CDR by name.
That's a small price to pay however as the bridging code will be called a lot
more often.
This patch also does two other minor changes:
(1) It reduces the container size of the channels in a bridge snapshot to 1.
In order to be predictable for multi-party bridges, the order of the
channels in the container must be stable; that is, it must always devolve
to a linked list.
(2) CDRs and the multi-party test was updated to show the relationship between
two dialed channels. You still want to know if they talked - previously,
dialed channels were always ignored, which is wrong when they have
managed to get a Party B.
(closes issue ASTERISK-22488)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2861/
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During load time in res_pjsip if an error occurred the operation would attempt to rollback all
operations done during load. This is not permitted by PJSIP as it will assert if the operation has
not been done. This fix changes the code so it will only rollback what has been initialized already.
Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to
limitations within PJSIP itself. The library environment can only be changed to a certain extent
and does not provide the ability, currently, to deinitialize certain required functionality.
(closes issue ASTERISK-22474)
Reported by: Corey Farrell
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Moved rtcp_report RAII_VAR declaration into the loop so it is unref'ed
after every loop. Moved message_blob to loop and switched it to a regular
variable. The regular variable was used since message_blob is used in a
very contained way.
(closes issue ASTERISK-22565)
Reported by: Corey Farrell
Patches:
rtcp_report-leak.patch (license #5909) patch uploaded by Corey Farrell
Tested by: Corey Farrell
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The config framework is supposed to be able to load configs that come from
multiple config files. The principle example is chan_sip's sip.conf and
users.conf. Unfortunately, it only does this correctly on initial load.
This patch causes the module's config to be reloaded entirely if any of
the config files change.
(closes issue ASTERISK-22009)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2859/
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pjsip's message technology was being registered as 'sip', which was causing it
to not load due it conflicting with chan_sip's registered 'sip' technology for
messaging. It now registers as 'pjsip'. However, due to this change the "to"
field for outgoing pjsip messages need to be prefixed with 'pjsip:' instead of
'sip:'. Incoming messages to res_pjsip_messaging will automatically have their
"to" fields altered in order to accommodate the change. Outgoing messages also
handle changing it back to 'sip' before being sent so the pjsip library will
properly handle it.
(closes issue ASTERISK-22445)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2833/
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The endpoint option does not apply to communication with external entities. Rather,
the option is applied to all communications with the endpoint. The external_media_address
transport configuration option may override the endpoint option if it turns out that
we are going to be communicating with an external entity.
Two things of note:
1) I have not updated the XML documentation. This is being taken care of by Rusty as part
of his work on issue ASTERISK-22405
2) This commit is likely to cause testsuite failures since there are tests that use the
external_media_address endpoint option, and they will need to be changed over. Well, I'm
planning to get that updated ASAP after this commit.
(closes issue ASTERISK-22528)
reported by Rusty Newton
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The remote console continued to have issues with its output. In this case CLI
command output would either not show up (if verbose level = 0) or would contain
verbose prefixes (if verbose level > 0) once log messages were sent to the
remote console. The fix now now adds verbose prefix data to all new lines
contained in a verbose log string.
(closes issue ASTERISK-22450)
Reported by: David Brillert
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2825/
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Confbridge would not properly tear down an empty conference bridge when all
users were kicked via end_marked=yes and at least one user was also set to
wait_marked. This occurred because while end_marked users were being kicked
and at least one was also set to wait_marked then the leave wait_marked handler
would be called on that user, but there would be no waiting user (still
considered active). The waiting users would decrement and now be negative. The
conference would remain, but be put into an inactive state. The solution was
to move from the active list to the wait list, those users with wait_marked set
right before kicking. This allows both the active and wait users to decrement
correctly and the confbridge to tear down properly.
A crashed also occurred when trying to list the specific conference from the CLI.
This happened because the conference specified was invalid. Since the
conference properly tears down now there is no way to reference it thus
alleviating the crash as well.
(closes issue ASTERISK-21859)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2848/
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You cannot use a function pointer variable with an external function from
another dynamically loaded module because data variables are always
resolved even with RTLD_LAZY.
* Added wrapper functions for ast_ari_validate_int() and
ast_ari_validate_string() to use instead for the function pointer
variable.
(closes issue ASTERISK-22457)
Reported by: David M. Lee
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Fixes regression introduced by -r374096.
* Made res_speech.export.in export ast_* symbols instead of specific
functions.
* Made app_speech_utils.c declare that it is dependent upon res_speech.
(issue ASTERISK-17136)
Reported by: Richard Kenner
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When a new IAX2 client registers, the astdb database is updated with the
value of minregexpire defined in iax.conf instead of using the expiry time
that is provided by the client. The provided expiry time of the client is
updated after inserting the astdb entry. As a consequence, restarting or
reloading asterisk creates clients whose registration may expire before
they reregister. The clients are therefore unavailable after minregexpire
seconds until they reregister.
* Move updating of the expiry time to before inserting into the astdb.
(closes issue ASTERISK-22504)
Reported by: Stefan Wachtler
Patches:
chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler
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Some channels exist merely as an implementation detail in Asterisk, such as
ConfBridge's announcer/recorder channels. These channels should never be
exposed to the outside world, or to interfaces that report on Asterisk. We
already filter out such channels in snapshot processing; however, we failed to
filter out bridge related messages that involved these channels.
This patch filters out bridge related messages that are for such channels. This
prevents a spurious WARNING message from being displayed when those channels
move in and out of bridges.
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