We must continue using the serializer that the original INVITE came in on
for the dialog. There may be retransmissions already enqueued in the
original serializer that can result in reentrancy and message sequencing
problems.
Outgoing call legs create the pjsip/outsess/<endpoint> serializers for
their dialogs.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc
* Resolves potential reentrancy problems if system restarted in the middle
of subscription message transactions.
* Fixes memory leak recreating persistent subscriptions when the
subscription resource tree could not be created.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be
We must continue using the serializer that the original SUBSCRIBE came in
on for the dialog. There may be retransmissions already enqueued in the
original serializer that can result in reentrancy and message sequencing
problems. The "sip_transaction Unable to register SUBSCRIBE transaction
(key exists)" message is a notable symptom of this issue.
Outgoing subscriptions still create the pjsip/pubsub/<endpoint>
serializers for their dialogs.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0
Incoming messages that are not part of a dialog or a recognized response
to one of our requests need to be sent to a consistent serializer. Under
load we may be queueing retransmissions before we can process the original
message. We don't need to throw these messages onto random serializers
and cause reentrancy and message sequencing problems.
* Created a pool of pjsip/distributor serializers that get picked by
hashing the call-id and remote tag strings of the received messages.
* Made ast_sip_destroy_distributor() destroy items in the reverse order of
creation.
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I2ce769389fc060d9f379977f559026fbcb632407
We should not be processing any incoming messages until we are fully
booted. We may not have dialplan or other needed configuration loaded
yet.
ASTERISK-26089 #close
Reported by: Scott Griepentrog
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264
Testing has shown that our usage of UnixODBC is problematic
due to bugs within UnixODBC itself as well as the heavy weight
cost of connecting and disconnecting database connections, even
when pooling is enabled.
For users of UnixODBC 2.3.1 and earlier crashes would occur due
to insufficient protection of the disconnect operation. This was
fixed in UnixODBC 2.3.2 and above.
For users of UnixODBC 2.3.3 and higher a slow-down would occur
under heavy database use due to repeated connection establishment.
A regression is present where on each connection the database
configuration is cached again, with the cache growing out of
control.
The connection pool implementation present in this change helps
to mitigate these issues by reducing how much we connect and
disconnect database connections. We also solve the issue of
crashes under UnixODBC 2.3.1 by defaulting the maximum number of
connections to 1, returning us to the previous working behavior.
For users who may have a fixed version the maximum concurrent
connection limit can be increased helping with performance.
The connection pool works by keeping a list of active connections.
If the connection limit has not been reached a new connection is
established. If the connection limit has been reached then the
request waits until a connection becomes available before
continuing.
ASTERISK-26074 #close
ASTERISK-26054 #close
Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff
Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore.
Therefore, the symbol RAND_bytes is used instead of crypto_get_random.
ASTERISK-24436 #close
Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96
The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting
to use UniqueConstraint and failing. It was not imported and after
importing it also continued to fail.
I've changed the script to use the explicit name of the constraint
instead.
Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9
The stringfields refactor to allow adding stringfields to the end of a
structure (f6f4cf459f) exposed some
incomplete cleanup code by some stringfield users.
The most noticeable leaker is the logging system where there is a leak for
every log message generated.
ASTERISK-26078 #close
Reported by: Etienne Lessard
Patches:
jira_asterisk_26078_v13.patch (license #5621) patch uploaded
by Richard Mudgett
Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782
Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.
Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
ASTERISK-26059 #close
Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
The pjproject doxygen for rdata->msg_info.info says to call
pjsip_rx_data_get_info() instead of accessing the struct member directly.
You need to call the function mostly because the function will generate
the struct member value if it is not already setup.
Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799
Re-ordered the body items so Message-Account is second.
Messages-Waiting: no
Message-Account: sip:1571@<IP Removed>:5060
Voice-Message: 0/0 (0/0)
ASTERISK-26065 #close
Reported-by: Ross Beer
Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3
Although all the patches we had against 2.4.5 were applied by Teluu,
a new bug was introduced preventing re-use of tcp and tls transports
This patch removes all the previous patches against 2.4.5, updates
the version to 2.5, and adds a new patch to correct the transport
re-use problem.
Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068
Added notes about when you can read or write headers. Specifically
about being able to read on the inbound channel and write on an
outbound channel.
ASTERISK-26063 #close
Reported by: Private Name
Tested by: Rusty Newton
Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5
The pjproject doxygen for rdata->msg_info.info says to call
pjsip_rx_data_get_info() instead of accessing the struct member directly.
You need to call the function mostly because the function will generate
the struct member value if it is not already setup.
Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2
Fixed some bugs:
- create dirpath when save downloading message from IMAP storage.
- create IMAP folder if not exists when saving to IMAP storage
- check if file successfully opened before write to it
- some IMAP checks
- remove non-standard flag 'Unseen'
etc
Change to debug IMAP mm_status log instead of verbose.
Remove unused X-Asterisk-VM-Caller-channel message header
for security reason. The clients should not know name of peer/endpoint.
ASTERISK-26045 #close
Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b
As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered
endpoint.
Added "via_addr", "via_port", "call_id" to contact.
Added new fields ViaAddress, CallID to AMI event ContactStatus.
ASTERISK-26011
Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576
There are a lot of verbose messages about Endpoint and Contact status
changes if there are many dynamic endpoints.
The patch sets verbose level 2 for Endpoint status changes
and verbose level 3 for Contact status changes.
ASTERISK-26055 #close
Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7
worker_start checked for ZOMBIE status without holding a lock. All
other read/write of worker status are performed with a lock, so this
check should do the same.
ASTERISK-25777 #close
Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781
There are more specific checks for the platform.
Specifically this allows installing OS/X init scripts.
ASTERISK-26038 #close
Change-Id: If08933621145b10362a0cfe73c079301d9c13f50
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of
a signed comparison.
ASTERISK-25669 #close
Reported by: Jesper
patches:
strings.curl.trim.patch submitted by Jesper (License 5518)
Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a
func_odbc was changed in Asterisk 13.9.0
to make func_odbc use a single database connection per DSN
because of reported bug ASTERISK-25938
with MySQL/MariaDB LAST_INSERT_ID().
This is drawback in performance when func_odbc is used
very often in dialplan.
Single database connection should be optional.
ASTERISK-26010
Change-Id: I57d990616c957dabf7597dea5d5c3148f459dfb6
When receiving an incoming response to a dialog-starting INVITE, we were
not matching the response to the INVITE dialog. Since we had not
recorded the to-tag to the dialog structure, the PJSIP-provided method
to find the dialog did not match.
Most of the time, this was not a problem, because there is a fall-back
that makes the response get routed to the same serializer that the
request was sent on. However, in cases where an asynchronous DNS lookup
occurs in the PJSIP core, the thread that sends the INVITE is not
actually a threadpool serializer thread. This means we are unable to
record a serializer to handle the incoming response.
Now, imagine what happens when an INVITE is sent on a non-serialized
thread, and an error response (such as a 486) arrives. The 486 ends up
getting put on some random threadpool thread. Eventually, a hangup task
gets queued on the INVITE dialog serializer. Since the 486 is being
handled on a different thread, the hangup task can execute at the same
time that the 486 is being handled. The hangup task assumes that it is
the sole owner of the INVITE session and channel, so it ends up
potentially freeing the channel and NULLing the session's channel
pointer. The thread handling the 486 can crash as a result.
This change has the incoming response match the INVITE transaction, and
then get the dialog from that transaction. It's the same method we had
been using for matching incoming CANCEL requests. By doing this, we get
the INVITE dialog and can ensure that the 486 response ends up being
handled by the same thread as the hangup, ensuring that the hangup runs
after the 486 has been completely handled.
ASTERISK-25941 #close
Reported by Javier Riveros
Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0
This change introduces the same filtering that is done in res_sorcery_realtime
to the res_sorcery_astdb module. This allows persisted sorcery objects
that may contain unknown fields to still be read in from the AstDB
and used. This is particularly useful when switching between different
versions of Asterisk that may have introduced additional fields.
ASTERISK-26014 #close
Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2
Some SBCs require responses to empty SIP INFO packets
after establishing call via INVITE, if not responded to
they may drop your call after unspecified timeout of X minutes.
They are identified by having no Content-Type, check for this
and respond with 200 - OK message.
ASTERISK-24986 #close
Reported-by: Ilya Trikoz, Federico Santulli
Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0
