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The send request callback function currently assumes that it
will only ever be called on transaction state changes. This is
not always true. If our own timer callback occurs we will call
the callback with a timer event instead of a transaction state
change event. In this case the transaction on the event is
invalid and accessing it will result in a crash.
ASTERISK-26049 #close
Change-Id: I623211c8533eb73056b0250b4580b49ad4174dfc
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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
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* Fixed NULL crash potential if parameters are missing.
* Reordered some operations so further diagnostic messages can be
more helpful.
Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70
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ASTERISK-26029
Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260
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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: I7091783a7150252de8eeb455115bd00514dfe843
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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
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This patch adds a new feature to ARI that allows a client to download
the media associated with a stored recording. The new route is
/recordings/stored/{name}/file, and transmits the underlying binary file
using Asterisk's HTTP server's underlying file transfer facilities.
Because this REST route returns non-JSON, a few small enhancements had
to be made to the Python Swagger generation code, as well as the
mustache templates that generate the ARI bindings.
ASTERISK-26042 #close
Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181
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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
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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
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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>
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This change uses the newly added multi-user support for
outbound publish to publish to the specific user that an
extension state change is for.
This also extends the res_pjsip_outbound_publish support
to include the user specific From and To URI information in
the outbound publishing of extension state. Since the URI
is used when constructing the body it is important to ensure
that the correct local and remote URIs are used.
Finally the max string growths for the dialog-info+xml
body generator has been increased as through testing it has
proven to be too conservative.
ASTERISK-25965
Change-Id: I668fdf697b1e171d4c7e6f282b2e1590f8356ca1
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Added a new multi_user option that when specified allows a particular
configuration to be used for multiple users. It does this by replacing
the user portion of the server uri with a dynamically created one.
Two new API calls have been added in order to make use of the new
functionality:
ast_sip_publish_user_send - Sends an outgoing publish message based on the
given user. If state for the user already exists it uses that, otherwise
it dynamically creates new outbound publishing state for the user at that
time.
ast_sip_publish_user_remove - Removes all outbound publish state objects
associated with the user. This essentially stops outbound publishing for
the user.
ASTERISK-25965 #close
Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc
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Scenario:
Local fax -> Asterisk w/ firewall -> Provider -> Remote fax
* Local fax starts rtp call to remote fax
* Remote fax starts t38 call back to local fax.
* Local fax sends t38 no-signal to Asterisk before sending an OK.
* udptl processes the frame and increments the expected sequence number.
* chan_sip drops the frame because the call isn't up so nothing goes out
the external interface to open the port for incoming packets.
* Local fax sends OK and Asterisk sends OK to the remote fax.
* Remote fax sends t38 packets which are dropped by the firewall.
* Local fax re-sends t38 no-signal with the same sequence number.
* udptl drops the frame because it thinks it's a dup.
* Still no outgoing packets to open the firewall.
* t38 negotiation fails.
The patch drops frames t38 received before udptl sequence processing
when the call hasn't been answered yet. The second no-signal frame
is then seen as new and is relayed out the external interface which
opens the port and allows negotiation to continue.
ASTERISK-26034 #close
Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9
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* Provide consistent indenting of lines in bulleted paragraphs
* Respect the 80 character column width
* Group all like items together, e.g., all dialplan applications under
"Applications", etc.
* Use a single blank line to break up functionality changes within a
larger section
* Use two blanks lines to delineate larger sections
Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647
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Many ARI applications will want to play multiple media files in a row to
a resource. The most common use case is when building long-ish IVR prompts
made up of multiple, smaller sound files. Today, that requires building a
small state machine, listening for each PlaybackFinished event, and triggering
the next sound file to play. While not especially challenging, it is tedious
work. Since requiring developers to write tedious code to do normal activities
stinks, this patch adds the ability to play back a list of media files to a
resource.
Each of the 'play' operations on supported resources (channels and bridges)
now accepts a comma delineated list of media URIs to play. A single Playback
resource is created as a handle to the entire list. The operation of playing
a list is identical to playing a single media URI, save that a new event,
PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final
media URI. When the entire list is finished being played, a PlaybackFinished
event is raised.
In order to help inform applications where they are in the list playback, the
Playback resource now includes a new, optional attribute, 'next_media_uri',
that contains the next URI in the list to be played.
It's important to note the following:
- If an offset is provided to the 'play' operations, it only applies to the
first media URI, as it would be weird to skip n seconds forward in every
media resource.
- Operations that control the position of the media only affect the current
media being played. For example, once a media resource in the list
completes, a 'reverse' operation on a subsequent media resource will not
start a previously completed media resource at the appropiate offset.
- This patch does not add any new operations to control the list. Hopefully,
user feedback and/or future patches would add that if people want it.
ASTERISK-26022 #close
Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f
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When chan_sip does a re-INVITE to refresh a session and authentication
is required, the INVITE with the Authorization header containes a
second Session-Expires header without the ";refersher=" parameter.
This is causing some proxies to return a 400. Also, when Asterisk is
the uas and the refresher, it is including the Session-Expires and
Min-SE headers in OPTIONS messages which is not allowed per RFC4028.
This patch (based on the reporter's) Checks to see if a Session-Expires
header is already in the message before adding another one. It also
checks that the method is INVITE or UPDATE.
ASTERISK-26030 #close
Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9
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Nothing was cleaning up the registration state object when ast_sorcery_delete
was called on a registration. So, the registration was deleted from sorcery
but the state object went right on refreshing the registration (or failing
to refresh the registration) with the peer.
* Added a 'deleted' observer on registration that removes the state object.
ASTERISK-25964 #close
Reported-by Matt Jordan
Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23
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Although it's perfectly legal to place multiple SIP messages in the same packet,
it can cause problems because the Linux default is to enable Path MTU Discovery
which sets the Don't Fragment bit on the packets. If adding a second message to
the packet causes the MTU to be exceeded, and the destination isn't equipped to
send a FRAGMENTATION NEEDED response to a large packet, the packet will just be
dropped.
We can't specifically tell the stack to send only 1 message per packet, but we
can turn on TCP_NODELAY when we create the transport. This will at least tell
the stack to send packets as soon as possible.
ASTERISK-26005 #close
Reported-by: Ross Beer
Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd
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When 2d7a4a3357 was merged, it missed the fact that Verbose log messages
are formatted and handled by 'verbosers'. Verbosers are registered
functions that handle verbose messages only; they exist as a separate
class of callbacks. This was done to handle the 'magic' that must be
inserted into Verbose messages sent to remote consoles, so that the
consoles can format the messages correctly, i.e., the leading
tabs/characters.
In reality, verbosers are a weird appendage: they're a separate class of
formatters/message handlers outside of what handles all other log
messages in Asterisk. After some code inspection, it became clear that
simply passing a Verbose message along with its 'sublevel' importance
through the normal logging mechanisms removes the need for verbosers
altogether.
This patch removes the verbosers, and makes the default log formatter
aware that, if the log channel is a console log, it should simply insert
the 'verbose magic' into the log messages itself. This allows the
console handlers to interpret and format the verbose message
themselves.
This simplifies the code quite a lot, and should improve the performance
of printing verbose messages by a reasonable factor:
(1) It removes a number of memory allocations that were done on each
verobse message
(2) It removes the need to strip the verbose magic out of the verbose
log messages before passing them to non-console log channels
(3) It now performs fewer iterations over lists when handling verbose
messages
Since verbose messages are now handled like other log messages (for the
most part), the JSON formatting of the messages works as well.
ASTERISK-25425
Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
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A ':' is not a valid token for starting a comment.
Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad
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When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its
local address the 'any' address, as opposed to the IP address we
actually received the packet on. This can cause some confusion in Homer,
as it will dutifully report what we send it.
This patch uses the PJSIP inspection routines to determine which IP
address we probably received the packet on based on the remote party's
IP address. In the event that this fails, it falls back to the IP
address natively reported by the transport.
Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3
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The Location headers returned by:
* /bridges/{bridgeId}/play
* /bridges/{bridgeId}/record
* /channels/{channelId}/play
* /channels/{channelId}/record
Did not have the '/ari' prefix, and in the case of the 'play' resources, were
using 'playback' instead of 'playbacks.'
Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c
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At one point in time, it seemed like a good idea to use the Asterisk
channel name as the HEP correlation UUID. In particular, it felt like
this would be a useful identifier to tie PJSIP messages and RTCP
messages together, along with whatever other data we may eventually send
to Homer. This also had the benefit of keeping the correlation UUID
channel technology agnostic.
In practice, it isn't as useful as hoped, for two reasons:
1) The first INVITE request received doesn't have a channel. As a
result, there is always an 'odd message out', leading it to be
potentially uncorrelated in Homer.
2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
This causes RTCP information to be uncorrelated to the SIP message
traffic seen by those capture nodes.
In order to support both (in case someone is trying to use res_hep_rtcp
with a non-PJSIP channel), this patch adds a new option, uuid_type, with
two valid values - 'call-id' and 'channel'. The uuid_type option is used
by a module to determine the preferred UUID type. When available, that
source of a correlation UUID is used; when not, the more readily available
source is used.
For res_hep_pjsip:
- uuid_type = call-id: the module uses the SIP Call-ID header value
- uuid_type = channel: the module uses the channel name if available,
falling back to SIP Call-ID if not
For res_hep_rtcp:
- uuid_type = call-id: the module uses the SIP Call-ID header if the
channel type is PJSIP and we have a channel,
falling back to the Stasis event provided
channel name if not
- uuid_type = channel: the module uses the channel name
ASTERISK-25352 #close
Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c
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With the old SIP module we can use IP access controls per peer.
PJSIP module missing this feature.
This patch added next configuration Endpoint options:
"acl" - list of IP ACL section names in acl.conf
"deny" - List of IP addresses to deny access from
"permit" - List of IP addresses to permit access from
"contact_acl" - List of Contact ACL section names in acl.conf
"contact_deny" - List of Contact header addresses to deny
"contact_permit" - List of Contact header addresses to permit
This patch also better logging failed request:
add custom message instead of "No matching endpoint found"
add SIP method to logging
ASTERISK-25900
Change-Id: I456dea3909d929d413864fb347d28578415ebf02
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In 13.9.0, there was an issue where PJSIP contacts added to an AOR would
be deleted at seemingly random times.
One reason this was happening was because of an operation to retrieve
the contacts whose expiration time was less than or equal to the current
time. When retrieving existing contacts, the contact's expiration time
and the current time were converted from a string to a float, and those
two floats were compared.
On some systems, including mine, this conversion was horribly off. For
instance, I could regularly see the string "1463079214" get converted
into 1463079168.000000. When switching from using a float to using a
double, the conversion was as expected.
Why was the conversion to float off? My best guess is that the
conversion to float was attempting to store the entire value in the 23
bit significand of the IEEE-754 floating point number. In particular, if
you take only the 23 most significant bits of 1463079214, you get the
messed up 1463079168 that we were seeing in the conversion. It likely
was possible to get a more precise value by composing the number using
an exponent, but the conversion did not work that way. With a double,
you have a 52 bit significand, allowing the entire value to fit there,
and thereby allowing an accurate conversion.
ASTERISK-26007 #close
Reported by Greg Siemon
Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070
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There was a newline missing from the end of the "no matching endpoint" notice.
Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181
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There are two types of SIP URIs indicating a secure transport:
* sips:user@example.org
* sip:user@example.org;transport=tls
When using a sips URI, Asterisk checks incoming INVITEs and answers from
the other side for sips URIs, and rejects the packet if there are only
sip URIs. So Asterisk should only generate a sips Contact URI if the
other side supports it.
This patch makes Asterisk generate either a sip or sips Contact URI
depending on the format of the server URI.
If you want a sip URI, use:
server_uri=sip:example.org\;transport=tls
If you want a sips URI, use:
server_uri=sips:example.org
ASTERISK-25990 #close
Reported-by: Sebastian Damm
Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2
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During refactoring of this support the addition of
the PID to messages was removed. This change adds it
back in.
ASTERISK-25538 #close
Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36
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When running on a system that does not support or use AST_UNDEFINED_SANITIZER
or AST_LEAK_SANITIZER, the configure script would incorrectly set those
constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would
cause menuselect to error out, complaining that a blank value is not a
valid option. This patch corrects the issue by setting the value to 0 if
the options that those constants enable/disable is not found.
Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba
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reloads/realtime fetches"
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