| Age | Commit message (Collapse) | Author |
|---|
|
|
|
into 13
|
|
The res_pjsip_caller_id module wrongly assumed that a
saved From header would always exist on sessions. This
is true until an inbound call is received and a session
timer causes an UPDATE to be sent. In this case there will
be no saved From header and a crash will occur. This change
makes it fall back to the From header of the outgoing request
if no saved From header is present.
ASTERISK-26307 #close
Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa
|
|
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
|
|
When an explicit media_address is provided the address family
in the SDP needs to be set to reflect it.
ASTERISK-26309
Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79
|
|
|
|
|
|
This change adds support for dual stack automatically. No
configuration is required and the IP address and version
in the SIP messages and SDP will be automatically changed
based on the transport over which the message is being
sent. RTP usage has also been changed to listen on both
IPv4 and IPv6 simultaneously to allow media to flow, and
to allow ICE support on both simultaneously. This also
allows failover between IPv6 and IPv4 to work as expected.
ASTERISK-26309 #close
Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d
|
|
ARI and AMI allow for an explicit channel ID to be specified
when originating channels. Unfortunately, there is nothing in
place to prevent someone from using the same ID for multiple
channels. Further complicating things, adding ID validation to channel
allocation makes it impossible for ARI to discern why channel allocation
failed, resulting in a vague error code being returned.
The fix for this is to institute a new method for channel errors to be
discerned. The method mirrors errno, in that when an error occurs, the
caller can consult the channel errno value to determine what the error
was. This initial iteration of the feature only introduces "unknown" and
"channel ID exists" errors. However, it's possible to add more errors as
needed.
ARI uses this feature to determine why channel allocation failed and can
return a 409 error during origination to show that a channel with the
given ID already exists.
ASTERISK-26421
Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06
|
|
Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the
form ice_blacklist = <subnet spec>, e.g. ice_blacklist =
192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay
discovery. This is useful for optimizing the ICE process where a system
has multiple host address ranges and/or physical interfaces and certain
of them are not expected to be used for RTP. Multiple ice_blacklist
configuration lines may be used. If left unconfigured, all discovered
host addresses are used, as per previous behavior.
Documention in rtp.conf.sample.
ASTERISK-26418 #close
Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9
|
|
This patch adds the Asterisk EID field to all outgoing ARI events.
Because this field should be added to all events as they are
transmitted, it is appended to the JSON message just prior to it being
handed off to the application message handler. This makes it somewhat
resilient to both new events being added to ARI, as well as other
potential event transport mechanisms.
ASTERISK-26470 #close
Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d
|
|
Added needed UTF-8 checks before constructing json objects in various
files for strings obtained outside the system. In this case string values
from a channel driver's peer and not from the user setting channel
variables.
* aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
object construction.
ASTERISK-26466
Reported by: Richard Mudgett
Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096
|
|
When T.38 gets rejected and G711 failback occurs there is a period of
time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set,
leading to a crash.
Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982
|
|
If a bridge switched to P2P when a DTMF was in progress it
was possible for the DTMF to continue being sent indefinitely.
Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29
|
|
Fixed a memory leak. It removes only the first element.
Added a useful feature in vector.h to remove all items
under the CMP through a callback function / macro.
ASTERISK-26453 #close
Change-Id: I84508353463456d2495678f125738e20052da950
|
|
codec_opus" into 13
|
|
Preparation
ASTERISK-26409
Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3
(cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9)
|
|
Some external packages have multiple variants that apply to different
builds of asterisk. The DPMA for instance has a "bundled" variant that
needs to be downloaded if asterisk was configured with
--with-pjproject-bundled.
There are 2 ways to specify variants:
If you need the user to make the decision about which variant to
download, simply create multiple menuselect "member" entries like so...
<member name="res_digium_phone" displayname="..snipped..">
<support_level>external</support_level>
<depend>xmlstarlet</depend>
<depend>bash</depend>
<defaultenabled>no</defaultenabled>
</member>
<member name="res_digium_phone-bundled" displayname="..snipped..">
<support_level>external</support_level>
<depend>xmlstarlet</depend>
<depend>bash</depend>
<defaultenabled>no</defaultenabled>
</member>
Note that the second entry has "-<variant>" appended to the name.
You can then use the existing menuselect facilities to restrict which
members to enable or disable. Youy probably don't want the user to
enable multiple at the same time.
If you want to hide the details of the variants, the better way to
do it is to create 1 member with "variant" elements.
<member name="res_digium_phone" displayname="..snipped..">
<support_level>external</support_level>
<depend>xmlstarlet</depend>
<depend>bash</depend>
<defaultenabled>no</defaultenabled>
<member_data>
<downloader>
<variants>
<variant tag="bundled"
condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/>
</variants>
</downloader>
</member_data>
</member>
The condition must be a bash expression suitable for use with an "if"
statement. Any environment variable can be used plus those available
in makeopts.
In this case, if asterisk was configured with --with-pjproject-bundled
the bundled variant will be automatically downloaded. Otherwise the
normal version will be downloaded.
Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e
|
|
This changes the notice for the deprecation of the old
pooling options to point to the new option for doing
pooling. This gives a clearer direction as to what to
look into.
ASTERISK-26389 #close
Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10
|
|
|
|
Creating ODBC SQL queries resulted in queries too large to fit into the
supplied buffer. The resulting truncated buffer contained an invalid SQL
query.
* Made SQL query generation code use a thread storage buffer that can
increase in size as needed.
* Fixed bad multi-line warning messages.
ASTERISK-26263 #close
Reported by: Jeppe Ryskov Larsen
Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae
|
|
The res_pjsip_multihomed module determines what interface and transport
a request is going out on and updates the SIP message accordingly with
the address information. This currently incorrectly updates the Contact
header for connectionful protocols to the ephemeral connection port,
instead of the bound address for the listening socket which can actually
accept the connection back. If the remote side attempts to connect back on
the epehemeral port it will fail.
This change makes it so the port is updated to the bound port on
connectionful protocols and is maintained on UDP (as there can be
multiple of those).
ASTERISK-26374 #close
Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab
|
|
seconds." into 13
|
|
Currently when receiving video over RTP we store only
a calculated samples on the frame. When starting the video
it can take some time for this calculation to actually yield
a value as it requires constant changing timestamps. As well
if a video frame passes over multiple RTP packets this calculation
will fail as the timestamp is the same as the previous RTP
packet and the number of samples calculated will be 0.
This change preserves the timestamp on the frame and allows
it to pass through the core. When sending the video this timestamp
is used instead of a new one being calculated.
ASTERISK-26367 #close
Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd
|
|
|
|
ASTERISK-26375 #close
Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc
|
|
|
|
This implements the chan_sip legacy_useroption_parsing option but with a
better name.
* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.
ASTERISK-26316 #close
Reported by: Kevin Harwell
Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
|
|
The code was incorrectly invoking the unidentified logic when
an endpoint had actually been identified, causing log messages
to be output.
ASTERISK-26349 #close
Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f
|
|
The endpoint identification PJSIP module is intended to identify which
endpoint an incoming request is from. If an endpoint is not identified,
then an artificial endpoint is used in its place when proceeding.
The problem is that the ACK request type is an exception to the rule.
The artificial endpoint is not used when processing an ACK. This results
in the possibility of having a NULL endpoint being used further on.
The reason ACK is an exception is an attempt not to spam security logs
with unidentified requests. Presumably, you've already logged the
unidentified request on the preceeding INVITE.
Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion
didn't cause an issue. A new change in 13.10 added endpoint ACL checking
shortly after endpoint identification. Because we are accessing a NULL
endpoint, this ACL check resulted in a crash.
The fix here is to be sure to retrieve the artificial endpoint for all
request types. ACKs still do not generate unidentified request security
events.
ASTERISK-26264 #close
Reported by nappsoft
AST-2016-006
Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703
|
|
* Eliminated RAII_VAR in get_outbound_endpoint().
* Simplify update_to() coding. However, this function can only be a NoOp
because the To string can only be a URI and not a name-address formatted
string.
* Simplify update_from() coding. Also fixed a code path modifying the
from string when the caller could still want to use the original string.
* Fixed msg_data_create() incompletely removing the "pjsip:" to then add
back the "sip:" string if needed. The code didn't handle the "pjsip:sip:"
case because it left the colon after pjsip in the string.
Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db
|
|
Currently when you add global headers from the dialplan both
the header in the dialplan and the globally configured header
are added to the resulting SIP INVITE. This change makes it
so the headers in the dialplan take precedence and are the
only ones added.
Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad
|
|
|
|
|
|
|
|
|
|
The DPMA and g729a, silk, siren7 and siren14 codecs hosted at
http://downloads.digium.com/pub/telephony/ are now listed in the
"External" sections of the "Resource Modules" and "Codec Translators"
pages in menuselect. Any that are selected will automatically be
downloaded and installed when "make install" is run. Their LICENSE and
README (if avaialble) files will be installed to
ASTVARLIBDIR/documentation/thirdparty/<product_name>.
Example use with codecs:
The codecs/codecs.xml file is a menuselect style xml file that lists
the codecs to be included. Their support levels are 'external', which
triggers the download and install, and defaultenabled is no. Also
because codec_g729a is actually in a directory named codec_g729 on the
download server, the newly added 'member_data' element is used to
override the default of the directory name being the package name. You
can use the 'directory_name' attribute to keep default base URL
(http://downloads.digium.com/pub/telephony/) but use the new directory,
or you use the 'remote_url' attribute to specify a full URL to the
download directory. In this case, you must still follow the same
subdirectory naming conventions as that used for the packages located
at 'http://downloads.digium.com/pub/telephony'.
A new configure option '--with-externals-cache' was added and like
'--with-sounds-cache' it allows the installer to cache tarballs so
they're not downloaded every time.
To assist with the download and install process, each external package
now has a manifest.xml file that, among other things, contains a package
version and checksums for each file in the tarball. The manifest is
saved to both the cache directory and ASTMODDIR and together with the
manifest.xml on the downloads site, tells the install scripts whether
a download and/or update is needed.
bash and xmlstarlet are required for downloader operation. If they're
not installed, the external items in menuselect will be unavailable.
Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a
|
|
Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09
|
|
Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838
|
|
Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3
|
|
On heavy loaded system the TCP/TLS incoming calls could be
disconnected by pjproject while these calls are being
processed by asterisk which could use the session's memory pools.
If the session in the disconnected state then the session memory
pools were already freed, so we get segfault.
This patch adds a lifetime control on an INVITE session to pjproject.
The lifetime of the session is manipulated by calling
pjsip_inv_add_ref/pjsip_inv_dec_ref.
This patch uses these functions to inform pjproject that the
session is in use.
This patch adds check if the session state is not disconnected
and also checks if the memory pool is not NULL.
This patch also places tasks 'session_end' and 'session_end_completion'
into session's serializer to avoid race condition.
ASTERISK-26291 #close
Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7
|
|
|
|
If the PJSIP endpoint's AOR with the permanent contact
was deleted from the realtime storage the res_pjsip module
continues trying to qualify this contact.
The error 'Unable to find an endpoint to qualify contact'
appeares every 'qualify_frequency' seconds.
This patch deletes this contact in this case.
The PJSIP endpoint's AOR with the permanent contact
is never qualified if it is added to realtime storage
after asterisk started.
This patch adds qualifying for the AOR's permanent contacts
on the first handling of this AOR.
ASTERISK-26319 #close
Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe
|
|
|
|
|
|
We may check a global config option hundreds of times a second or more.
Asking sorcery for the global configuration from the config files backend
involves several allocations and container traversals. Using realtime
without a memory cache is a lot worse because you have to lookup in the
realtime database each time to reconstitute the sorcery object. With a
memory cache for realtime, there is about the same amount of overhead as
for config files. Either way, it is still fairly expensive to access the
sorcery object that much.
* Cache the global config options so we can access them faster. You must
now always perform a res_pjsip reload to change the global options.
Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7
|
|
ast_channel_get_t38_state() calls ast_channel_queryoption() with
AST_OPTION_T38_STATE. If the passed in channel is a local channel then a
deadlock can happen if a channel lock is held when called.
* Made ast_channel_get_t38_state() callers not hold a channel lock before
calling.
* Update ast_channel_get_t38_state() doxygen to note that no channel locks
can be held when calling the function.
ASTERISK-26203 #close
Reported by: Etienne Lessard
ASTERISK-24822 #close
Reported by: David Brillert
ASTERISK-22732 #close
Reported by: Richard Mudgett
Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214
|
|
ASTERISK-25980 added the FAXMODE channel variable to res_fax.c.
Unfortunately, it also introduced a deadlock potential because
set_channel_variables() which sets FAXMODE can be called during a
masquerade. The ast_channel_get_t38_state() which gets the value used to
set FAXMODE cannot be called with the channel locked. As a result, local
channels can deadlock because of how they must acquire the locks necessary
to operate.
The intent of FAXMODE is for dialplan to know how a fax was transferred
after the fax completes. However, the previous patch sets FAXMODE to the
channel's current T.38 state AFTER the fax has completed and where T.38
may have already disconnected.
* Set FAXMODE based upon T.38 negotiations exchanged either with the fax
applications or the fax framehooks.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1
|
|
fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be
called with any channel locks already held. A deadlock can happen if the
function is operating on a local channel.
* Made fax_gateway_indicate_t38() unlock the channel before calling
ast_indicate_data() since fax_gateway_indicate_t38() is always called with
the channel locked.
* Made fax_gateway_indicate_t38() return void since nothing cared about
its return value.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407
|
|
Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7
|
