AGI recently was modified to defer important frames. This was because
when AGI was used in a connected line interception routine, the
resulting connected line frame would end up getting discarded by the
AGI.
However, this caused bad behavior in other cases. Specifically, during a
transfer, if someone attempted to manually set the Caller ID on a
channel in an AGI, the deferred connected line frame would end up
overwriting what had been manually set in the AGI.
Since the initial issue was specific to interception routines, this
change removes the manual frame deferral from AGI and instead uses the
new frame deferral API in interception routines.
ASTERISK-26343 #close
Reported by Morton Tryfoss
Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208
A NULL bridge has special meaning in res_stasis for
unsubscribing. It means that a subscription to ALL
bridges should be removed. This should not be done
as part of the normal subscription management in
the res_stasis channel loop.
ASTERISK-26468
Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0
Adds an identifier (with a getter and setter) to detect channels with
interleaved audio.
This is needed by the binaural bridge_softmix patch (ASTERISK-26292) and
was already discussed here:
http://lists.digium.com/pipermail/asterisk-dev/2016-October/075900.html
The identifier can be set during fmtp parsing (to be seen in the
res_format_attr_opus.c change).
ASTERISK-26292
Change-Id: I359801cc5f98c35671c48dabc81a7f4ee1183d63
Not surprisingly, using Respoke (and possibly other systems) it is
possible to blow past the 16k limit for a WebSocket packet size. This
patch bumps it up to 32k, which, at least for Respoke, is sufficient.
For now.
Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that
matter), this patch adds a LOW_MEMORY directive that sets the buffer to
8k for systems who have asked for their reduced memory availability to
be considered.
Change-Id: Id235902537091b58608196844dc4b045e383cd2e
When a bridge is created via ARI (through res_stasis), no video source
mode is set by default. As a result, any endpoint sending video media
won't ever see any video reflected back to it.
This patch defaults a bridge to a 'follow the talker' video mode.
Further work can be done to add routes that allow for the video mode to
be controlled through the /bridges resource.
Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866
The readdir_r function has been deprecated and should no longer be used. This
patch removes the readdir_r dependency (replaced it with readdir) and also moves
the directory search code to a more centralized spot (file.c)
Also removed a strict dependency on the dirent structure's d_type field as it
is not portable. The code now checks to see if the value is available. If so,
it tries to use it, but defaults back to using the stats function if necessary.
Lastly, for most implementations of readdir it *should* be thread-safe to make
concurrent calls to it as long as different directory streams are specified.
glibc falls into this category. However, since it is possible that there exist
some implementations that are not safe, locking has been added for those other
than glibc.
ASTERISK-26412
ASTERISK-26509 #close
Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba
The res_pjsip_sdp_rtp module did not restrict the number of
formats added to a media stream in the SDP to the defined
limit. If allow=all was used with additional loaded codecs this
could result in the next media stream being overwritten some.
This change restricts the module to limit it to the defined
maximum and also increases the maximum in our bundled pjproject.
ASTERISK-26541 #close
Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8
This patch adds three new CLI commands:
- ari show apps: list the registered ARI applications
- ari show app: show detailed information about an ARI application
- ari set debug: dump events being sent to an ARI application
Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.
ASTERISK-26488 #close
Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5
While publishing device state between multiple instances of Asterisk,
a crash will sporadically occur under high CPS which looks to be a
race condition operating on the publisher queue.
ASTERISK-26506
Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
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
This is similar to what is done for origination, but for the 14 and up
channel creation method. When attempting to create a channel, if a
channel ID is specified and a channel already exists with that ID, then
a 409 is returned.
Change-Id: I77f9253278c6947939c418073b6b31065489187c
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
We use a lot res_calendar, we are very happy with that, especially
because you use libical, the almost alone opensource library that
supports really ical format with all types of recurrency.
Nevertheless, some features are missed for our business use cases.
This first patch adds a new option in calendar.conf:
fetch_again_at_reload. Be my guest for a better name.
If it's true, when you'll launch "module reload res_calendar.so",
Asterisk will download again the calendar.
The business use case is that we have a WebUI with a scheduler planner,
we know when the calendars are modified.
For now, we need to define 1 minute of timeout to have a chance that
our user doesn't wait too long between the modification and the real
test. But it generates a lot of useless HTTP traffic.
ASTERISK-26422 #close
Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077
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
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
When performing DNS resolution the failover code present in
res_pjsip currently assumes that a request will always have
at least one viable address. In practice this is not true.
A domain may be used that has no records.
The code now checks that at least one address exists on the
request which prevents looping.
ASTERISK-26364 #close
Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c
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
