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This works the same as for AMI manager variables. Set
"channelvars=foo,bar" in your ari.conf general section, and then the
channel variables "foo" and "bar" (along with their values), will
appear in every Stasis websocket channel event.
ASTERISK-26492 #close
patches:
ari_vars.diff submitted by Mark Michelson
Change-Id: I5609ba239259577c0948645df776d7f3bc864229
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ASTERISK-26558
Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e
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Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK
(Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the
dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges.
Consequently, when the dynamic range is exhausted, this change utilizes payload
types in the range between 35 and 63 giving room for another 29 payload types.
ASTERISK-26311 #close
Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964
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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
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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
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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
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- Call "rc_openlog" optional. If you do not call,
you will simply NULL instead of a name.
- On the one PID can be only one syslog channel.
And it can already be run in logger.c
- Calling rc_openlog we assigns a new name for
the channel syslog. This unexpected behavior for logger.c.
Most lesser evil, is to agree on a NULL name syslog
if the channel was not launched in logger.c.
It also solves the problem of memory leaks.
ASTERISK-26455 #close
Change-Id: Ic17c38de67583e971d78fe18807d1a9faf8f0afd
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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
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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
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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
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The "Q" option will set the cause on the unanswered channels when
another channel answers. It overrides the default of
ANSWERED_ELSEWHERE.
NOTE: chan_sip does not support setting the cause on a CANCEL to
anything other than ANSWERED_ELSEWHERE.
ASTERISK-26446 #close
Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47
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This allows asterisk to compiled with LOW_MEMORY to load modules built
without LOW_MEMORY.
ASTERISK-26398 #close
Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d
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sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).
To use this, use a systemd unit with 'Type=notify' for Asterisk.
This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.
Also adds support for libsystemd detection in the configure script.
Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811
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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
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This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn
Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
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Confbridge announcements tend to block a channel while they are being
played. In some circumstances, this is warranted since you want that
particular channel not to hear the announcement (Example: "John Doe has
entered the conference"). For others it makes less sense.
This change first introduces methods for playing sounds asynchronously
into the conference. This is very similar to how synchronous sounds are
played, except the channel initiating the playback does not wait for the
sound to complete before moving on.
Asynchronous announcements are used for two circumstances:
* Sounds played for a user after they have left the bridge
* Sounds that play first to a single user and then the rest of the
conference (if the channel and conference use the same language)
ASTERISK-26289 #close
Reported by Mark Michelson
Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a
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Following the Encrypt-all-the-things paradigm:
The user enters his SIP-URI and password. Thanks to DNS-NAPTR, the phone
determines SIP-over-TLS as preferred transport. In SIP/SDP, the phone starts
the call with a crypto attribute, but not as RTP/sAVP but the RTP/AVP profile
(sRTP is preferred aka optional; not mandatory). If the VoIP server does not
support sRTP and TLS, the phone shows an open padlock icon.
This paradigm is supported by several VoIP/SIP clients on default. Some
implementations even cannot be changed to RTP/sAVP. Therefore here, this
change allows Preferred sRTP for ingress. For egress, please, create a dial
plan which starts with RTP/SAVP, and when rejected tries again with RTP/AVP.
ASTERISK-20234 #close
Reported by: tootai
Tested by: tootai, Alexander Traud
patches:
srtp_patches.diff submitted by Matt Jordan
Change-Id: I42cb779df3a9c7b3dd03a629fb3a296aa4ceb0fd
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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
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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
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Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth
is not supported in IAX2 protocol. Please refer to section 8.6.13 of
RFC 5456.
But plaintext auth is still supported by Asterisk implementation of IAX2.
This support should be dropped.
Patch, based on asterisk-dev discussion, adds deprecation warning on
startup if 'auth' is set to 'plaintext', changes default values of
'auth' from 'md5, plaintext' to 'md5'.
Patch is safe in terms of backwards compatibility, will work even if
remote peers have auth=plaintext and we have defaults.
auth=plaintext setting will remain deprecated in Asterisk 14 and 15,
and IAX2 plaintext support will be removed in Asterisk 16.
ASTERISK-22820 #close
Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf
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contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
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Make it clear that we're talking about device state hints and add
an entry to the sample config.
Change-Id: Iaef58ffb960191a21b713e8e0b51ce1fcd47e433
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This patch adds a new PJSIP specific dialplan function,
PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media
session will be refreshed via either an UPDATE or re-INVITE request.
When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function,
the formats in use on a PJSIP channel can be re-negotiated and changed
dynamically after call setup.
ASTERISK-26277 #close
Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b
(cherry picked from commit eec60dd77394f0519895fc6abce3a6f90f6470f1)
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The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.
This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.
This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.
ASTERISK-26230 #close
Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a
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On heavy loaded system with IMAP or DB storage,
'app_voicemail' taskprocessor queue could reach 500 scheduled tasks.
It could happen when the IMAP or DB server dies or is unreachable.
It could happen on startup when there are many (thousands)
realtime endpoints configured with unsolicited mwi.
If the taskprocessor queue reaches the high water level
then the alert is triggered and pjsip stops processing new requests
until the queue reaches the low water level to clear the alert.
This patch adds 2 new 'general' configuration options
to tune taskprocessor alert levels:
'tps_queue_high' - Taskprocessor high water alert trigger level.
'tps_queue_low' - Taskprocessor low water clear alert level
ASTERISK-26229 #close
Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8
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Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that
scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables
Session-Timers for SIP over TCP (and for SIP over TLS).
However with longer international calls via TCP, the SIP channel might break,
because all hops on the Internet route must stay online (have not a single power
outage, for example). Therefore with Session-Timers enabled (which are enabled
at default), you might see dropped calls. Consequently even with this change,
you might be better-off going for session-timers=refuse in your sip.conf.
ASTERISK-19968 #close
Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957
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Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk
defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests
this.
ASTERISK-26218 #close
ASTERISK-26221 #close
Reported by: Aaron Meriwether
Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa
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Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc
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The new option allows the channel driver's faxdetect option to timeout on
a call after the specified number of seconds into a call. The new feature
is disabled if the timeout is set to zero. The option is disabled by
default.
* Don't clear dsp_features after passing them to the dsp code in
my_pri_ss7_open_media(). We should still remember them especially for the
new faxdetect_timeout option.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Ieffd3fe788788d56282844774365546dce8ac810
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The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call. The new feature is disabled if the timeout is set
to zero. The option is disabled by default.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d
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Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS)
support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added
for DTLS. The source code from main/tcptls.c should have been re-used to ease
security audits. Therefore, this change rolls back the change from July 2015 and
re-uses the code from July 2014. This has the additional benefits to work under
CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well.
ASTERISK-25659 #close
Reported by: StefanEng86, urbaniak, pay123
Tested by: sarumjanuch, traud
patches:
res_rtp_asterisk.patch submitted by sarumjanuch
dtls_centos_step_1.patch submitted by traud
dtls_centos_step_2.patch submitted by traud
Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c
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The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.
1) It restarted any OPTIONS RTT ping cycle.
2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.
3) It cleared the RTT time each time the endpoint was refreshed.
4) The cleared RTT time was sent out as a statsd update each time.
5) It created two AMI events for each update.
* Revert the original patch and reimplement it. Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration. The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.
ASTERISK-26160 #close
Reported by: Matt Jordan
Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
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If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.
ASTERISK-25471 #close
Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514
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ASTERISK-18995 #close
Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a
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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
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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
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Icelandic has some weird grammar rules when dealing with dates and
numbers. There are different genders used depending on which number
you're dealing with, and only a handful of numbers do change depending
on the gender. There is also an implied gender in several cases.
This patch was originally written for asterisk 1.6, and has been in use
for several years without crashes. I cleaned it up a bit and rewrote
what was necessary for Asterisk 13.
The functions were copied from other similar languages and modified
where appropriate. If i recall correctly, the German and Danish
functions were used as a base.
ASTERISK-26087
Reported by: Örn Arnarson
Tested by: Örn Arnarson
Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665
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Change the awkward and not as flexible UnicastRTP options format
From:
Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
To:
Dial(UnicastRTP/127.0.0.1[/[<options>]])
Where <options> can be standard Asterisk flag options:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
More option flags can be easily added later such as the codec's RTP
payload type to use when the codec does not have a static payload type
defined.
Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9
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