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ASTERISK-20987 #close
Reported by: hristo
Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38
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Since res_pjsip now depends on res_pjproject, this has been added to
basic-pbx modules.conf.
Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d
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Update the version number in the comments from Asterisk 12 to Asterisk 12+
Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b
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Added new global option (regcontext) to pjsip. When set, Asterisk will
dynamically create and destroy a NoOp priority 1 extension
for a given endpoint who registers or unregisters with us.
ASTERISK-25670 #close
Reported-by: Daniel Journo
Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62
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On a system with multiple ip addresses in the same subnet, if a
transport is bound to a specific ip address and endpoint/media_address
is set, the SIP/SDP will have the correct address in all fields but
the rtp stream MAY still originate from one of the other ip addresses,
most probably the "primary" ip address. This happens because
res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
the "all" ip address (0.0.0.0 or ::).
The new option causes res_pjsip_sdp_rtp/create_rtp to call
ast_rtp_instance_new with the endpoint's media_address (if specified)
instead of the "all" address. This causes the packets to originate from
the specified address.
ASTERISK-25632
ASTERISK-25637
Reported-by: Olivier Krief
Reported-by: Dan Journo
Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88
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In the sample sip.conf this is written with regard to notifyringing:
;notifyringing = no ; Control whether subscriptions already INUSE get sent
RINGING when another call is sent (default: yes)
However, this setting changes whether or not any RINGING indications are sent
to subscriptions. There is no separate configurable setting that allows
to control whether INUSE subscriptions also get sent RINGING. This is however
a useful option, to see (using BLF) if somebody else is able to handle an
incoming call or if everybody is busy.
This patch corrects the documentation for notifyringing (so the documentation
matches the functionality) and make notifyringing a tri-state option, by adding
the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing =
notinuse, only subscriptions that are not INUSE are sent the RINGING signal.
The default setting for notifyringing remains set to yes, so the default
behaviour is not affected.
ASTERISK-25558
Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa
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- The maximum_number_of_words was previously documented as being
the number of words that when exceeded, would result in the AMD
application returning that the audio represents a machine.
This was inconsistent with its actual functionality - it was
a number of words that when REACHED, would result in determination
as a machine.
This update corrects the functionality to match the previously
documented functionality. This is a backwards incompatible change
in configuration file, and has been added to UPGRADE.txt as a result.
The sample configuration file and application defaults have been updated
so that the default value is now 2, which reflects the same default
functionality as previous versions.
- Update documentation for silence_threshold, which previously implied
that it was measuring time, rather than noise averages in the sample.
- Update the comments in amd.conf.sample.
ASTERISK-25639 #close
Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093
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This option adds the ability to specify a timeout, in seconds, for a
participant in a ConfBridge. When the user's timeout has been reached,
the user is ejected from the conference with the CONFBRIDGE_RESULT
channel variable set to "TIMEOUT".
The rationale for this change is that there have been times where we
have seen channels get "stuck" in ConfBridge because a network issue
results in a SIP BYE not being received by Asterisk. While these
channels can be hung up manually via CLI/AMI/ARI, adding some sort of
automatic cleanup of the channels is a nice feature to have.
ASTERISK-25549 #close
Reported by Mark Michelson
Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98
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This patch adds a new setting "websockets_enabled" to sip.conf.
Setting this to false allows chan_sip to be used without causing
conflicts with res_pjsip_transport_websocket.
ASTERISK-24106 #close
Reported by: Andrew Nagy
Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7
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During outbound registration it is possible to receive a fatal (any permanent/
non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due
to a problem with the registrar itself. Upon receiving the failure response
Asterisk terminates outbound registration for the given endpoint.
This patch adds an option, 'fatal_retry_interval', that when set continues
outbound registration at the given interval up to 'max_retries' upon receiving
a fatal response.
ASTERISK-25485 #close
Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2
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When Asterisk is part of a larger distributed system, log files are often
gathered using tools (such as logstash) that prefer to consume information
and have it rendered using other tools (such as Kibana) that prefer a
structured format, e.g., JSON. This patch adds support for JSON formatted
logs by adding support for an optional log format specifier in Asterisk's
logging subsystem. By adding a format specifier of '[json]':
full => [json]debug,verbose,notice,warning,error
Log messages will be output to the 'full' channel in the following
format:
{
"hostname": Hostname or name specified in asterisk.conf
"timestamp": Date/Time
"identifiers": {
"lwp": Thread ID,
"callid": Call Identifier
}
"logmsg": {
"location": {
"filename": Name of the file that generated the log statement
"function": Function that generated the log statement
"line": Line number that called the logging function
}
"level": Log level, e.g., DEBUG, VERBOSE, etc.
"message": Actual text of the log message
}
}
ASTERISK-25425 #close
Change-Id: I8649bfedf3fb7bf3138008cc11565553209cc238
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This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
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Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19
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* In sip.conf.sample fix sentence where we said that WS or WSS are supported
transports for use in an outbound register definition. They are not
supported in that case.
* In func_cdr.c made it clear that the Disable option for CDR_PROP can be used
to enable CDR on a channel.
ASTERISK-24867 #close
Reported by: Rusty Newton
ASTERISK-24853 #close
Reported by: PSDK
Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca
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Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
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server socket."
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When a client connects to a server via SSL/TLS, the server commonly utilizes an
RSA key-pair. However, other such algorithms exist (i.e. DSA and ECDSA), and if
the server socket is configured with a certificate for either one of those, it
would lose its compatibility with RSA-only clients.
Now, the server socket can be configured with up to one RSA, ECDSA and DSA key
each. For example, if a client is not compatible with SHA-2 hashed certificates
like Nokia mobile phones, the server socket still can use RSA/SHA-1 for legacy
clients and ECDSA/SHA-2 for everyone else.
ASTERISK-24815 #close
Reported by: Alexander Traud
patches:
tls_rsa_ecc_dsa.patch uploaded by Alexander Traud (License 6520)
Change-Id: Iada5e00d326db5ef86e0af7069b4dfa1b979da9a
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The main IVR was playing demo-congrats. I've switched it over to the
basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt
has Allison prompting the user with the actual IVR menu.
ASTERISK-24892 #close
Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d
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ASTERISK-25034
Reported by: Richard Mudgett
Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30
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Add feature to set optional schema parameter on configuration file via
'schema' setting.
Fix query to get columns from table while considering schema. If in
the database there exists two tables with same name in distinct schemas
it will return an error when inserting record.
ASTERISK-24967 #close
Change-Id: I691fd2cbc277fcba10e615f5884f8de5d8152f2c
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Added the ability to set the character to quote identifiers. This
allows adding the character at the start and end of table and column
names. This setting is configurable for cdr_adaptive_odbc via the
quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
ASTERISK-25006
Change-Id: I0b9a56b79ca13a727a803d88ed3b8643e37632b8
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This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR
columns added in Asterisk 1.8. The columns are:
* peeraccount
* linkedid
* sequence
When enabled, the columns in the database entry will be populated with the data
from the CDR.
ASTERISK-24976 #close
Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b
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The sample pjsip.conf has a few comment lines that are missing the
semicolons at the start of the comment, causing the config to fail
load.
Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352
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Some telco switches occasionally ignore ISDN RESTART requests. The fix
for ASTERISK-19608 added an escape clause for B channels in the restarting
state if the telco ignores a RESTART request. If the telco fails to
acknowledge the RESTART then Asterisk will assume the telco acknowledged
the RESTART on the second call attempt requesting the B channel by the
telco. The escape clause is good for dealing with RESTART requests in
general but it does cause the next call for the restarting B channel to be
rejected if the telco insists the call must go on that B channel.
chan_dahdi doesn't really need to issue a RESTART request in response to
receiving a cause 44 (Requested channel not available) code. Sending the
RESTART in such a situation is not required (nor prohibited) by the
standards. I think chan_dahdi does this for historical reasons to deal
with buggy peers to get channels unstuck in a similar fashion as the
chan_dahdi.conf resetinterval option.
* Add the chan_dahdi.conf force_restart_unavailable_chans compatability
option that when disabled will prevent chan_dahdi from trying to RESTART
the channel in response to a cause 44 code.
ASTERISK-25034 #close
Reported by: Richard Mudgett
Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65
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This switches files used to generate other sources to use the new
ASTERISK_REGISTER_FILE macro.
ASTERISK-25026 #close
Reported by: Corey Farrell
Change-Id: Ieb2537b83421cad07c8955e5f90c405ccf079740
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Version"
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* The REF_DEBUG compiler flag no longer has any effect on code that uses
Astobj2. It is used to determine if reference debugging is enabled by
default. Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
This was possible now that we no longer require a dual ABI.
ASTERISK-24974 #close
Reported by: Corey Farrell
Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1
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Add new column to INSERT new columns added in cdr 1.8 version. The columns are:
* peeraccount
* linkedid
* sequence
This feature is configurable in cdr_odbc.conf using a new configuration
option, 'newcdrcolumns'.
ASTERISK-24976 #close
Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127
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Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup. So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.
This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies. This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.
If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random(). If not set,
qualify_timeout is used.
The default is "0" (disabled).
ASTERISK-24863 #close
Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: George Joseph <george.joseph@fairview5.com>
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This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint. Only dynamic contact add/delete actions
update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.
This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...
1. A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message. The default is 3000ms. When the timer expires, the contact is
marked unavailable.
2. Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'. When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
existing endpoint events are generated appropriately.
ASTERISK-24863 #close
Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>
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After the "progressinband" value setting of "never" was updated to never send a
183 this separated its use from the "no" value. Since "never" was the default,
but most users probably expect "no" this patch updates the default for the
"progressinband" setting to "no."
ASTERISK-24835 #close
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4606/
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This patch adds a new configuration parameter, 'calltokenexpiration', that
controls how long before an authentication call token is expired. The default
maintains the RFC specified 10 seconds. Setting it to a higher value may be
useful in lossy networks.
Review: https://reviewboard.asterisk.org/r/4588
ASTERISK-24939 #close
Reported by: Y Ateya
patches:
ctoken_configuration.diff submitted by Y Ateya (License 6693)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds a new option to cel_pgsl, "usegmtime", which causes timestamps
to be logged in GMT.
Review: https://reviewboard.asterisk.org/r/4571/
ASTERISK-23186 #close
Reported by: Rodrigo Ramirez Norambuena
patches:
cel_pgsql.c_add_usegmtime2.patch submitted by Rodrigo Ramirez Norambuena (License 6577)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Review: https://reviewboard.asterisk.org/r/4504/
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Example configuration files for a "basic PBX" deployment for the fictitious
Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/
and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
Patch 4488 includes all functionality needed for SAC's outside connectivity
and some externally accessed features, as well as outbound dialing.
Reported by: Malcolm Davenport
Tested by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/4488/
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This change adds an abstracted core DNS API which resembles the API described
here[1]. The API provides a pluggable mechanism for resolvers and also a
consistent view for records. Both synchronous and asynchronous queries are
supported.
This change also adds a res_resolver_unbound module which uses the libunbound
library to provide resolution.
Unit tests have also been written for all of the above to confirm the API and
functionality.
ASTERISK-24834 #close
Reported by: Matt Jordan
ASTERISK-24836 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4474/
Review: https://reviewboard.asterisk.org/r/4512/
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API
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messages.
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
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This patch fixes previously reverted code that caused binary incompatibility
problems with some modules. And like the original patch it makes sure that
no matter what order the endpoint identifier modules were loaded, priority is
given based on the ones specified in the new global 'endpoint_identifier_order'
option.
ASTERISK-24840
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4489/
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This patch corrects the documentation for the AMD application. Specifically:
* It documents the maximum_word_length option, which limits the maximum allowed
length of a single utterance.
* It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH
was documented as MAXWORDS, while MAXWORDS was undocumented.
Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues.
ASTERISK-19470 #close
Reported by: Frank DiGennaro
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Due to a break in binary compatibility with some other modules these changes
are being reverted until the issue can be resolved.
ASTERISK-24840
Reported by: Mark Michelson
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It's possible to have a scenario that will create a conflict between endpoint
identifiers. For instance an incoming call could be identified by two different
endpoint identifiers and the one chosen depended upon which identifier module
loaded first. This of course causes problems when, for example, the incoming
call is expected to be identified by username, but instead is identified by ip.
This patch adds a new 'global' option to res_pjsip called
'endpoint_identifier_order'. It is a comma separated list of endpoint
identifier names that specifies the order by which identifiers are processed
and checked.
ASTERISK-24840 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4455/
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Example configuration files for a "basic PBX" deployment for the fictitious
Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4379/
and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
Reported by: Malcolm Davenport
Tested by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/4379/
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The "contact" object is not meant to be configured from the pjsip.conf
configuration file. It is meant to be created as a result of a registration
and stored elsewhere.
ASTERISK-24085 #close
Reported by: Rusty Newton
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Looking at the Super Awesome Company sample reminded me that creating hints is
just plain gruntwork. So you can now have the pjsip conifg wizard auto-create
them for you.
Specifying 'hint_exten' in the wizard will create
'exten => <hint_exten>,hint/PJSIP/<wizard_id>'
in whatever is specified for 'hint_context'.
Specifying 'hint_application' in the wizard will create
'exten => <hint_exten>,1,<hint_application>'
in whatever is specified for 'hint_context'.
The default for 'hint_context' is the endpoint's context.
There's no default for 'hint_application'. If not specified, no app is added.
There's no default for 'hint_exten'. If not specified, neither the hint itself
nor the application will be created.
Some may think this is the slippery slope to users.conf but hints are a basic
necessity for phones unlike voicemail, manager, etc that users.conf creates.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4383/
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Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. In this version, [servername] is uncommented by default.
ASTERISK-24316 #close
Reported By: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4374/
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across a bridge.
Calling ast_channel_bridge_peer() cannot be done while holding any channel
locks. The reported issue hit the deadlock in chan_iax2, but an audit of
the ast_channel_bridge_peer() calls found three more locations where the
same deadlock can occur.
* Made CHANNEL(peer), res_fax, and the SNMP agent not call
ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I
had to rework the logic to not hold the channel lock.
* Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done
for legacy reasons that no longer apply.
* Removed the iax.conf forcejitterbuffer option. It is now always enabled
when the jitterbuffer option is enabled. If you put a jitter buffer on a
channel it will be on the channel.
ASTERISK-24600 #close
Reported by: Jeff Collell
Review: https://reviewboard.asterisk.org/r/4342/
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Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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