This patch removes the following modules:
- pbx_functions: It never existed.
- res_pjsip_log_forwarder: It no longer exists.
- res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs
aren't going to be installing HOMER
- res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't
loaded, and we aren't configured to make use of the
module
Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5
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
Add the option 'enable_callee_prompt' to followme.conf. Enabled by
default. If disabled, a callee is not prompted to accept or reject
the forwarded call.
ASTERISK-26064 #close
Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
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
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
* No need to set language in a miniml configuration. 'en' will do just
fine.
* It would be useful to have an example of setting it to a different
language.
* Setting the documentation language explicitly is likewise not
required. Setting it to a different value is not common. At least
until there is a set of translated documentation.
Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
Note the default of remmed-out options. To clarify that those values are
not the defaults.
Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
A minimal configuration does not need to explicitly spell out the
directories. The built-in defaults will do just fine. In many cases
they are wrong.
Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
This patch allows for having app_confbridge register the name of the
conference as an extension into a specific context, similar to
regcontext for chan_sip. This variant is not quite as involved as the
one in chan_sip and doesn't allow for multiple contexts or custom
extensions, you can only specify the context and the conference name
will always be used as the extension to register.
ASTERISK-25989 #close
Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f
Note: When packagers use these files (as an example) the paths are never
really used when they are split using '='.
Note: Thirdparty applications will also have trouble parsing the file when
expecting '=>'.
Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004
A feature of chan_sip that service providers relied upon was the ability to
identify by the Authorization username. This is most often used when customers
have a PBX that needs to register rather than identify by IP address. From my
own experiance, this is pretty common with small businesses who otherwise
don't need a static IP.
In this scenario, a register from the customer's PBX may succeed because From
will usually contain the PBXs account id but an INVITE will contain the caller
id. With nothing recognizable in From, the service provider's Asterisk can
never match to an endpoint and the INVITE just stays unauthorized.
The fixes:
A new value "auth_username" has been added to endpoint/identify_by that
will use the username and digest fields in the Authorization header
instead of username and domain in the the From header to match an endpoint,
or the To header to match an aor. This code as added to
res_pjsip_endpoint_identifier_user rather than creating a new module.
Although identify_by was always a comma-separated list, there was only
1 choice so order wasn't preserved. So to keep the order, a vector was added
to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar
to find the aor. The res_pjsip_endpoint_identifier_* modules are called in
globals/endpoint_identifier_order.
Along the way, the logic in res_pjsip_registrar was corrected to match
most-specific to least-specific as res_pjsip_endpoint_identifier_user does.
The order is:
username@domain
username@domain_alias
username
Auth by username does present 1 problem however, the first INVITE won't have
an Authorization header so the distributor, not finding a match on anything,
sends a securty_alert. It still sends a 401 with a challenge so the next
INVITE will have the Authorization header and presumably succeed. As a result
though, that first security alert is actually a false alarm.
To address this, a new feature has been added to pjsip_distributor that keeps
track of unidentified requests and only sends the security alert if a
configurable number of unidentified requests come from the same IP in a
configurable amout of time. Those configuration options have been added to
the global config object. This feature is only used when auth_username
is enabled.
Finally, default_realm was added to the globals object to replace the hard
coded "asterisk" used when an endpoint is not yet identified.
The testsuite tests all pass but new tests are forthcoming for this new
feature.
ASTERISK-25835 #close
Reported-by: Ross Beer
Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
This patch added new global pjsip option 'disable_multi_domain'.
Disabling Multi Domain can improve Realtime performance by reducing
number of database requests.
ASTERISK-25930 #close
Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7
res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds
the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes
on endpoints for unsolicited mwi and on aors for subscriptions required
that the admin know in advance which the client wanted. If you specified
mailboxes on the endpoint, subscriptions were rejected even if you also
specified mailboxes on the aor.
Voicemail extension:
* Added a global default_voicemail_extension which defaults to "".
* Added voicemail_extension to both endpoint and aor.
* Added ast_sip_subscription_get_dialog for support.
* Added ast_sip_subscription_get_sip_uri for support.
When an unsolicited NOTIFY is constructed, the From header is parsed, the
voicemail extension from the endpoint is substituted for the user, and the
result placed in the Message-Account field in the body.
When a subscribed NOTIFY is constructed, the subscription dialog local uri
is parsed, the voicemail_extension from the aor (looked up from the
subscription resource name) is substituted for the user, and the result
placed in the Message-Account field in the body.
If no voicemail extension was defined, the Message-Account field is not added
to the NOTIFY body.
mwi_subscribe_replaces_unsolicited:
* Added mwi_subscribe_replaces_unsolicited to endpoint.
The previous behavior was to reject a subscribe if a previous internal
subscription for unsolicited MWI was found for the mailbox. That remains the
default. However, if there are mailboxes also set on the aor and the client
subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal
subscription is removed and replaced with the external subscription. This
allows an admin to configure mailboxes on both the endpoint and aor and allows
the client to select which to use.
ASTERISK-25865 #close
Reported-by: Ross Beer
Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea
There were a number of places in the res_pjsip stack that were getting
all endpoints or all aors, and then filtering them locally.
A good example is pjsip_options which, on startup, retrieves all
endpoints, then the aors for those endpoints, then tests the aors to see
if the qualify_frequency is > 0. One issue was that it never did
anything with the endpoints other than retrieve the aors so we probably
could have skipped a step and just retrieved all aors. But nevermind.
This worked reasonably well with local config files but with a realtime
backend and thousands of objects, this was a nightmare. The issue
really boiled down to the fact that while realtime supports predicates
that are passed to the database engine, the non-realtime sorcery
backends didn't.
They do now.
The realtime engines have a scheme for doing simple comparisons. They
take in an ast_variable (or list) for matching, and the name of each
variable can contain an operator. For instance, a name of
"qualify_frequency >" and a value of "0" would create a SQL predicate
that looks like "where qualify_frequency > '0'". If there's no operator
after the name, the engines add an '=' so a simple name of
"qualify_frequency" and a value of "10" would return exact matches.
The non-realtime backends decide whether to include an object in a
result set by calling ast_sorcery_changeset_create on every object in
the internal container. However, ast_sorcery_changeset_create only does
exact string matches though so a name of "qualify_frequency >" and a
value of "0" returns nothing because the literal "qualify_frequency >"
doesn't match any name in the objset set.
So, the real task was to create a generic string matcher that can take a
left value, operator and a right value and perform the match. To that
end, strings.c has a new ast_strings_match(left, operator, right)
function. Left and right are the strings to operate on and the operator
can be a string containing any of the following: = (or NULL or ""), !=,
>, >=, <, <=, like or regex. If the operator is like or regex, the
right string should be a %-pattern or a regex expression. If both left
and right can be converted to float, then a numeric comparison is
performed, otherwise a string comparison is performed.
To use this new function on ast_variables, 2 new functions were added to
config.c. One that compares 2 ast_variables, and one that compares 2
ast_variable lists. The former is useful when you want to compare 2
ast_variables that happen to be in a list but don't want to traverse the
list. The latter will traverse the right list and return true if all
the variables in it match the left list.
Now, the backends' fields_cmp functions call ast_variable_lists_match
instead of ast_sorcery_changeset_create and they can now process the
same syntax as the realtime engines. The realtime backend just passes
the variable list unaltered to the engine. The only gotcha is that
there's no common realtime engine support for regex so that's been noted
in the api docs for ast_sorcery_retrieve_by_fields.
Only one more change to sorcery was done... A new config flag
"allow_unqualified_fetch" was added to reg_sorcery_realtime.
"no": ignore fetches if no predicate fields were supplied.
"error": same as no but emit an error. (good for testing)
"yes": allow (the default);
"warn": allow but emit a warning. (good for testing)
Now on to res_pjsip...
pjsip_options was modified to retrieve aors with qualify_frequency > 0
rather than all endpoints then all aors. Not only was this a big
improvement in realtime retrieval but even for config files there's an
improvement because we're not going through endpoints anymore.
res_pjsip_mwi was modified to retieve only endpoints with something in
the mailboxes field instead of all endpoints then testing mailboxes.
res_pjsip_registrar_expire was completely refactored. It was retrieving
all contacts then setting up scheduler entries to check for expiration.
Now, it's a single thread (like keepalive) that periodically retrieves
only contacts whose expiration time is < now and deletes them. A new
contact_expiration_check_interval was added to global with a default of
30 seconds.
Ross Beer reports that with this patch, his Asterisk startup time dropped
from around an hour to under 30 seconds.
There are still objects that can't be filtered at the database like
identifies, transports, and registrations. These are not going to be
anywhere near as numerous as endpoints, aors, auths, contacts however.
Back to allow_unqualified_fetch. If this is set to yes and you have a
very large number of objects in the database, the pjsip CLI commands
will attempt to retrive ALL of them if not qualified with a LIKE.
Worse, if you type "pjsip show endpoint <tab>" guess what's going to
happen? :) Having a cache helps but all the objects will have to be
retrieved at least once to fill the cache. Setting
allow_unqualified_fetch=no prevents the mass retrieve and should be used
on endpoints, auths, aors, and contacts. It should NOT be used for
identifies, registrations and transports since these MUST be
retrieved in bulk.
Example sorcery.conf:
[res_pjsip]
endpoint=config,pjsip.conf,criteria=type=endpoint
endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error
ASTERISK-25826 #close
Reported-by: Ross Beer
Tested-by: Ross Beer
Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67
The 'reload' mechanism actually involves closing the underlying
socket and calling the appropriate udp, tcp or tls start functions
again. Only outbound_registration, pubsub and session needed work
to reset the transport before sending requests to insure that the
pjsip transport didn't get pulled out from under them.
In my testing, no calls were dropped when a transport was changed
for any of the 3 transport types even if ip addresses or ports were
changed. To be on the safe side however, a new transport option was
added (allow_reload) which defaults to 'no'. Unless it's explicitly
set to 'yes' for a transport, changes to that transport will be ignored
on a reload of res_pjsip. This should preserve the current behavior.
Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf
Previously you could add [!dnid] to the SIP dial string to alter the To:
header. This change allows you to alter the From header as well.
SIP dial string extra options now look like this:
[![touser[@todomain]][![fromuser][@fromdomain]]]
INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To:
header, that is no longer possible.
ASTERISK-25803 #close
Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7
Warnings and errors in the pjproject libraries are generally handled by
Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings
or errors so the messages emitted by pjproject directly are either superfluous
or misleading. A good exampe of this are the level-0 errors pjproject emits
when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't
consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS
client be treated any differently?
A config file for res_pjproject has bene added (pjproject.conf) and a new
log_mappings object allows mapping pjproject levels to Asterisk levels
(or nothing). The defaults if no pjproject.conf file is found are the same
as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR,
2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level>
Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898
* changes:
app_confbridge: Add ability to get the muted conference state.
app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.
app_confbridge: Make non-admin users join a muted conference muted.
During ICE negotiation the IPs of the local interfaces are sent to the remote
peer as host candidates. In many cases Asterisk is behind a static one-to-one
NAT, so these host addresses will be internal IP addresses.
To help in hiding the topology of the internal network, this patch adds the
ability to override the host candidates by matching them against a
user-defined list of replacements.
Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f
This change exposes the configuration of various aspects of the TLS
support and sets the default to the modern standards.
The TLS cipher is now set to the best values according to the
Mozilla OpSec team, different TLS versions can now be disabled, and
the cipher order can be forced to be that of the server instead of
the client.
ASTERISK-24972 #close
Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8
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
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
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
- 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
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
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
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
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
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
* 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
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
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
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
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
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
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
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
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
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
* 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
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
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>
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>
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
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
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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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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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The sample config was missing the configuration options for DTMF attended
transfer completion scenarios. The configuration options 'atxferabort',
'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the
appropriate configuration file.
ASTERISK-24678 #close
Reported by: Niklas Larsson
patches:
features.conf.sample.diff uploaded by Niklas Larsson (License 5068)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the ability to pass options and a command to MixMontor when
recording a conference using ConfBridge.
New options are -
* record_options: Options to MixMontor, eg: m(), W() etc.
* record_command: The command to execute when recording is over.
* record_file_timestamp: Append the start time to the file name.
These options can also be used with the CONFBRIDGE function, e.g.,
Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))
Review: https://reviewboard.asterisk.org/r/4023
ASTERISK-24351 #close
Reported by: Gareth Palmer
patches:
record_command-428838.patch uploaded by Gareth Palmer (License 5169)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_pjsip_config_wizard
------------------
* This is a new module that adds streamlined configuration capability for
chan_pjsip. It's targetted at users who have lots of basic configuration
scenarios like 'phone' or 'agent' or 'trunk'. Additional information
can be found in the sample configuration file at
config/samples/pjsip_wizard.conf.sample.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4190/
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Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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Configuration of DTLS in the general section will be applied to any users
or peers. If configuration exists at their level it overrides the general
section values.
ASTERISK-24128 #close
Reported by: Michael K.
patches:
dtls_default_settings.patch submitted by Michael K. (license 6621)
Review: https://reviewboard.asterisk.org/r/3867/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Testing has shown repeatedly that PJSIP's default behavior of switching
automatically to TCP for large messages can cause issues. The most common
issues are that devices that we are communicating with do not handle the
switch to TCP gracefully, thus causing situations such as broken calls or
broken subscriptions. Now, in order to have this behavior happen, you must
opt into it. The sample file has been updated to warn that enabling the
TCP switch behavior may cause issues for you, so use at your own risk.
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Virtual line support establishes a relationship between messages
related to an outbound registration and a local endpoint. This is
accomplished by attaching a parameter to the Contact of the outbound
registration and looking for it on any received requests. If the
parameter exists and can be matched to an outbound registration
the configured endpoint is associated with the request.
Review: https://reviewboard.asterisk.org/r/2964/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matt Jordan
zuul
Joshua Colp
Timo Teräs
Tzafrir Cohen
Alexei Gradinari
Tzafrir Cohen
Jaco Kroon
Diederik de Groot
George Joseph
Leif Madsen
Philip Correia
Walter Doekes
Sean Bright
Richard Mudgett
Mark Michelson
Daniel Journo
Ward van Wanrooij
Dade Brandon
Corey Farrell
Kevin Harwell
Rusty Newton
Alexander Traud
Rodrigo Ramírez Norambuena
Jonathan Rose
Ashley Sanders
Kinsey Moore
Olle Johansson