When option 'o' was not set, ChanSpy created its audiohook with the flag
AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one
direction only.
ASTERISK-25866 #close
Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0
With the old SIP module AMI sends PeerStatus event on every
successfully REGISTER requests, ie, on start registration,
update registration and stop registration.
With PJSIP AMI sends ContactStatus only when status is changed.
Regarding registration:
on start registration - Created
on stop registration - Removed
but on update registration nothing
This patch added contact.updated event.
ASTERISK-25904
Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f
For all OSes:
* Disabled third-party codecs in pjproject and added
'--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
configure options since we don't use the pjsip codec capability.
FreeBSD:
* Added FreeBSD support to install_prereq.
* Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
* Added __progname and environ to asterisk.exports.in.
* Reverted the use of ldconfig to create shared library symlinks to ln.
* Only enable epoll in pjproject if `uname -s` is Linux.
* Added a patch to pjproject to take the name of the 'make' command from
an environment variable if supplied. This is needed for the python bindings.
(merged by Teluu into pjproject trunk 5/3/2016)
FreeBSD support isn't complete. Still some general issues regarding
make/gmake having nothing to do with pjproject. With some handholding it DOES
build successfully.
CentOS:
Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
CentOS 6/7 32/64 build and run the pjsip testsuite successfully.
Ubuntu:
No changes required.
Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.
Debian:
No changes required.
Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.
There will utimately be a follow-up patch to create an install_prereq for
the testsuite as I've discovered a few missing requirements.
ASTERISK-25968 #close
Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c
Voicemail email addresses can be corrupt or voicemail
emails can end up being sent to the wrong email address if asterisk is
reading voicemail.conf during a reload and processing an email at the
same time. This patch always copies the struct that would otherwise only
be copied once.
ASTERISK-24463 #close
Reported by: John Campbell
Tested by: Etienne Lessard
Tested by: Andrew Nagy
Change-Id: I3a0643813116da84e2617291903d0d489b7425fb
If the Asterisk system name is set in asterisk.conf, it will be stored
into the "reg_server" field in the ps_contacts table to facilitate
multi-server setups.
ASTERISK-25931
Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8
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
Create PUBLISH messages to update a third party when an extension state
changes because of either a device or presence state change.
A configuration example:
[exten-state-publisher]
type=outbound-publish
server_uri=sip:instance1@172.16.10.2
event=presence
; Optional regex for context filtering, if specified only extension state
; for contexts matching the regex will cause a PUBLISH to be sent.
@context=^users
; Optional regex for extension filtering, if specified only extension
; state for extensions matching the regex will cause a PUBLISH to be sent.
@exten=^[0-9]*
; Required body type for the PUBLISH message.
;
; Supported values are:
; application/pidf+xml
; application/xpidf+xml
; application/cpim-pidf+xml
; application/dialog-info+xml (Planned support but not yet)
@body=application/pidf+xml
The '@' extended variables are used because the implementation can't
extend the outbound publish type as it is provided by the outbound publish
module. That means you either have to use extended variables, or
implement some sort of custom extended variable thing in the outbound
publish module. Another option would be to refactor that stuff to have an
option which specifies the use of an alternate implementation's
configuration and then have that passed to the implementation. JColp
opted for the extended variables method originally.
ASTERISK-25972 #close
Change-Id: Ic0dab4022f5cf59302129483ed38398764ee3cca
When starting the extension state publishers, check if the requested
message body generator is available. If not available give error message
and skip starting that publisher.
* res_pjsip_pubsub.c: Create new API if type/subtype generator
registered.
* res_pjsip_exten_state.c: Use new body generator API for validation.
ASTERISK-25922
Change-Id: I4ad69200666e3cc909d4619e3c81042d7f9db25c
When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled,
the input uri string will become corrupted if it contains escape sequences.
It's not possible to automatically strdup or strdupa the input string because
the output uri pj_str_t's will have pointers to chunks of the input string.
Getting around this would require more memory management code and wouldn't
be worth the savings of doing the unescape in place.
ASTERISK-25970 #close
Reported-by: Dmitriy Serov
Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88
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
A recent change to func_odbc made it so that a single connection was
maintained per DSN. The problem was that the code was optimistic about
the health of the connection after initially opening it and did nothing
to re-connect in case the connection had died.
This change adds a check before executing a query to ensure that the
connection to the database is still up and running.
ASTERISK-25963 #close
Reported by Ross Beer
Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d
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
ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC
and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when
queues grow too large or when read and write queues go out of sync.
Now these flags are set conditionally:
- AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set
- a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not
be set on the audiohook
ASTERISK-25866
Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd
This change adds the ability to configure outbound publishing of
extension state. Right now stuff is merely set up to store the
configuration and to register a global extension state callback. The
act of constructing the body and sending is not yet complete.
Configurable elements right now are a regex for filtering the context,
a regex for filtering the extension, and the body type to publish.
ASTERISK-25922 #close
Change-Id: Ia7e630136dfc355073c1cadff8ad394a08523d78
The unload process currently tells each TCP/TLS to terminate but
does not wait for them to do so. This introduces a race condition
where the container holding the threads may be destroyed before
the threads are able to remove themselves from it. When they
finally do the container is invalid and can't be used causing a
crash.
A previous change existed which waited a bit to wait for any
stranglers to finish. This change extends this and waits longer.
ASTERISK-25961 #close
Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6
When unloading the app_queue module the members in each queue are
destroyed and as part of this they are removed from the pending
members container. Unfortunately a crash would occur as the container
was destroyed before the members were removed.
This change tweaks ordering so the container destruction occurs
after the members are destroyed.
ASTERISK-16115
Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b
* changes:
test_message.c: Wait longer in case dialplan also processes the test message.
Manager: Short circuit AMI message processing.
manager.c: Eliminate most RAII_VAR usage.
A patch I did back in 2014 modified ast_config_text_file_save2 to check the
writability of the main file and include files before truncating and re-writing
them. An unintended side-effect of this was that if a file doesn't exist,
the check fails and the write is aborted.
This patch causes ast_config_text_file_save2 to check the writability of the
parent directory of missing files instead of checking the file itself. This
allows missing files to be created again. A unit test was also added to
test_config to test saving of config files.
The regression was discovered when app_voicemail's passwordlocation=spooldir
feature stopped working.
ASTERISK-25917 #close
Reported-by: Jonathan Rose
Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80
