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The same thing was happening in res_pjsip_publish_asterisk. When the library
was unloaded it did not unregister the object type from sorcery. Subsequent
loads resulted in a failed load due to the sorcery type already existing.
Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9
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into 13
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profiles." into 13
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verification" into 13
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From the issue reporter:
"res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of
the timestamp, the source address, the source port, a server UUID that is
calculated at startup, and the authentication realm.
Rather than caching nonces that we create, we instead attempt to re-calculate
the nonce when receiving an incoming request with authentication. We then
compare the re-calculated nonce to the incoming nonce, and if they don't match,
then authentication has failed early.
The problem is that it is possible, especially when using TCP, to receive two
requests from the same endpoint but have differing source ports for those
requests. Asterisk itself commonly will use different source ports for
outbound TCP requests."
This patch removes the source port dependency when building the nonce.
ASTERISK-25978 #close
Change-Id: I871b5f4adce102df1c4988066283095ec509dffe
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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
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The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
The python bindings are now built only if TEST_FRAMEWORK is set and a
python development package is installed.
libresample was also disabled.
ASTERISK-25993 #close
Reported-by: Joshua Colp
Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03
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The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_*
and res_pjsip_registrar modules should load ASAP
to avoid "No matching endpoint found" for legitimate endpoint.
ASTERISK-25994
Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b
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ASTERISK-25956 #close
Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38
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ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail.
ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail.
These additions should be also in stasis_endpoints
to include in command "manager show event ContactStatus"
Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a
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It is possible for the nativeformats of a channel to change
throughout its lifetime. As a result a user of it needs to either
ensure the channel is locked when accessing the formats or keep
a reference to the nativeformats themselves.
This change fixes the file playback support so it keeps a
reference to the nativeformats when accessing things.
ASTERISK-25998 #close
Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915
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ASTERISK-25931
Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549
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into 13
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into 13
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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
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The app_fax set FAXMODE variable, but res_fax missing this feature.
This patch add FAXMODE variable which is set to either "audio" or "T38".
ASTERISK-25980
Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b
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The channel and peer V.21 sessions are created on the same channel now.
The peer V.21 session should be created only on peer channel
when one of channel can handle T.38.
Also this patch enable debug for T.38 gateway session
if global fax debug enabled.
ASTERISK-25982
Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e
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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
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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
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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
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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
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Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb
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Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a
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Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67
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Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632
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Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353
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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
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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
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files" into 13
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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
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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
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into 13
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