Age | Commit message (Collapse) | Author |
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When res_odbc_transaction depended on res_odbc, it got the generic_odbc
headers and libs implicitly. Now that it no longer depends on res_odbc,
its dependency on generic_odbc must be explicit.
Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911
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PJProject supports a lot of platforms even Windows, some with different defaults
when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS,
"/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows.
Because of this, if configured with just an IPv6 address/transport, PJProject
listens to both IPv4 and IPv6. However, this is not supported by the PJProject
team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP,
incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only
server which accepts incoming connections on IPv4.
If you try to configure two transports, one with IPv4 and one with IPv6 on the
same interface, as expected by the PJProject team, the IPv4 transport is not
able to bind because the IPv6 transport listens to both already.
One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide.
Then, you are able to configure two transports, one for each IP version on the
same interface. That way, you get a server which works with IPv4 clients and
IPv6 clients at the same time over the same interface.
Here, this change sets this parameter directly within PJProject to match the
expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack
servers out of the box like in chan_sip. This change was accepted by the
PJProject team as <http://trac.pjsip.org/repos/changeset/5403> and is expected
to arrive in the next version, PJProject 2.6.0. Until then, this change is
incorporated in the bundled PJProject of Asterisk.
ASTERISK-26309
Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae
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res_odbc_transaction" into 13
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brief" into 13
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The compilation failed for devmode
--enable DONT_OPTIMIZE
--enable BETTER_BACKTRACES
--enable DO_CRASH
--enable TEST_FRAMEWORK
res_pjsip/pjsip_configuration.c: In function dtls_handler:
res_pjsip/pjsip_configuration.c:974:20: error:
back may be used uninitialized in this function [-Werror=maybe-uninitialized]
int size = strlen(front);
^
cc1: all warnings being treated as errors
Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580
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When using the migration script sip_to_pjsip.py, cert_file was not migrated to
pjsip.conf. A previous change regarding this contained a copy/paste error.
ASTERISK-22374
Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b
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When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL
SSLv23_method. This was documented incorrectly in the file sip.conf.sample.
SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method
enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that
function should have been called 'secure_method' or 'automatic_method' back in
the 90s.
Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if
you face a server which has problems like not falling back to TLSv1.0
automatically.
ASTERISK-24425
Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3
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Given resource paths did not have 'json' substituted in for the '{format}'. For
some auto generated documentation/comment strings it resulted in something like
the following:
"... REST handler for /api-docs/sounds.{format}"
This patch makes sure the resource api's path is properly substituted.
ASTERISK-25472 #close
Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23
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Historically, Asterisk has always specified annexb=no for the g729 format.
However, when using res_pjsip no format attribute was specified. This patch
makes it so the SDP now contains a format attribute line with annexb=no.
Note, that this means only g729a is negotiated. Even for pass through support.
According to rfc7261 the type of annex used (a or b) is dependent upon the
answerer. However, Asterisk being a back to back user agent makes this tricky
to support at this time, thus we only allow annex 'a' for now.
ASTERISK-26228 #close
patches:
res_format_attr_g729.c submitted by Jason Parker (license 4993)
Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0
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The MODULEINFO dependencies between these 2 modules was reversed.
res_odbc should depend on res_odbc_transaction, not the other way
around.
ASTERISK-25984 #close
Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f
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A recent update had a copy/paste error where the unused variable 'val' was
being passed to the set_value function instead of the 'method' value itself.
This patch passes in the right variable.
ASTERISK-22374
Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06
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contact_permit." into 13
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into 13
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When using the migration script sip_to_pjsip.py and tlsclientmethod is not set
in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to
overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is
offering/using not just TLSv1.0 but TLSv1.2 as well.
ASTERISK-22374
Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f
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When using the migration script sip_to_pjsip.py, no section of type=system or
type=general were created. Therefore the keys compactheaders, timerb, timert1,
and useragent were not migrated to pjsip.conf.
ASTERISK-22374
Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1
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When using the migration script sip_to_pjsip.py, session-timers=accept and
session-timers=refuse were mapped to wrong values.
ASTERISK-22374
Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092
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When using the migration script sip_to_pjsip.py, now the (mandatory) username is
written to pjsip.conf, even if there was no (optional) authname in the register
string in sip.conf.
ASTERISK-22374
Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f
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When using the migration script sip_to_pjsip.py and the register string
started with a transport in sip.conf - like tls://... - register was not parsed
correctly and therefore not migrated correctly to pjsip.conf.
ASTERISK-22374
Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2
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When using the migration script sip_to_pjsip.py, those keys got missing. These
keys might appear several times and the function "merge_value" tried to collect
those. However, because these keys have different names in sip.conf and
pjsip.conf, "merge_value" was not able to find the new key name in sip.conf.
This change lets "merge_value" search with the old key name in sip.conf and
write with the new key name in pjsip.conf.
ASTERISK-22374
Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2
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When using the migration script sip_to_pjsip.py, the externhost or externip of
sip.conf were erroneously written to Endpoints instead to Transports.
ASTERISK-22374
Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4
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When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and
minexpiry were not migrated to pjsip.conf.
ASTERISK-22374
Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b
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When using the migration script sip_to_pjsip.py, encryption=yes got missing and
media_encryption=sdes was not written to pjsip.conf, because of a typo.
ASTERISK-22374
Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05
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When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got
missed, because of a typo. Therefore, cos and tos were not written to
pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused
by a copy-and-paste error.
ASTERISK-22374
Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2
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When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were
not migrated to pjsip.conf.
ASTERISK-22374
Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825
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Commit 1b666549f33d69dc080b212bf92126f3bc3a18b2 broke the srv failover
functionality if a TCP connection gets disconnected. Under these
conditions, session_inv_on_state_changed() gets a
PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new
transport. Unfortunately, session_inv_on_tsx_state_changed() also gets
the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates
the session.
* Made session_inv_on_tsx_state_changed() complete terminating the session
on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still
PJSIP_INV_STATE_DISCONNECTED.
ASTERISK-26305 #close
Reported by: Richard Mudgett
Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d
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contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
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updated the uri handler to include the url prefix of the http server
this enables res_ari to add it to the uris when generating docs
Change-Id: I279335a2625261a8492206c37219698f42591c2e
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Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the
bundled PJProject but not in an external PJProject. Therefore, ca_list_path
could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2
is detected again to enable ca_list_path again.
ASTERISK-26303 #close
Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0
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Added to api.wiki.mustache so that the generated object pages
have the notation in the table header as well as under each method
that has path parameters.
ASTERISK-25492 #close
Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf
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