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
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
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
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
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
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
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
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
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
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
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
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
When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and
minexpiry were not migrated to pjsip.conf.
ASTERISK-22374
Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b
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
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
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
Commit 1b666549f3 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
contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
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
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
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
This updates func_channel.c and main/message.c to use a generic xpointer
include instead of including info from each channel driver. Now the
name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
documentation for func_channel. Setting the name attribute of info to
MessageToInfo or MessageFromInfo causes it to be included in the
MessageSend application and AMI action.
Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea
zuul
Alexander Traud
Joshua Colp
Kevin Harwell
George Joseph
Richard Mudgett
Torrey Searle
Corey Farrell