Incrementing stasis_app_playback.media_index directly in our playback
loop means that when we reach the end of our playlist the index into
the vector will be outside of the bounds of the vector.
Instead use a temporary variable and only assign when we're sure that
we are in bounds.
ASTERISK-28713 #close
Reported by: Sébastien Duthil
Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928
We need to wait for the message sending callback to finish to know if
we succeeded or failed.
ASTERISK-25421 #close
Reported by: Dmitriy Serov
Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059
Fixes no-audio issues when the media source is changed and
strictrtp is enabled (default).
If the peer media source changes, the SDP session version also changes.
If it is lower than the one we had stored, chan_sip would ignore it.
This changeset keeps track of the remote media origin identifier,
comparing that as well. If it changes, the session version needn't be
higher for us to accept the SDP.
Common scenario where this would've caused problems: a separate media
gateway that informs the caller about premium rates before handing off
the call to the final destination.
(An alternative fix would be to set ignoresdpversion=yes on the peer.)
ASTERISK-28686
Change-Id: I88fdbc5aeb777b583e7738c084254c482a7776ee
If chan_pjsip receives an RTP packet whose payload differs from the
channel's native format, and asymmetric_rtp_codec is disabled (the
default), Asterisk will switch the channel's native format to match
that of the incoming packet without regard to the negotiated payloads.
We now check that the received frame is in a format we have negotiated
before switching payloads which results in these packets being dropped
instead of causing the session to terminate.
ASTERISK-28139 #close
Reported by: Paul Brooks
Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3
When Alice calls Bob and Bob does a blind transfer to Charlie,
Bob's bridge leave event generates a finalize on both the party_a
and party_b CDRs but while the party_a CDR has the correct end time
set from the event time, party_b's leg did not. This caused that
CDR's end time to be equal to the answered time and resulted in a
billsec of 0.
* We now pass the bridge leave message event time to
cdr_object_party_b_left_bridge_cb() and set it on that CDR before
calling cdr_object_finalize() on it.
NOTE: This issue affected transfers using chan_sip most of the
time but also occasionally affected chan_pjsip probably due to
message timing.
ASTERISK-28677
Reported by: Maciej Michno
Change-Id: I790720f1e7326f9b8ce8293028743b0ef0fb2cca
Add a new configuration option 'enable_status' which allows the
/httpstatus URI handler to be administratively disabled.
We also no longer unconditionally register the /static and /httpstatus
URI handlers, but instead do it based upon configuration.
Behavior change: If enable_static was turned off, the URI handler was
still installed but returned a 403 when it was accessed. Because we
now register/unregister the URI handlers as appropriate, if the
/static URI is disabled we will return a 404 instead.
Additionally:
* Change 'enablestatic' to 'enable_static' but keep the former for
backwards compatibility.
* Improve some internal variable names
ASTERISK-28710 #close
Change-Id: I647510f796473793b1d3ce1beb32659813be69e1
The no-entry timeout set to 999999 == 16⅔ minutes, change to INT_MAX
to match behavior of "no timeout" defined in comment.
ASTERISK-28702 #close
Change-Id: I4ea015986e061374385dba247b272f7aac60bf11
SILK @ 24kHz is not shown in the 'core show translation' output because of an
off-by-one-error. Discovered while looking into ASTERISK~19871.
ASTERISK-28706
Reported by: Sean Bright
Change-Id: Ie1a551a8a484e07b45c8699cc0c90f1061029510
In af90afd90c, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.
Whitespace only, no functional change.
ASTERISK~23324
Reported by: Kevin McCoy
Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87
lws2sws() does not stop trying to handle header continuation lines
even after all headers have been found. This is problematic if the
first character of a SIP message body is a space or tab character, so
we update to recognize the end of the message header.
ASTERISK-28693 #close
Reported by: Frank Matano
Change-Id: Idec8fa58545cd3fd898cbe0075d76c223f8d33df
ast_store_realtime() is not NULL tolerant, so we need to initialize
the field values we pass to it to the empty string to avoid a crash.
ASTERISK-23739 #close
Reported by: Stas Kobzar
Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c
If voicemail.conf exists but is empty, the config parsing process will
default a number of global variables to non-zero values. On the other
hand, if voicemail.conf is missing (arguably semantically equivalent
to an empty file), this process is skipped and the globals are
defaulted to 0.
Set the globals to the same values they would be set to if a
configuration were present. This allows voicemail configuration to be
done completely by Realtime without the need to create an empty
voicemail.conf file.
ASTERISK-27622 #close
Reported by: Jim Van Meggelen
Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409
The QueueMemberPause AMI event includes two fields that return the
reason a member was paused.
* In release branches, deprecate Reason in favor of PausedReason.
* In master, remove the Reason field entirely.
ASTERISK-28349 #close
Reported by: Niksa Baldun
Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296
The change in 9b99ef50b5 updated the
syntax of the 'realtime update2' CLI command but did not correctly
update the calls to ast_update2_realtime().
The issue this addresses was originally opened because we aren't
allowing a SQL NULL to be set as part of the update, but this is a
limitation of the Realtime API and is not a bug.
Additionally, this patch:
* Corrects the example in the command documentation to reflect
'update2' instead of 'update.'
* Fixes the leading spacing of the command documentation.
* Checks that the required 'NULL' literal argument is present where we
expect it to be.
ASTERISK-21794 #close
Reported by: Cédric Bassaget
Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd
We allow for 'maxredirs' to be set, but this value is ignored when
followlocation is not enabled which, by default, it is not.
ASTERISK-17491 #close
Reported by: candrews
Change-Id: I96a4ab0142f2fb7d2e96ff976f6cf7b2982c761a
ast_addressfamily_to_sockaddrize will determine the size that's
required, and ast_sockaddr_from_sockaddr then wraps this new function
and ast_sockaddr_copy_sockaddr to copy arbitrary sockaddr's (without
knowing the address family) into the ast_sockaddr structure.
Change-Id: Iee604e96e9096c79b477d6e5ff310cf0b06dae86
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
Additionally alter the warning to mention that it was "beep" which could
not be streamed to give admins a better clue about what the warning
means.
ASTERISK-28682
Change-Id: If5aed21226a173117ed17589f44826dd1ba6576e
This patch fixes some wrongly formatted documentation for the AgentLogin
application. A couple of "see also" links should contain only the function
name, and no parameters.
Change-Id: I3f788b47dce3292e311f8a9856938d59a0bd0661
If you do a "make all" when building Asterisk the xml documentation
produced will be missing certain AMI events where their
documentation is located not at the top of the c source file but
embedded further down next to the event's manager_event()
registration call. See main/manager_mwi.c for an example.
"make full" does produce the correct documentation so we're changing
it in the build script. A separate commit/issue will address the
problem with "make all".
ASTERISK-28507
Reported by: David Lee
Change-Id: I4a22635d6eef99eacecc0efb69e28360eebdb86c
