If any errors occur during the TLS connection setup, we currently dump a
fairly generic error message. So instead we try to pull in something
useful from OpenSSL to report instead.
ASTERISK-24712
Reported by: Matthias Urlichs
Change-Id: I288500991a9681f447d92913b11fedaf426087f4
SSL_connect returns non-zero for both success and some error conditions
so simply negating is inadequate.
Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1
chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL
(44) when a channel is hung up due to an RTP timeout. So do the same
when it happens with PJSIP for parity.
Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8
Updated the AMI version for the following reason (see CHANGES for more details):
The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now
contains a new optional parameter, 'MatchHeader'.
Change-Id: I9aeac4decc89f9b464b3f026e97c7ef1acc79242
After configuring Asterisk with '--with-pjproject-bundled' the configure/build
process attempts to download pjproject from its download site. Currently, a
timeout of 10 seconds is used that will stop the download process if pjproject
has not been fully downloaded in that time. For some systems this was not enough
time and the process was timing out too early.
This patch raises the download timeout value to '60'. Also, this patch fixes
another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported
due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to
DOWNLOAD_TIMEOUT.
ASTERISK-26814 #close
Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842
The documentation for JABBER_STATUS (and the deprecated JabberStatus
app) indicate that a return value of 7 indicates that the specified
buddy was not in the roster. It also indicates that you can specify a
"bare" JID (one without a resource). Unfortunately the actual behavior
does not match the documented behavior.
Assuming that our roster includes the buddy online and available
"valid@example.org/Valid" and does *not* include the buddy
"invalid@example.org", the JABBER_STATUS() function returns the
following before this patch:
+------------------------------+------------+--------------------------+
| Buddy | Status | Result |
+------------------------------+------------+--------------------------+
| valid@example.org | Online | 7 (Not in roster) |
| valid@example.org/Valid | Online | 1 (Online) |
| valid@example.org/Invalid | N/A | 7 (Not in roster) |
| invalid@example.org | N/A | Error logged, no return |
| invalid@example.org/Valid | N/A | Error logged, no return |
+------------------------------+------------+--------------------------+
And after this patch:
+------------------------------+------------+--------------------------+
| Buddy | Status | Result |
+------------------------------+------------+--------------------------+
| valid@example.org | Online | 1 (Online) |
| valid@example.org/Valid | Online | 1 (Online) |
| valid@example.org/Invalid | N/A | 6 (Offline) |
| invalid@example.org | N/A | 7 (Not in roster) |
| invalid@example.org/Valid | N/A | 7 (Not in roster) |
+------------------------------+------------+--------------------------+
This brings the behavior in line with the documentation.
ASTERISK-23510 #close
Reported by: Anthony Critelli
Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf
If we never establish a connection to our Jabber server, iksemel never sets up
its internal transport pointer, so attempting to send a message dereferences a
NULL pointer and causes a crash.
ASTERISK-21855 #close
Reported by: Jeremy Kister
Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c
The CDR code previously did not allow the user field to be set
from the 'h' extension in the dialplan. This change removes that
limitation and allows it to be set.
ASTERISK-26818
Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6
Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched
destinations) as currently provided by chan_sip is missing from res_pjsip.
This patch adds a new endpoint attribute (allow_overlap) [defaults to yes]
which when set to yes enables 484 responses to partial destination
matches rather than the current 404.
ASTERISK-26864
Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6
Rather than hard-coding UDP, allow consumers of the HEP API to specify
which protocol is in use. Update the PJSIP provider to pass in the
current protocol type.
ASTERISK-26850 #close
Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978
We aren't validating that the URI we just parsed is a SIP/SIPS one before
trying to access the user, host, and port members of a possibly uninitialized
structure.
Also update the MessageSend documentation to indicate what 'from' formats are
accepted.
ASTERISK-26484 #close
Reported by: Vinod Dharashive
Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30
Some codecs - codec_speex specifically - take voice frames and return
other types of frames, like CNG. If we subsequently treat those as
voice frames, we'll run into trouble when destroying the frame because
of the requirement that each voice frame have an associated format.
ASTERISK-26880 #close
Reported by: Kirsty Tyerman
Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c
Fixed a bug in function "ast_audiohook_write_frame" that checked the
variable other_factory_samples and only flushed the factories, so they
would be in sync, when other_factory_samples > 0. When there is not any
rtp incoming the variable other_factory_samples will be 0, and although
the result of "our_factory_ms - other_factory_ms" may be very large,
this led to the record file not syncing.
ASTERISK-26875 #close
Reported-by: Aaron An
Tested-by: Aaron An
Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22
POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.
Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.
Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48
We are currently passing in the capacity of the read buffer instead of the
number of bytes that we actually read off the wire.
Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36
Queue member will get stuck in pending_members if queue calls a device
that is different from the one observed for state changes.
This patch removes members from pending_members as a result of channel stasis
events such as blind or attended transfers and hangup.
ASTERISK-26862 #close
Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727
* Added CHANNEL(callid) to retrieve the call identifier log tag associated
with the channel. Dialplan now has access to the call log search key
associated with the channel so it can be saved in case there is a problem
with the call.
ASTERISK-26878
Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f
The queue_stasis_data structure contains various mutable fields that require
appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and
'caller_uniqueid' fields need to be locked when read from or written to.
Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088
We were inadvertenly referencing the cos_video option to determine if we
should set the tos_audio and cos_audio value on the RTP instance.
Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0
