Most channel drivers let you specify a default accountcode to be set on
channels associated with a particular peer/endpoint/object. Prior to this
patch, chan_pjsip/res_pjsip did not support such a setting.
This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'.
When a channel is created that is associated with an endpoint with this value
set, the channel will automatically have its accountcode property set to the
value configured for the endpoint.
Review: https://reviewboard.asterisk.org/r/3724/
ASTERISK-24000 #close
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Holding a reference to the peer during mwi subscriptions
resulted in a circular reference because the final event
message would not be sent until destruction of the peer.
Instead, pass the name of the peer to the event callback
so that it can fail gracefully after the peer has gone.
ASTERISK-23959
Review: https://reviewboard.asterisk.org/r/3754/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Create a Stasis bridge sub-class to propagate linkedids and
accountcodes.
* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.
* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.
* Fixed the basic bridge sub-class to not call the base bridge class pull
method twice.
AFS-105 #close
ASTERISK-23852 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3720/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The dtls_perform_handshake function was mistakenly placed under the guards for
USE_PJPROJECT. If PJPROJECT was not installed, the function would not be
defined, while other functions would attempt to still use it. This prevented
res_rtp_asterisk from being loaded.
ASTERISK-24001 #close
Reported by: Don Fanning
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two bugs:
1. When originating a channel into a Stasis application, we already create a
subscription for the channel that is going into our Stasis app.
Unfortunately, when you create a Local channel and pass it off to a Stasis
app, you really aren't creating just one channel: you're creating two. This
patch snags the second half of the Local channel pair (assuming it is a
Local channel pair, but luckily core_local is kind about such assumptions)
and subscribes to it as well.
2. Subscriptions are a bit sticky right now. If a subscription is made, the
'interest' count gets bumped on the Stasis subscription - but unless
something explicitly unsubscribes the channel, said subscription sticks
around. This is not much of a problem is a user is creating the subscription
- if they made it, they must want it. However, when we are creating
implicit subscriptions, we need to make sure something clears them out.
This patch takes a pessimistic approach: it watches the cache updates
coming from Stasis and, if we notice that the cache just cleared out an
object, we delete our subscription object. This keeps our ao2 container of
Stasis forwards in an application from growing out of hand; it also is a
bit more forgiving for end users who may not realize they were supposed to
unsubscribe from that channel that just hung up.
Review: https://reviewboard.asterisk.org/r/3710/
ASTERISK-23939 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This corrects two issues with the extra field information in Asterisk
12+ in channel event logs.
It is possible to inject custom values into the dialstatus provided by
ast_channel_dial_type() Stasis messages that fall outside the
enumeration allowed for the DIALSTATUS channel variable. CEL now
filters for the allowed values and ignores other values.
The "hangupsource" extra field key is always blank if the far end
channel is a chan_pjsip channel. This is because the hangupsource is
never set for the pjsip channel driver. This change sets the
hangupsource whenever a hangup is queued for chan_pjsip channels.
This corrects an issue with the pjsip channel driver where the
hangupcause information was not being set properly.
Review: https://reviewboard.asterisk.org/r/3690/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@418071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new inband_on_setup_ack option causes Asterisk to assume inband audio
may be present when a SETUP_ACKNOWLEDGE message is received.
Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
dialtone is sent from the network side, progress indicator 8 "Inband info
now available" MAY be sent to the CPE if no digits were received with the
SETUP. It is thus implied that the ie is mandatory if digits came with
the SETUP and dialtone is needed. This option should be enabled, when the
network sends dialtone and you want to hear it, but the network doesn't
send the progress indicator when needed.
NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
dialing is also enabled because Q.SIG does not send the progress indicator
with the SETUP ACK.
The commit -r413714 (AST-1338) which causes this issue was dealing with a
SIP-to-ISDN interoperability issue.
This commit is a merge of the two patches indicated below.
ASTERISK-23897 #close
Reported by: Pavel Troller
Patches:
pri-4.diff (license #6302) patch uploaded by Pavel Troller
jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/3633/
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Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Removed some incorrect newlines on ast_http_error() messages in
manager.c.
* Removed an incorrect newline in res_ari_channels.c.
Addendum to ASTERISK-23552
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Persistent HTTP connection support is needed due to the increased usage of
the Asterisk core HTTP transport and the frequency at which REST API calls
are going to be issued.
* Add http.conf session_keep_alive option to enable persistent
connections.
* Parse and discard optional chunked body extension information and
trailing request headers.
* Increased the maximum application/json and
application/x-www-form-urlencoded body size allowed to 4k. The previous
1k was kind of small.
* Removed a couple inlined versions of ast_http_manid_from_vars() by
calling the function. manager.c:generic_http_callback() and
res_http_post.c:http_post_callback()
* Add missing va_end() in ast_ari_response_error().
* Eliminated unnecessary RAII_VAR() use in http.c:auth_create().
ASTERISK-23552 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/3691/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The variables body parameter under the originate and originate with id
operations of the channel resource showed invalid JSON in its description.
The variables body parameter under the userEvent operation of the event
resource made no mention that the custom key/value pairs should be wrapped
in a variables key in order to be added to the custom user event.
ASTERISK-23975 #close
Review: https://reviewboard.asterisk.org/r/3692/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A loop in ast_careful_fwrite exists that will continually attempt to write to
a file stream, even in the presence of EAGAIN/EINTR errors. However, if a
connection that uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
call to fflush may return EAGAIN/EINTER along with EOF. A subsequent call to
fflush will return EOF but not clear errno, resulting in an infinite loop.
This patch clears errno after it is detected and handled the loop, such that
any subsequent call to fflush will not get erroneously stuck.
Review: https://reviewboard.asterisk.org/r/3704
#ASTERISK-23984 #close
Reported by: Steve Davies
patches:
fflush_loop_fix uploaded by one47 (License 5012)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.
This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide
a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation
completes. Configuration options to chan_sip and chan_pjsip have also been added to
allow behavior to be tweaked (such as forcing the AVP type media transports in SDP).
ASTERISK-22961 #close
Reported by: Jay Jideliov
Review: https://reviewboard.asterisk.org/r/3679/
Review: https://reviewboard.asterisk.org/r/3686/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is essentially a backport of a small portion of r397526 from
ASTERISK-21981. In that patch, pass through support and format attribute
negotiation was added for Opus. Part of that included being more tolerant to
whitespace in the fmtp line of an SDP; that part of the patch is being
applied here.
As the author of the backport pointed out, in SDP, the fmtp line is allowed to
include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001)
includes an example for this. This was not removed in the updated RFC 4867 in
2007.
Review: https://reviewboard.asterisk.org/r/3658
#ASTERISK-23916 #close
Reported by: Alexander Traud
patches:
sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In v12+ the type values from the table are only used by the CEL unit
tests. Since the unit tests were only comparing a generated expected
event with a real event to see if the ie contents matched and using the
same table IE_PLTYPE values to read the event contents, the type
mismatches were not detected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A number of various PJSIP AMI actions were failing to parse out and place the
ActionID into their responses. This patch updates the various PJSIP actions
such that the passed in ActionID is emitted on any event list complete events,
as well as any intermediate events created as a result of the action.
ASTERISK-23947 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3675/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This will drive the test on review r3419. Note that the patch for this was
done by Ben Ford, although it was slightly modified for this commit.
ASTERISK-23562
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a client takes a long time to process information received from Asterisk,
a write operation using fwrite may fail to write all information. This causes
the underlying file stream to be in an unknown state, such that the socket
must be disconnected. Unfortunately, there are two problems with this in
Asterisk's existing websocket code:
1. Periodically, during the read loop, Asterisk must write to the connected
websocket to respond to pings. As such, Asterisk maintains a reference to
the session during the loop. When ast_http_websocket_write fails, it may
cause the session to decrement its ref count, but this in and of itself
does not break the read loop. The read loop's write, on the other hand,
does not break the loop if it fails. This causes the socket to get in a
'stuck' state, preventing the client from reconnecting to the server.
2. More importantly, however, is that the fwrite in ast_http_websocket_write
fails with a large volume of data when the client takes awhile to process
the information. When it does fail, it fails writing only a portion of
the bytes. With some debugging, it was shown that this was failing in a
similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite
with a long enough timeout solved the problem.
Note that this version of the patch, unlike r417310 in Asterisk 11, exposes
configuration options beyond just chan_sip's sip.conf. Configuration options
to configure the write timeout have also been added to pjsip.conf and ari.conf.
#ASTERISK-23917 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3624/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Almost every reference operation against container node's uses
__ao2_alloc or __ao2_ref, thereby preventing ref logging for
the nodes. One node reference is released with ao2_t_ref, causing
refcounter.py to falsely report skews and leaks for many nodes.
ASTERISK-23922 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3670/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Appending the ;2 to the user supplied ;1 uniqueid to create the ;2 version
if the user did not also supply an extra uniqueid for the ;2 channel
resulted in allocating a buffer that was one byte too small.
* Fix off by one error in ast_unreal_new_channels() when generating the ;2
uniqueid from the user suppled ;1 version.
* Pulled some long assignment lines from if tests to improve line break
readability in ast_unreal_new_channels().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@417119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This macro replaces one object reference with another cleaning up the original.
param dst Pointer to the object that will be cleaned up.
param src Pointer to the object replacing it.
src's ref count is bumped if it's non-NULL.
dst's ref count is decremented if it's non-NULL.
src is assigned to dst,
This patch was reviewed on IRC by coreyfarrell and mjordan.
Tested by: George Joseph
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_ext_tool_check.m4 isn't handling cases where a path to a package is
provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config
tool (E.G. mysql_config) and the package has its own subdirectories in include
or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql
but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib.
libxml2 has the same problem with its includes. They're in
${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include.
Both cause configure to fail and there are others in the same boat.
The problem is caused by logic in ast_ext_tool_check that overrides the result
of the config tool's --cflags and --libs options if package_DIR is set.
This patch prepends package_DIR (if specified) to the -L and -I results from
the package's config tool instead of overriding them.
A regenerated ./configure and include/asterisk/autoconfig.h.in are included
but can be regenerated by running ./bootstrap.sh at any time.
Tested by: George Joseph
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3550/
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Merged revisions 416930 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_ext_tool_check.m4 isn't handling cases where a path to a package is
provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config
tool (E.G. mysql_config) and the package has its own subdirectories in include
or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql
but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib.
libxml2 has the same problem with its includes. They're in
${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include.
Both cause configure to fail and there are others in the same boat.
The problem is caused by logic in ast_ext_tool_check that overrides the result
of the config tool's --cflags and --libs options if package_DIR is set.
This patch prepends package_DIR (if specified) to the -L and -I results from
the package's config tool instead of overriding them.
Tested by: George Joseph
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3550/
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Merged revisions 416870 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Move some implementation specific code from astobj2_container.c into
astobj2_hash.c and astobj2_rbtree.c. This completely removes the need for
astobj2_container to switch on RTTI and it no longer has any knowledge of
the implementation details.
Also adds AO2_DEBUG as a new compile option in menuselect which controls
astobj2 debugging independently of AST_DEVMODE and REF_DEBUG.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3593/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added ast_sockaddr_cidr_bits() to count the 1 bits in an ast_sockaddr.
* Added ast_ha_join_cidr() which uses ast_sockaddr_cidr_bits() for the netmask
instead of ast_sockaddr_stringify_addr.
* Changed res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() instead
of ast_ha_join() for the CLI output.
This is a CLI change only. AMI was not affected.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be incorrectly loaded
before pbx_config. pbx_config was therefore blowing away contexts that were
created by pbx_lua. With AST_MODFLAG_DEFAULT the load order is now correct
and contexs are being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not
needed anyway since no other modules needed its global symbols that early.
ASTERISK-23818 #close
Reported by: Dennis Guse
Tested by: Dennis Guse
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3629/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The sample extensions.lua was causing pbx_lua to fail to load when parsing
'app.goto("default", "s", 1)' because in Lua 5.2, 'goto' is now a reserved
word. This patch adds guidance to extensions.lua.sample and changed
'app.goto("default", "s", 1)' to 'app.['goto']("default", "s", 1)'.
ASTERISK-23844 #close
Reported by: rnewton
Tested by: gtjoseph
Review: https://reviewboard.asterisk.org/r/3627/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r416211, the publishing of variable changes was modified such that a
cached channel snapshot was used if manager variables were not requested
with each AMI event. This was done to reduce the amount of channel snapshots
created.
However, an assumption was made that generating a channel snapshot and
publishing the snapshot to the channel topic was sufficient to ensure that
the cache would be updated; this is not the case. The channel snapshot type
must be used to force a snapshot update.
This patch updates the publication of channel variables such that the cache
is updated prior to publication of the channel variable message if manager
variables are in use. This ensures that all AMI events receive the variable
update when they are supposed to.
Note that this issue was caught by the Asterisk Test Suite (go go testing)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@416557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
