This fixes address lookup for incoming calls without a peer definition.
The address family was unset instead of being set to AST_AF_UNSPEC
which was causing lookup failures on "127.0.0.1". This is one of the
causes of the current failure of the app_page integration test.
Review: https://reviewboard.asterisk.org/r/2933/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows a user of ARI to know what channel it has originated and also follow any
progress. If a Stasis application is provided it will be automatically subscribed to the
originated channel immediately.
(closes issue ASTERISK-22485)
Reported by: David Lee
Review: https://reviewboard.asterisk.org/r/2910/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A common idiom in Asterisk is to due something like:
for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) {
...do stuff...
}
This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge. With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.
* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.
* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper(). The call to
ast_translator_best_choice() got them backwards.
* Updated some callers of ast_channel_make_compatible() and the function
documentation. There is actually a difference between the two channels
passed in.
* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible(). The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.
(closes issue ASTERISK-22542)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2915/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When trying to determine if a peer is behind NAT, we should not be using the
ports when comparing addresses.
This patch removes the port from being checked and just useds the addresses
now.
(closes issue ASTERISK-22729)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-remove-using-port-for-nat-check.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2927/
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In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells
the devstate system to not cache states for non-real devices. However,
when optimizing away channels (ast_do_masquerade), that flag wasn't
copied.
In my case, using Local devices as queue members created a situation
where the endpoint was considered in use, but the state change of the
device being available again was ignored (not cached). The endpoint
channel was optimized into the (previously) Local channel, but kept
the do-not-cache flag. The end result being that the queue member
apparently stayed in use forever.
(closes issue ASTERISK-22718)
Reported by: Walter Doekes
Review: https://reviewboard.asterisk.org/r/2925/
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A condition was added in a commit to fix ASTERISK-21374, that, if the
SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT
flag to the dialog. This condition should not have been there since it assumed
that if Asterisk is in an environment where NAT is involved, that the auto_* nat
settings or force_rport setting would be on in the global settings. If the nat
setting in the global setting is set to 'nat=no' and then turned on for peers
(which is not quite the recommended way, although it is allowed) this flag is
never copied to the dialog resulting in problems like, REGISTER replies going
to the wrong port.
This patch removes this conditional check and will now always use the peer's
flag which by this point in the code the checks on whether the peer is behind
NAT or not (if using auto_force_rport) have already been run.
(closes issue ASTERISK-22236)
Reported by: Filip Frank
Tested by: Michael L. Young
Patches:
asterisk-2236-always-set-rport.diff uploaded
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2919/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.
(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
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This makes it clear that the ARI API calls for listing channels and
bridges will list all channels or bridges in the system and not just
those that are in or are controlled by a Stasis application.
(closes issue ASTERISK-22635)
Reported by: Kevin Harwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.
(closes issue ASTERISK-22627)
Reported by: Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@401018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a 200 OK for an initial INVITE is received, we were doing
the right thing by ACKing and sending an immediate BYE. However,
we also were doing the wrong thing and queuing an answer frame,
thus causing the call to be answered. This would cause the call
to be hung up by the channel thread, thus resulting in a second
BYE being sent out.
In this fix, I also have set the hangupcause to be correct since
the initial BYE being sent by Asterisk had an unknown hangup
cause. I have changed to using "Bearer capabilty not available"
since the call was hung up due to an SDP offer/answer error.
(closes issue ASTERISK-22621)
reported by Kinsey Moore
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Because I added a wiki_description to models and not their properties, the
rendered wiki page had the model description instead of the property
descriptions, which looks very silly indeed.
(closes issue ASTERISK-22705)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the server didn't exist that we are trying to register to the log message
would say that a 408 was received from that server when in reality one wasn't.
Added log messages stating no response was received if the response does not
exist.
(closes issue ASTERISK-22554)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2893/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The INVITE session state callback wrongly assumes that a session will always exist, but
when rapidly terminating the session this assumption goes out the window. As all handler
code for the INVITE session state callback requires the session it will now just exit
immediately if no session exists.
(closes issue ASTERISK-22668)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When generating the list of authentication credentials to pass to
PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not
always NULL-terminated. This sometimes resulted in incorrect text for
the realm and a failure to match the realm for authentication purposes
which was causing the outbound nominal auth pjsip basic call test to
bounce. This now uses the pj_str_t that contains the realm instead of
generating a new one. Thanks to John Bigelow for helping to narrow this
down.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The crash is caused by a race condition when switching between native RTP
and softmix bridging technologies. In this situation, the bridging
technology is switched from native RTP to softmix, and then back to native
RTP fast enough that the softmix private data gets destroyed before the
softmix mixing thread gets started.
Thanks to Kinsey Moore for the crash analysis.
* Fix race condition when starting the softmix mixing thread and switching
to another bridge technology.
(closes issue ASTERISK-22678)
Reported by: John Bigelow
Patches:
jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett
Tested by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change fixes two issues when setting an outbound proxy:
1. The outbound proxy URI was not parsed and validated during configuration.
2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would
occur because the usage count on the dialog was not decremented.
The documentation has also been updated to specify that a full URI must be specified for
the outbound proxy.
(closes issue ASTERISK-22672)
Reported by: Antti Yrjola
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Using 'lu' will produce a compiler warning for some versions of gcc and on some
architectures. 'z' should be portable as a format specifier for size_t.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support to the PJSIP stack in Asterisk for SIP header
manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader.
For PJSIP_HEADER, an incoming supplemental session callback is registered that
takes the pjsip_hdrs from the incoming session and stores them in a linked
list in the session datastore. Calls to PJSIP_HEADER traverse over the list
and return the nth matching header where 'n' is the 'number' argument to the
function.
When adding a header, the first call creates a datastore and linked list and
adds the datastore to the session. The header is then created as a pjsip_hdr
and added to the list. An outgoing supplemental session callback then
traverses the list and adds the headers to the outgoing pjsip_msg.
When removing a header, the list created with PJSIP_HEADER(add,...) is
traversed and all matching entries are removed.
(closes issue ASTERISK-22498)
Reported by: George Joseph
patch:
res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When running configure, libiodbc2 development headers will fulfill the
requirement for ODBC development headers, but will not function
properly. This adds a warning when libiodbc2 development headers are
detected instead of unixodbc development headers.
(closes issue ASTERISK-22459)
Reported by: Patrick Maille
Tested by: Walter Doekes
Patches:
issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pjsip_strerror is only aware of PJSIP-specific error
codes. pj_strerror() is aware of all PJProject error
codes and OS-specific error codes.
This specifically fixes an oft-seen error in transport
configuration code where EADDRINUSE would result in
"Unknown PJSIP error 120098" instead of a useful
message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AST_CONFIG dialplan function defined in func_config.c allocates its
config file list entries using ast_malloc. List entry allocations
destined for use with Asterisk's linked list API must be ast_calloc()d
or otherwise initialized so that list pointers are set to NULL. These
uses of ast_malloc have been replaced by ast_calloc to prevent
dereferencing of uninitialized pointer values when traversing the list.
(closes issue ASTERISK-22483)
Reported by: Brian Scott
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.
(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If you run Asterisk in the background and then connect to
it through a separate console, the thread that runs CLI commands
is not registered with PJLIB. Thus PJLIB does not like it when
you attempt to send OPTIONS requests from that thread. So now
we push the task into the threadpool, which we know to be registered
with PJLIB.
Thanks to Antti Yrjola for reporting this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The https://reviewboard.asterisk.org/r/2888/ review changes manager to not
subscribe to stasis when it is disabled for performance reasons. When
manager is disabled app_queue and res_agi decline to load and fail to
clean up what they have already allocated.
* Made app_queue and res_agi clean up allocated resources when they
decline to load.
* Made app_queue and res_agi use their own subscriptions to the stasis
topics instead of borrowing manager's message router structure
inappropriately.
(closes issue ASTERISK-22604)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/2902/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
