It turned out that this was causing NAT=Yes to always use rport when present which was
against 1.6.2 behavior and the check itself was redundant since the only way this
segment of code could be reached was if RPORT_PRESENT was already evaluated as true
earlier.
(closes issue ASTERISK-17789)
Reported by: byronclark
Patches:
use_sip_nat_force_rport.patch uploaded by byronclark (license 1200)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note that bindaddr will only enable UDP instead of both UDP and TCP which is
what I would expect for backwards compatibility with systems being upgraded
which only support UDP transportation.
(closes issue ASTERISK-17976)
Reported by: Sean Darcy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@323234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Explicity check the last entry in the DB and make sure that we don't iterate
past it. Since there can be no duplicates, this just makes sure that we stop
after matching the last key.
This patch also refactors the code to get away from some code duplication. A
previous patch added many astdb tests and this patch passed them.
Review: https://reviewboard.asterisk.org/r/1259/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Deadlock is possible in ast_do_pickup() when holding the target channel
lock and trying to get the chan channel lock. Also, holding the target
lock when calling ast_channel_masquerade() is not a good idea because that
routine does deadlock avoidance.
* Removed the need to hold the target lock after marking the target with a
datastore and getting the connected line data off of the target channel.
* Moved can_pickup() to ast_can_pickup() in features.c. Now all the call
pickup methods use the same basic call pickup availability check.
Review: https://reviewboard.asterisk.org/r/1234/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit backports a feature in trunk affecting initreqprep so that display name won't
be encoded improperly. Also includes unit tests for the ast_escape_quoted function.
This patch gives 1.8 a much improved outlook in countries which don't use standard
ASCII characters.
(closes issue ASTERISK-16949)
Reported by: Örn Arnarson
Review: https://reviewboard.asterisk.org/r/1235/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. Create a ring-all queue with 500 permanent agents.
2. Call it.
3. Asterisk will crash.
The watchers array in app_queue.c has a hard limit of 255. Bounds
checking is not done on this array. No sane person should put 255 people
in a ring-all queue, but we should not crash anyway.
* Added bounds checking to the watchers array.
JIRA AST-464
JIRA SWP-2903
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk attempts to SRV lookup a host name even if the host name is an IP
address. Regression introduced when IPv6 support was added.
* Restored the check in ast_dnsmgr_lookup() to see if the given host name
is an IP address. The IP address could be in either IPv4 or IPv6 formats.
(closes issue ASTERISK-17815)
Reported by: Byron Clark
Tested by: Byron Clark, Richard Mudgett
Patches:
issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621)
Review: https://reviewboard.asterisk.org/r/1240/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes handle_request_publish so that it does dialog expiration and destruction.
Without this patch the incoming PUBLISH requests will get stuck in the dialog list.
Restarting asterisk is the only way to remove them.
Personal observation on one system the server hung up while looping through the channels
rendering asterisk unusable and all sip phones unregisterd when they try reregister
more requests are added.
(closes issue #18898)
Reported by: gareth
Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot
Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915
Review: https://reviewboard.asterisk.org/r/1253
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@322322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The rc_openlog() API call is passed a string that is used by openlog() to
format log messages. The openlog() does not copy the string it just keeps
a pointer to it. When the module is unloaded, the string is gone from
memory. Depending upon module load order and if the other module then has
an error, a crash happens.
* Pass rc_openlog() a strdup'd string with the understanding that there
will be a small memory leak if the cdr_radius/cel_radius modules are
unloaded.
* Call rc_destroy() to free the rc handle memory when the module is
unloaded.
JIRA AST-483
JIRA SWP-3062
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Must commit the subscription fixes together with the integration
subscription tests. The subscription fixes cause an erroneously passing
test to fail. The new subscription tests detect errors without the
subscription fixes.
* Added missing event_names[] table entry.
* Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to
correctly detect if a subscriber exists for the proposed event.
* Made match_ie_val() and match_sub_ie_val_to_event() check the buffer
length for RAW payload types.
* Fixed error handling memory leak in ast_event_sub_activate(),
ast_event_unsubscribe(), and ast_event_queue().
* Made ast_event_new() and ast_event_check_subscriber() better protect
themselves from an invalid payload type.
* Added container lock protection between removing old cache events and
adding the new cached event in
ast_event_queue_and_cache()/event_update_cache().
* Added new event subscription tests.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix double alerting (it wasn't fixed here by issue #18542)
Add forced alerting before connect (if it wasn't before)
Try to send all packets from outgoing queue rather than one only
Call goes into clearing state when disconnect command is received
(closes issue #19361)
Reported by: vmikhelson
Patches:
issue19361-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When none of the servers returned by the SRV querey respond, asterisk
crashes. The problem is that if the loop over all the SRV entries
finishes then the srv_context has already been cleaned up.
* Make ast_srv_cleanup() check to see if the context is already cleaned
up.
(closes issue #19256)
Reported by: byronclark
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add documention for unused "options" position to match existing code.
The trunk(v1.10) version will remove the unused options position.
(closes issue #19273)
Reported by: mdavenport
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method,
This stop the clash of 2 threads trying to write audio to same channel.
In addition fixes choppy audio beep in issue 19177.
(issue #18654)
(issue #19177)
Reported by: Docent
Patches:
review1232-1.88888888 alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1232/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails.
(closes issue #19346)
Reported by: kobaz
Tested by: kobaz,JonathanRose
Review: [full review board URL with trailing slash]
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables
(closes issue #19346)
Reported by: kobaz
Patches:
netsock2.patch uploaded by kobaz (license 834)
Tested by: kobaz, Marquis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is important to always initialize ast_sockaddrs before use--even if they
are passed to ast_sockaddr_copy as the underlying storage could be bigger
than what ends up being copied--leaving part of the data unitialized.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SUBSCRIBE message used to cancel a CC request has incorrect To/From
SIP headers. They are reversed and the dialog tags are the same when they
should not be. If pedantic mode was disabled, then the cancel would have
succeeded despite the incorrect message.
* The SIP_OUTGOING flag was not set correctly for the dialog and I had to
move some CC subscribe handling code as a result.
* Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a
CC request SUBSCRIBE message comes in and the CC instance is not found,
the 404 response was duplicated.
JIRA AST-568
JIRA SWP-3493
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The addition of connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer time
shares with the connected line ID on outgoing call legs. The timing of
some AMI events/responses output the connected line ID as caller ID.
These party ID's are now separate.
* The ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were also
present.
(closes issue #18252)
Reported by: gje
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1227/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Recent crashes from zombie channels suggests that they need a safe home to
goto. When a masquerade happens, the physical part of the zombie channel
is hungup. The hangup normally sets the channel private pointer to NULL.
If someone then blindly does a callback to the channel driver, a crash is
likely because the private pointer is NULL.
The masquerade now sets the channel technology of zombie channels to the
kill channel driver.
Related to the following issues:
(issue #19116)
(issue #19310)
Review: https://reviewboard.asterisk.org/r/1224/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add ConnectedLineNum and ConnectedLineName headers to the output of the
AMI action Status. This makes it easier to find out who the channel is
connected to without having to lookup BridgedChannel or when they are
connected to an application (e.g.: VoiceMail) which has no bridged
channel.
* Bridged channels with no CallerID had "" instead of "<unknown>" output,
that might be a bug as "<unknown>" was what older versions used.
(closes issue #18158)
Reported by: gareth
Patches:
svn-292308.diff uploaded by gareth (license 208)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320650 65c4cc65-6c06-0410-ace0-fbb531ad65f3
