https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines
Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT
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: http://svn.digium.com/svn/asterisk/trunk@323372 f38db490-d61c-443f-a65b-d21fe96a405b
Current behavior in ConfBridge menu definitions is that first definition takes
precedence, even in templated situations. This change allows inheritance and
overriding to work as expected so that the last definition takes precedence.
(closes ASTERISK-17986)
Review: https://reviewboard.asterisk.org/r/1267/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323272 f38db490-d61c-443f-a65b-d21fe96a405b
This adds the playing_moh flag to the conference_bridge_user struct that
signifies when MOH should be playing so code doesn't have to guess whether
MOH is playing.
This change also adds the necessary checking to ensure that MOH continues
playing for a single user in a conference after the join sound is played when
configured to do so.
(closes ASTERISK-17988)
Review: https://reviewboard.asterisk.org/r/1263/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323107 f38db490-d61c-443f-a65b-d21fe96a405b
Bridge and user profiles are not checked for existence before use. The lack
of a fully formed bridge profile can cause a segfault when sounds are accessed.
This change ensures that bridge and user profiles exist prior to usage
attempts.
Review: https://reviewboard.asterisk.org/r/1264/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323106 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines
Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop.
(closes issue ASTERISK-17798)
tested by mnicholson
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323041 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) | 11 lines
Avoid a DB1 infinite loop bug
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: http://svn.digium.com/svn/asterisk/trunk@322982 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines
Remove potential deadlock in call pickup race.
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: http://svn.digium.com/svn/asterisk/trunk@322750 f38db490-d61c-443f-a65b-d21fe96a405b
........
r322585 | jrose | 2011-06-09 09:06:42 -0500 (Thu, 09 Jun 2011) | 11 lines
Adds ast_escape_encoded utility to properly handle escaping of quoted field before uri.
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: http://svn.digium.com/svn/asterisk/trunk@322586 f38db490-d61c-443f-a65b-d21fe96a405b
Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER.
Initial value must be the time to autoanswer in ms, then optionally :BEEP
to play a tone when answered and :MUTE to mute the mic when answering.
eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and
play a beep. just 3000 would answer afer 3 secs of ringing with no
beep and full two way audio.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322544 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines
Ring all queue with more than 255 agents will cause crash.
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: http://svn.digium.com/svn/asterisk/trunk@322485 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines
SRV lookup attempted for SIP peers listed as an IP address.
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: http://svn.digium.com/svn/asterisk/trunk@322426 f38db490-d61c-443f-a65b-d21fe96a405b
The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything.
Review: https://reviewboard.asterisk.org/r/1256/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322381 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines
Make handle_request_publish do dialog expiration and destruction.
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: http://svn.digium.com/svn/asterisk/trunk@322323 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) | 18 lines
Asterisk crash when unloading cdr_radius/cel_radius.
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: http://svn.digium.com/svn/asterisk/trunk@321927 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines
Be more explicit for CCSS generic device state event subscription.
Make CCSS generic device state event subscription specify the
AST_EVENT_IE_STATE ie exists to be safe.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321925 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines
Event subscription fixes.
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: http://svn.digium.com/svn/asterisk/trunk@321872 f38db490-d61c-443f-a65b-d21fe96a405b
........
r321753 | russell | 2011-06-03 13:32:45 -0500 (Fri, 03 Jun 2011) | 2 lines
Backport an astobj2 unit test so that it runs on 1.8 as well.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321754 f38db490-d61c-443f-a65b-d21fe96a405b
Don't send all messages to 's'. Get the destination from the request URI.
(Found using automated test cases).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321617 f38db490-d61c-443f-a65b-d21fe96a405b
Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321546 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
Fix double alerting, add forced alerting before answer
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: http://svn.digium.com/svn/asterisk/trunk@321529 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) | 12 lines
Crash when using hagi and no servers are available.
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: http://svn.digium.com/svn/asterisk/trunk@321393 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines
The app_privacy args have undocumented "options" position, interferes with "context" position.
* 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: http://svn.digium.com/svn/asterisk/trunk@321331 f38db490-d61c-443f-a65b-d21fe96a405b
jrose
Patrick McHardy