When we send out a CN packet (for instance, in the case of using rtpkeepalives),
we are not setting the payload code properly. Also, we are setting the marker
bit when we shouldn't be according to RFC 3389, section 4.
AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using
ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup().
11 and trunk already use the appropriate function.
* In 1.8, use ast_rtp_codecs_payload_code()
* Remove the setting of the marker bit
* Fix the debug message by incrementing the seqno after the debug message is set
in order to display the correct seqno that was sent out
(closes issue ASTERISK-21246)
Reported by: Peter Katzmann
Tested by: Peter Katzmann, Michael L. Young
Patches:
asterisk-21246-rtp-cng-payload-error_1.8_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2500/
........
Merged revisions 388111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the "ignorebusy" setting was deprecated, we added some code to allow us to
be compatible with older setups that are still using the "ignorebusy" setting
instead of "ringinuse". We set a char *variable with the column name to use,
which helps the realtime functions to use the correct column in their SQL
queries. When "persistentmembers" is enabled, we are not setting this variable
before the realtime functions were called to load members. This results in the
variable being NULL and therefore causing a segfault when loading members during
the module's process of loading.
The solution was to move the code that sets that variable to be before these
realtime functions are called during the loading of the module.
(closes issue ASTERISK-21738)
Reported by: JoshE
Tested by: JoshE
Patches:
asterisk-21738-rt-ringinuse-field-not-set.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2499/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC6665 4.2.2: ... after a failed State NOTIFY transaction remove the subscription
The problem is that the State Notify requests rely on the 200OK reponse for pacing control
and to not confuse the notify susbsystem.
The issue is, the pendinginvite isn't cleared if a response isn't received,
thus further notify's are never sent.
The solution, follow RFC 6665 4.2.2's 'SHOULD' and remove the subscription after failure.
(closes issue ASTERISK-21677)
Reported by: Dan Martens
Tested by: Dan Martens, David Brillert, alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2475/
........
Merged revisions 387875 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reload support was originally not included for SLA. It was added later,
but in a fairly non-traditional way. It basically sets a flag
indicating that a reload is pending, and then waits for a time where it
thinks everything SLA related is idle and unused, and *then* executes
the reload. It does this because the reload process is destructive. It
starts by throwing everything away and starting over.
There are a number of problems with this approach. One of them is that
the check to see if anything in use was incomplete. This patch makes it
more complete and thus less likely for a crash to occur during reload
processing. However, this approach still has problems so some much more
significant reworking of this code will need to come in as a next step.
Patch credit and testing by CoreDial, LLC.
........
Merged revisions 387688 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Alec's patch that added the Asterisk version to 'core show locks' angered the
items in utils, as they exist somewhat outside of the Asterisk build system.
Some day, this Makefile should get nuked from high orbit, but for now, include
version.c in its list of stuff to pile in.
........
Merged revisions 387421 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 4028 Section 10
if the side not performing refreshes does not receive a
session refresh request before the session expiration, it SHOULD send
a BYE to terminate the session, slightly before the session
expiration. The minimum of 32 seconds and one third of the session
interval is RECOMMENDED.
Prior to this asterisk would refresh at 1/2 the Session-Expires interval,
or if the remote device was the refresher, asterisk would timeout at interval end.
Now, when not refresher, timeout as per RFC noted above.
(closes issue ASTERISK-21742)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2488/
........
Merged revisions 387344 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 4028 Section 7.2
"UACs MUST be prepared to receive a Session-Expires header field in a
response, even if none were present in the request."
What changed
After ASTERISK-20787, inbound calls to asterisk with no Session-Expires in the INVITE are now are offered
a Session-Expires (1800 asterisk default) in the response, with asterisk as the refresher.
Symptom:
After 900 seconds (asterisk default refresher period 1800), asterisk RE-INVITEs the device, the device
may respond with a much lower Session-Expires (180 in our case) value that it is now using.
Asterisk ignores this response, as it's deemed both an INBOUND CALL, and a RE-INVITE.
After 180 seconds the device times out and sends BYE (hangs up), asterisk is still working with the
refresher period of 1800 as it ignored the 'Session Expires: 180' in the previous 200OK response.
Fix:
handle_response_invite() when 200OK, remove check for outbound and reinvite.
(closes issue ASTERISK-21664)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2463/
........
Merged revisions 387312 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In certain situations, when the RTP engine goes to send a DTMF end digit
it may be in a situation where the remote address is no longer available,
or the digit that was supposed to be sent is invalid. In such cases, we
need to clear the RTP counters appropriately. Otherwise, when the RTP
source is set again, we'll continue to think that we're in the middle of
sending a DTMF digit, which can confuse the remote party (signficantly).
(closes issue ASTERISK-21522)
Reported by: Corey Farrell
patches:
rtp_dtmf_process_end.patch uploaded by Corey Farrell (License 5909)
........
Merged revisions 387213 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When you have lots of SIP peers (according to the issue reporter, around 3500),
the 'sip show peers' CLI command or AMI action can crash due to a poorly placed
string duplication that occurs on the stack. This patch refactors the command
to not allocate the string on the stack, and handles the formatting of a single
peer in a separate function call.
(closes issue ASTERISK-21466)
Reported by: Guillaume Knispel
patches:
fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Way back when in the dark days of Asterisk 1.8.9, blind transferring a call
in a context that included the 'h' extension would inadvertently execute the
hangup code logic on the transferred channel. This was a "bad thing". The fix
was to properly check for the softhangup flags on the channel and only execute
the 'h' extension logic (and, in later versions, hangup handler logic) if the
channel was well and truly dead (Jim).
Unfortunately, CDRs are fickle. Setting the softhangup flag when we detected
that the channel was leaving the bridge (but not to die) caused some crucial
snippet of CDR code, lying in ambush in the middle of the bridging code, to
not get executed. This had the effect of blowing away one of the CDRs that is
typically created during a blind transfer.
While we live and die by the adage "don't touch CDRs in release branches", this
was our bad. The attached patch restores the CDR behavior, and still manages to
not run the 'h' extension during a blind transfer (at least not when it's
supposed to).
Thanks to Steve Davies for diagnosing this and providing a fix.
Review: https://reviewboard.asterisk.org/r/2476
(closes issue ASTERISK-21394)
Reported by: Ishfaq Malik
Tested by: Ishfaq Malik, mjordan
patches:
fix_missing_blindXfer_cdr2 uploaded by one47 (License 5012)
........
Merged revisions 387036 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@387038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a system allows for its usage, Asterisk will use glob to help parse
Asterisk .conf files. The config file loading routine was leaking the memory
allocated by the glob() routine when the config file was in an unmodified
or invalid state.
This patch properly calls globfree in those off nominal paths.
(closes issue ASTERISK-21412)
Reported by: Corey Farrell
patches:
config_glob_leak.patch uploaded by Corey Farrell (license 5909)
........
Merged revisions 386672 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch cleans up two things features:
* It properly unregisters the CLI commands that features registered
* It cancels and performs a pthread_join on the created parking thread. This
not only properly joins a non-detached thread, but also prevents disposing
of the parking lots prior to the parking thread completely exiting.
(closes issue ASTERISK-21407)
Reported by: Corey Farrell
patches:
features_shutdown-r2.patch uploaded by Corey Farrell (License 5909)
........
Merged revisions 386641 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The two party bridging loops were changing the bridge peer pointers
without the channel locks held. Thus when ast_channel_massquerade()
tested and used the pointer there is a small window of opportunity for the
pointers to become NULL even though the masquerade code has the channels
locked.
(closes issue ASTERISK-21356)
Reported by: William luke
Patches:
jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: William luke
........
Merged revisions 386256 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were several reports of deadlock when using
res_timing_pthread. Backtraces indicated that one thread was blocked
waiting for the write to the pipe to complete and this thread held
the container lock for the timers. Therefore any thread that wanted
to create a new timer or read an existing timer would block waiting
for either the timer lock or the container lock and deadlock ensued.
This patch changes the way the pipe is used to eliminate this source
of deadlocks:
1) The pipe is placed in non-blocking mode so that it would never
block even if the following changes someone fail...
2) Instead of writing bytes into the pipe for each "tick" that's
fired the pipe now has two states--signaled and unsignaled. If
signaled, the pipe is hot and any pollers of the read side
filedescriptor will be woken up. If unsigned the pipe is idle. This
eliminates even the chance of filling up the pipe and reduces the
potential overhead of calling unnecessary writes.
3) Since we're tracking the signaled / unsignaled state, we can
eliminate the exta poll system call for every firing because we know
that there is data to be read.
(closes issue ASTERISK-21389)
Reported by: Matt Jordan
Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis
patches:
0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417)
(closes issue ASTERISK-19754)
Reported by: Nikola Ciprich
(closes issue ASTERISK-20577)
Reported by: Kien Kennedy
(closes issue ASTERISK-17436)
Reported by: Henry Fernandes
(closes issue ASTERISK-17467)
Reported by: isrl
(closes issue ASTERISK-17458)
Reported by: isrl
Review: https://reviewboard.asterisk.org/r/2441/
........
Merged revisions 386109 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In messages.c, there are several places in the code where we create a
tmp_tech_holder and pass that into an ao2_find call. Unfortunately, we
weren't initializing the rwlock on the tmp_tech_holder, which the hash
function was locking. It's apparently harmless, but still not the best
code.
This patch extracts all that copy/pasted code into two functions,
msg_find_by_tech and msg_find_by_tech_name, which properly initialize
and destroy the rwlock on the tmp_tech_holder.
Review: https://reviewboard.asterisk.org/r/2454/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_xmpp was not adding AST_EVENT_IE_CACHABLE to the event as each message came in,
then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event,
so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE.
(issue ASTERISK-20175)
(closes issue ASTERISK-21429)
(closes issue ASTERISK-21069)
(closes issue ASTERISK-21164)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2452/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res_jabber/res_xmpp were not adding AST_EVENT_IE_CACHABLE to the event as each message came in,
then devstate_change_collector_cb() was unable to find AST_EVENT_IE_CACHABLE in the event,
so defaulted incorrectly to AST_DEVSTATE_NOT_CACHABLE.
(issue ASTERISK-20175)
(closes issue ASTERISK-21429)
(closes issue ASTERISK-21069)
(closes issue ASTERISK-21164)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2452/
........
Merged revisions 385916 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.
(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
rtp-timestamp.patch uploaded by pbertera (License 5943)
........
Merged revisions 385636 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, a read error in snd_pcm_readi would still be treated as a
nominal result when constructing a voice frame from the expected data. Since
the value returned is negative, as opposed to the number of samples read,
this could result in a crash. With this patch, we now return a null frame
when a read error is detected.
Note that the patch on ASTERISK-21329 was modified slightly for this commit,
in that we bail immediately on detecting the read error, rather than bypassing
the construction of the voice frame.
(closes issue ASTERISK-21329)
Reported by: Keiichiro Kawasaki
patches:
chan_alsa.diff uploaded by kawasaki (License 6489)
........
Merged revisions 385633 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When app_queue is unloaded, some manager commands are not being unregistered
which result in a segfault. This patch corrects this.
(closes issue ASTERISK-21397)
Reported by: Peter Katzmann, Corey Farrell
Tested by: Corey Farrell
Patches:
asterisk-21397-missing-unreg-manager-cmd_1.8.diff
Michael L. Young (license 5026)
asterisk-21397-missing-unreg-manager-cmd_11.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2444/
........
Merged revisions 385593 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original report was that app_voicemail would crash. This was caused by
ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being
performed for that return status. After adding the initial patch to fix this
issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered.
During review, Walter Doekes (wdoekes) suggested adding a helper function in
order to determine if we had a valid configuration or not.
This patch does the following:
* Creates a helper function to check if the configuration is valid
* Adds calls to the new helper function where appropiate
* Fixes memory leaks where the code returned without running
ast_config_destroy() on the configuration that was loaded
(closes issue ASTERISK-21302)
Reported by: Jaco Kroon
Tested by: Jaco Kroon, Michael L. Young
Patches:
asterisk-11.3.0-app_voicemail-ast_config-fixes.patch
Jaco Kroon (license 5671)
asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2443/
........
Merged revisions 385551 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we reload Asterisk or chan_sip, the flags force_rport and comedia that are
turned on and off when using the auto_force_rport and auto_comedia nat settings
go back to the default setting off. These flags are turned on when needed or
off when not needed at the time that a peer registers, re-registers or initiates
a call. This would apply even when only the default global setting
"nat=auto_force_rport" is being used, which in this case would only affect the
force_rport flag.
Everything is good except for the following: The nat setting is set to
auto_force_rport and auto_comedia. We reload Asterisk and the peer's
registration has not expired. We load in the settings for the peer which turns
force_rport and comedia back to off. Since the peer has not re-registered or
placed a call yet, those flags remain off. We then initiate a call to the peer
from the PBX. The force_rport and comedia flags stay off. If NAT is involved,
we end up with one-way audio since we never checked to see if the peer is behind
NAT or not.
This patch does the following:
* Moves the checking of whether a peer is behind NAT into its own function
* Create a function to set the peer's NAT flags if they are using the auto_* NAT
settings
* Adds calls in sip_request_call() to these new functions in order to setup the
dialog according to the peer's settings
(closes issue ASTERISK-21374)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2421/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On startup, it's possible for a frame to arrive before the processing threads were ready.
In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive
before we are at ast_cond_wait, the signal will be ignored.
The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued.
Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled
after each thread is started.
(issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2427/
........
Merged revisions 385402 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pjproject is what actually requires libuuid.
(closes issue ASTERISK-21125)
reported by Private Name
(Ed. note: Really? Private Name? I am rolling my eyes so hard right now.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk:
invalid pointer errors) can occur as the memory is being allocated with
Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the
LDAP library's wrappers.
This patch uses the LDAP library's wrappers where appropriate, so that
compiling with MALLOC_DEBUG doesn't cause more problems than it solves.
Note that the patch listed below was modified slightly for this commit
to account for some additional memory allocation/deallocations.
(closes issue ASTERISK-17386)
Reported by: John Covert
Tested by: Andrew Latham
patches:
issue18789-1.8-r316873.patch uploaded by seanbright (License 5060)
........
Merged revisions 385190 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a BYE request is processed in chan_sip, the current SIP dialog is detached
from its associated Asterisk channel structure. The tech_pvt pointer in the
channel object is set to NULL, and the dialog persists for an RFC mandated
period of time to handle re-transmits.
While this process occurs, the channel is locked (which is good).
Unfortunately, operations that are initiated externally have no way of knowing
that the channel they've just obtained (which is still valid) and that they are
attempting to lock is about to have its tech_pvt pointer removed. By the time
they obtain the channel lock and call the channel technology callback, the
tech_pvt is NULL.
This patch adds a few checks to some channel callbacks that make sure the
tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit
callbacks, which would crash if AMI initiated a DTMF on the channel at the
same time as a BYE was received from the UA. This patch also adds checks on
sip_transfer (as AMI can also cause a callback into this function), as well
as sip_indicate (as lots of things can queue an indication onto a channel).
Review: https://reviewboard.asterisk.org/r/2434/
(closes issue ASTERISK-20225)
Reported by: Jeff Hoppe
........
Merged revisions 385170 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends.
........
Merged revisions 385047 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
Fix For Not Overriding The Default Settings In chan_sip
The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting. Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.
This patch works similar to what occurs in build_peer(). We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.
In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.
(closes issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2386/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting. Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.
This patch works similar to what occurs in build_peer(). We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.
In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.
This issue is also present in Asterisk 1.8 and a separate patch will be applied
to it.
(issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
asterisk-21225-handle-options-default-prob_v4.diff
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2385/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
Backport Appropiate NAT Setting Cleanup
In ASTERISK-20904, the focus was around the changes to NAT that took place in
Asterisk 11. Since the report stated that 1.8 was fine, we didn't take a look
at 1.8 at the time.
While working on ASTERISK-21225, I could see that 1.8 would benefit from having
some of those changes applied to it.
This patch does the following:
* The important part of this patch is that it sets the peer's flags earlier in
build_peer so that the code properly uses the peer's flags based on the peer's
configuration.
* constify req parameter in check_via()
* update realtime schemas under the contrib directory to handle properly the NAT
settings available in 1.8 as well as to handle the changes made in 11 to make
upgrading easier when installing newer versions of Asterisk
(closes issue ASTERISK-21243)
Reported by: Michael L. Young
Patches:
asterisk-20904-changes_for_1.8.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2422/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.
Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio. However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.
ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.
(closes issue ASTERISK-21151)
Reported by: Gianluca Merlo
Tested by: rmudgett
........
Merged revisions 384685 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
