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/
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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/
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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/
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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
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)
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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/
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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
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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
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
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When authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The
information is disclosed when:
* A "407 Proxy Authentication Required" response is sent instead of a
"401 Unauthorized" response
* The presence or absence of additional tags occurs at the end of "403
Forbidden" (such as "(Bad Auth)")
* A "401 Unauthorized" response is sent instead of "403 Forbidden" response
after a retransmission
* Retransmission are sent when a matching peer did not exist, but not when a
matching peer did exist.
This patch resolves these various vectors by ensuring that the responses sent
in all scenarios is the same, regardless of the presence of a matching peer.
This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
the testing and the solution to this problem was done by Walter as well - a
huge thanks to his tireless efforts in finding all the ways in which this
setting didn't work, providing automated tests, and working with Kinsey on
getting this fixed.
(closes issue ASTERISK-21013)
Reported by: wdoekes
Tested by: wdoekes, kmoore
patches:
AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674)
AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674)
AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@384003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r373424, several reentrancy problems in chan_sip were addressed. As a
result, the SIP channel driver is now properly locking the channel driver
private information in certain operations that it wasn't previously. This
exposed two latent problems either in register_verify or by functions called
by register_verify. This includes:
* Holding the private lock while calling sip_send_mwi_to_peer. This can create
a new sip_pvt via sip_alloc, which will obtain the channel container lock.
This is a locking inversion, as any channel related lock must be obtained
prior to obtaining the SIP channel technology private lock.
Note that this issue was already fixed in Asterisk 11.
* Holding the private lock while calling sip_poke_peer. In the same vein as
sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
the same locking inversion.
Note that this locking inversion typically occured when CLI commands were run
while a SIP REGISTER request was being processed, as many CLI commands (such
as 'sip show channels', 'core show channels', etc.) have to obtain the channel
container lock.
(issue ASTERISK-21068)
Reported by: Nicolas Bouliane
(issue ASTERISK-20550)
Reported by: David Brillert
(issue ASTERISK-21314)
Reported by: Badalian Vyacheslav
(issue ASTERISK-21296)
Reported by: Gabriel Birke
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The CALLEDTON channel variable is set for incoming ISDN calls to the lower
7 bits of the Q.931 type-of-number/numbering-plan octet. The
CALLERID(dnid-num-plan) should have the same value.
(closes issue ASTERISK-21248)
Reported by: rmudgett
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AMI, HTTP, and chan_sip all support TLS in some way, but none of them
support all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from setting
TLS/SSL options that they do not support.
This also gets tlsverifyclient closer to a working state by requesting
the client certificate when tlsverifyclient is set. Currently, there is
no consumer of main/tcptls.c in Asterisk that supports this feature and
so it can not be properly tested.
Review: https://reviewboard.asterisk.org/r/2370/
Reported-by: John Bigelow
Patch-by: Kinsey Moore
(closes issue AST-1093)
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When a session timer expires during a dialog that has re-negotiated to T.38
and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP
containing audio media only. This causes some hilarity with the poor fax
session under weigh.
This patch corrects that by sending T.38 parameters if we are in the middle of
a T.38 session.
(closes issue ASTERISK-21232)
Reported by: Nitesh Bansal
patches:
dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418)
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In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed
to include the Username field. Somehow, one of the events was missed. This
patch corrects that - the Username field should be included in all AMI Registry
events involving SIP registrations.
(issue ASTERISK-17888)
(closes issue ASTERISK-21201)
Reported by: Dmitriy Serov
patches:
chan_sip.c.diff uploaded by Dmitriy Serov (license 6479)
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Prior to this change, certain conditions for sending the message would
result in an address of '(null)' being used in the via header of the
SIP message because a NULl value of pvt->ourip was used when initially
generating the via header. This is fixed by adding a call to build_via
when the address is set before sending the message.
(closes issue ASTERISK-21148)
Reported by: Zhi Cheng
Patches:
700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382739 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original report had to do with a realtime peer behind NAT being pruned and
the peer's private address being used instead of its external address. Upon
debugging, it was discovered that this was being caused by the addition of
the auto_force_rport and auto_comedia settings.
This patch does the following:
* Adds a missing note to the CHANGES file indicating that the default global nat
setting is auto_force_rport
* Constify the 'req' parameter for check_via()
* Add calls to check_via() in a couple of places in order for the auto_*
settings to do their job in attempting to determine if NAT is involved
* Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
settings are in use where it was needed
* Moves the copying of peer flags up in build_peer() to before they are used;
this fixes the realtime prune issue
* Update the contrib/realtime schemas to allow the nat column to handle the
different nat setting combinations we have
This patch received a review and "Ship It!" on the issue itself.
(closes issue ASTERISK-20904)
Reported by: JoshE
Tested by: JoshE, Michael L. Young
Patches:
asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it
already holds the iax2 private lock and improperly fails to obtain the channel
lock before calling ast_set_callerid. By not safely obtaining the channel lock,
a locking inversion can take place, causing a deadlock.
This patch solves this by calling the required deadlock avoidance functions
that obtain the channel lock before setting the caller ID.
Thanks to Pavel for fixing my syntax errors and testing this patch out.
(closes issue ASTERISK-21128)
Reported by: Pavel Troller
Tested by: Pavel Troller
patches:
ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302)
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Somehow, chan_jingle has managed to operate for years without setting the
sin_family on its bindaddr socket. This patch properly sets the field during
initial module load to AF_INET.
Note that the patch on the issue was modified slightly to change the
initialization of the socket from allocation of a chan_jingle private to the
module initialization, as the bindaddr object (which is static) only needs to
have the address set once.
(closes issue ASTERISK-19341)
Reported by: andre valentin
patches:
0105-chan_jingle.patch uploaded by avalentin (License 6064)
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Previously, presencestate information was sent whenever the state was not
NOT_SET. When r381594 actually returned INVALID presence state in all the
places it was supposed to, it caused chan_sip to start adding presence
state information to NOTIFY requests that it previously would not have
added. chan_sip shouldn't be adding presence state information when the
provider is in an invalid state; users can't set the state to invalid and
an invalid state always implies that the provider is in an error condition.
(issue AST-1084)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reference counting for the channel and its tech_pvt got messed up at
some point between 1.8 and 11. The result was that if a BYE for a dialog
that had been replaced (via an INVITE with Replaces) was received, Asterisk
would crash due to trying to access data on a channel that was no longer there.
The fix I introduced is to remove code that both unrefs the sip_pvt and sets
the channel's tech_pvt to NULL when an INVITE with Replaces is handled. This
way when a BYE is received, the tech_pvt will be non-NULL and so the BYE can
be processed and not cause a crash.
(closes issue ASTERISK-20929)
reported by Kristopher Lalletti
patches:
ASTERISK-20929.patch uploaded by Mark Michelson (License #5049)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some bad copy/pasting resulted in using the audio crypto attribute for both
text and video RTP. Also the audio crypto isn't set until after these, so it
was really just bad all around.
(closes ASTERISK-20905)
Reported by: Kristopher Lalletti
patches:
rtp_crypto_video_text.diff uploaded by Jonathan Rose (license 6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When sip_ref_peer and sip_unref_peer were exported to be usable in
channels/sip/security_events.c, modifications to those functions when
building under REF_DEBUG were not taken into account. This change
moves the necessary defines into sip.h to make them accessible to
other parts of chan_sip that need them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When I added my extensive suite of session timer unit tests, apparently one of
them was failing and I never noticed. If neither Min-SE nor Session-Expires is
set in the header, it was responding with a Session-Expires of the global
maxmimum instead of the configured max for the endpoint.
(issue ASTERISK-20787)
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Patch ensures that d->activeline and l->activesub are moved over to the
new device and line so that on callend the appropriate subs can be found
to complete hangup before device resets.
(closes issue ASTERISK-16610)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-reloadactive01.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Make skinny reset vmexten '\0' on reload to ensure that
it is set to '\0' if the appropriate item is removed/commented in
skinny.conf. part of ASTERISK-21037
Reported by: snuffy
Tested by: snuffy, myself
Patches:
part of immed_dial_fix.diff uploaded by snuffy (license 5024)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, Asterisk only processed session timer information if both the
'Supported: timer' and 'Session-Expires' headers were present. However, the
Session-Expires header is optional. If we were to receive a request with a
Min-SE greater than our configured session-expires, we would respond with a
'Session-Expires' header that was too small.
This patch cleans the situation up a bit, always processing timer information
if the 'Supported: timer' header is present.
(closes issue ASTERISK-20787)
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2299/
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A user in #asterisk ran into a problem where a configuration error prevented
the chan_sip module from being loaded. Upon fixing their configuratione error,
they could no longer load the chan_sip module. This was because the
configuration checking happened after the SIP provider was registered with the
Asterisk core, and subsequent attempts to load the SIP module failed as the
provider was already registered.
Since we want to detect any failure in registering chan_sip as early as
possible (as that could be emblematic of a deeper mismatch between module
and Asterisk core), this patch does not change the registration location, but
does ensure that if a module load is declined, we unregister the module as
the SIP api provider.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC5347 section 2.5.2 states the following:
...
The attribute "T38MaxBitRate" was once incorrectly registered with
IANA as "T38maxBitRate" (lower-case "m"). In accordance with T.38
examples and common implementation practice, the form "T38MaxBitRate"
SHOULD be generated by implementations conforming to this package.
In general, it is RECOMMENDED that implementations of this package
accept lowercase, uppercase, and mixed upper/lowercase encodings of
all the T.38 attributes.
...
Asterisk currently does not perform case insensitive matching on the T.38
attributes. This causes the T38MaxBitRate attribute to be negotiated at
2400 baud instead of 14400 (or whatever value you actually wanted).
This patch makes it so that when we compare T.38 attributes, we do so in a case
insensitive fashion.
Note that while the issue reporter did not directly write the patch, they
contributed to it (and would have provided one themselves if the license had
gone through a tad faster), and hence get attribution for it.
Review: https://reviewboard.asterisk.org/r/2298/
(closes issue ASTERISK-20897)
Reported by: Eric Hill
Tested by: Eric Hill
patches:
-- uploaded by Eric Hill
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Merged revisions 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Multiple channels logging in as the same agent can result in dead channels
waiting for a condition signal that will never come because another
channel thread stole it. A symptom is chan_sip repeatedly generating
warning messages about rescheduling autodestruction of dialogs with an
agent channel owner.
* Made only login_exec() (the app AgentLogin) clear the agent_pvt->chan
pointer to prevent multiple channels from logging in as the same agent.
agent_read(), agent_call(), and agent_set_base_channel() no longer
disconnect the agent channel from the agent_pvt. This also eliminates the
need to keep checking for agent_pvt->chan being NULL.
* Made agent_hangup() not wake up the AgentLogin agent thread until it is
done.
* Made agent_request() not able to get the agent until he has logged in
and any wrapup time has expired.
* Made agent_request() use ast_hangup() instead of agent_hangup() to
correctly dispose of a channel.
* Removed agent_set_base_channel(). Nobody calls it and it is a bad thing
in general.
* Made only agent_devicestate() determine the current device state of an
agent. Note: Agent group device states have never been supported.
Review: https://reviewboard.asterisk.org/r/2260/
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Merged revisions 380364 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The original fix (r380043) for getting Asterisk to respond with the correct
tag overlooked some corner cases, and the fact that the same code is in 1.8.
This patch moves the building of the crypto line out of
sdp_crypto_process(). Instead, it merely copies the accepted tag. The call to
sdp_crypto_offer() will build the crypto line in all cases now, using a tag of
"1" in the case of sending offers.
(closes issue ASTERISK-20849)
Reported by: José Luis Millán
Review: https://reviewboard.asterisk.org/r/2295/
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Merged revisions 380347 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r369028, chan_sip's processing of media streams in an SDP was modified to
better handle multiple offered media streams. Part of that change modified
how streams were declined. Previously, declined media streams were not
handled in an RFC compliant manner; now, we set the port number to 0 in the
media stream definition and proceed on with the next media stream.
Unfortunately, the formatting of the declined media stream forgot to append a
'\r\n' to the end of the media stream. This is normally added to the accepted
media streams later on in the processing of the SDP. Since the declined media
stream uses a different buffer than the accepted media streams (and is a
malloc'd buffer as opposed to a struct ast_str), it's easier to just slap the
'\r\n' on the declined media stream buffer rather than attempt to append it
later on.
So, that's what we do. And now some devices (and probably some providers) will
be a bit happier (but probably not terribly happy, since we just rejected
something they offered).
Review: https://reviewboard.asterisk.org/r/2297/
(closes issue ASTERISK-20908)
Reported by: Dennis DeDonatis
Tested by: Dennis DeDonatis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk responds with an SDP ANSWER for SRTP, it had the code to
correctly fill in the crypto data, which was overwritten by a call to
sdp_crypto_offer. Corrected the situation by changing sdp_crypto_offer
to not replacing crypto data if it already exists.
(closes issue ASTERISK-20849)
Reported by: José Luis Millán
Tested by: Iñaki Baz Castillo
Patches:
fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Generate a warning message if sound files do not exist when trying to
play the user count to the conference. Use the new helper routine
sound_file_exists() for consistency.
* Put the new user into autoservice when playing user counts to the
conference.
* Check the return value of ast_bridge_impart().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Skinny device call logging (ie missed, place and received calls) has issues
because the incorrect sequence of callstates is/can be sent to the device.
This patch removes some extra callstate updates driven by forces external
to skinny and ensures the needed intermediary callstate messages are sent.
(closes issue ASTERISK-20964)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
ast11-skinny-calllog01.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Record-Route parsing copied the header into a char[256] array, which can
be a problem if the header is longer than that. This patch parses the
header in place, without the copy, avoiding the issue.
In addition to the original patch, I added a unit test for the new
get_in_brackets_const function.
(closes issue ASTERISK-20837)
Reported by: Corey Farrell
Patches:
chan_sip-build_route-optimized-rev1.patch uploaded by Corey Farrell (license 5909)
(with minor changes by dlee)
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Merged revisions 379392 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
