The CALLERID(num) and CALLERID(name) and other info are placed into the
`char from[256]` in initreqprep. If the name was too long, the addr-spec
and params wouldn't fit.
Code is moved around so the addr-spec with params is placed there first,
and then fitting in as much of the display-name as possible.
ASTERISK-25396 #close
Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260
Although unlikely, if the tech private is returned as
a NULL, chan_pjsip_get_rtp_peer() would crash.
ASTERISK-25323
Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a
Change validation on reload module because now used the cli function for
reload. The sip_reload() function never fail and ever return NULL for this
reason on reload() now use the call the sip_reload() and return
AST_MODULE_LOAD_SUCCESS.
This problem is dectected on reload by PUT method on ARI, getting always
404 http code when the module is reloaded.
ASTERISK-25325 #close
Reporte by: Rodrigo Ramírez Norambuena
Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb
This patch adds a new option to the CHANNEL function that allows for the
extraction of the SIP call-id. It is used in conjunction with the 'pjsip'
option, and will return the Call-ID of the INVITE request that established
the PJSIP channel.
ASTERISK-25352
Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a
In chan_pjsip_new, if allocation of the pvt
structure fails, ast_hangup is called. But
it was written to assume pvt was valid, and
this change corrects that.
ASTERISK-25323
Reported by: Scott Griepentrog
Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87
The call pickup implementation in chan_sip currently sets the channel
hangup cause to "normal clearing" if call pickup is successfully
performed. This action overwrites the "answered elsewhere" hangup cause
set by the call pickup code and can result in the SIP device in
question showing a missed call when it should not.
This change sets the hangup cause to "normal clearing" as a
default initially but allows the call pickup to change it as
needed.
ASTERISK-25346 #close
Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff
There are numerous problems with the current implementation of the RTP
payload type mapping in Asterisk. It uses only one mapping structure to
associate payload types to codecs. The single mapping is overkill if all
of the payload type values are well known values. Dynamic payload type
mappings do not work as well with the single mapping because RFC3264
allows each side of the link to negotiate different dynamic mappings for
what they want to receive. Not only could you have the same codec mapped
for sending and receiving on different payload types you could wind up
with the same payload type mapped to different codecs for each direction.
1) An independent payload type mapping is needed for sending and
receiving.
2) The receive mapping needs to keep track of previous mappings because of
the slack to when negotiation happens and current packets in flight using
the old mapping arrive.
3) The transmit mapping only needs to keep track of the current negotiated
values since we are sending the packets and know when the switchover takes
place.
* Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
use the new function because ast_rtp_codecs_payload_code() was used for
mappings in both directions.
* Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
to pass preferred codec mappings to the peer channel for early media
bridging or when we need to prefer the offered mapping that RFC3264 says
we SHOULD use.
* ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
the only new public functions created. All the others were only used for
the tx or rx mapping direction so the function doxygen now reflects which
direction the function operates.
* chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
that makes no sense when processing an incoming SDP. We would be wiping
out any mappings that we set for the possible outgoing SDP we sent
earlier.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac
In chan_sip, after handling an incoming invite a security event is raised
describing authorization (success, failure, etc...). However, it was doing
a lookup of the peer by extension. This is fine for register messages, but
in the case of an invite it may search and find the wrong peer, or a non
existent one (for instance, in the case of call pickup). Also, if the peers
are configured through realtime this may cause an unnecessary database lookup
when caching is enabled.
This patch makes it so that sip_report_security_event searches by IP address
when looking for a peer instead of by extension after an invite is processed.
ASTERISK-25320 #close
Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4
Pressing DTMF digits on a phone to go out on a DAHDI channel can result in
the digit not being recognized or even heard by the peer.
Phone -> Asterisk -> DAHDI/channel
Turns out the DAHDI behavior with DTMF generation (and any other generated
tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When
Asterisk requests to start sending DTMF then DAHDI waits until its write
buffer is empty before generating any samples for the DTMF tones. When
Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI
immediately stops generating the DTMF samples. As a result, the more
samples there are in the DAHDI write buffer the shorter the time DTMF
actually gets sent on the wire. If there are more samples in the write
buffer than the time DTMF is supposed to be sent then no DTMF gets sent on
the wire. With the "buffers=12,half" setting and each buffer representing
20 ms of samples then the DAHDI write buffer is going to contain around
120 ms of samples. For DTMF to be recognized by the peer the actual sent
DTMF duration needs to be a minimum of 40 ms. Therefore, the intended
duration needs to be a minimum of 160 ms for the peer to receive the
minimum DTMF digit duration to recognize it.
A simple and effective solution to work around the DAHDI behavior is for
Asterisk to flush the DAHDI write buffer when sending DTMF so the full
duration of DTMF is actually sent on the wire. When someone is going to
send DTMF they are not likely to be talking before sending the tones so
the flushed write samples are expected to just contain silence.
* Made dahdi_digit_begin() flush the DAHDI write buffer after requesting
to send a DTMF digit.
ASTERISK-25315 #close
Reported by John Hardin
Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a
There is a window of opportunity for DTMF to not go out if an audio frame
is in the process of being written to DAHDI while another thread starts
sending DTMF. The thread sending the audio frame could be past the
currently dialing check before being preempted by another thread starting
a DTMF generation request. When the thread sending the audio frame
resumes it will then cause DAHDI to stop the DTMF tone generation. The
result is no DTMF goes out.
* Made dahdi_write() lock the private struct before writing to the DAHDI
file descriptor.
ASTERISK-25315
Reported by John Hardin
Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb
iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is
supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5,
only iLBC 30 is negotiated now.
ASTERISK-25309 #close
Change-Id: I92d724600a183eec3114da0ac607b994b1a793da
We have seen a rash of test failures on a 32-bit build agent. Commit
48698a5e21 solved an obvious problem where
we were not encoding a 64-bit value correctly over the wire. This
commit, however, did not solve the test failures.
In the failing tests, ARI is attempting to send a 537 byte text frame
over a websocket. When sending a frame this small, 16 bits are all that
is required in order to encode the payload length on the websocket
frame. However, ast_websocket_write() thinks that the payload length is
greater than 65535 and therefore writes out a 64 bit payload length.
Inspecting this payload length, the lower 32 bits are exactly what we
would expect it to be, 537 in hex. The upper 32 bits, are junk values
that are not expected to be there.
In the failure, we are passing the result of strlen() to a function that
expects a uint64_t parameter to be passed in. strlen() returns a size_t,
which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit
unsigned value to somewhere where a 64-bit unsigned value is expected
would cause no problems. In fact, in manual runs of failing tests, this
works just fine. However, ast_websocket_write() uses the Asterisk
optional API, which means that rather than a simple function call, there
are a series of macros that are used for its declaration and
implementation. These macros may be causing some sort of error to occur
when converting from a 32 bit quantity to a 64 bit quantity.
This commit changes the logic by making existing ast_websocket_write()
calls use ast_websocket_write_string() instead. Within
ast_websocket_write_string(), the 64-bit converted strlen is saved in a
local variable, and that variable is passed to ast_websocket_write()
instead.
Note that this commit message is full of speculation rather than
certainty. This is because the observed test failures, while always
present in automated test runs, never occur when tests are manually
attempted on the same test agent. The idea behind this commit is to fix
a theoretical issue by performing changes that should, at the least,
cause no harm. If it turns out that this change does not fix the failing
tests, then this commit should be reverted.
Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67
This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
Receipt of an RTP packet currently causes the formats on an PJSIP channel to
change to the format of the RTP packet. In some off-nominal cases it's possible
for this to be a format that has not been configured or negotiated. This change
makes it so only formats explicitly configured on the endpoint are allowed.
ASTERISK-25258 #close
Change-Id: If93d641fb6418a285928839300d7854cab8c1020
In channels/sig_pri.h, struct sig_pri_span, the field
force_restart_unavailable_chans is only defined if
#if defined(HAVE_PRI_MCID) is true.
All other occurences of force_restart_unavailable_chans are outside of the
#if defined(HAVE_PRI_MCID)
endif
scope.
ASTERISK-25257 #close
Reported by: Patric Marschall
Change-Id: I071de89cc2cd0d85927a013036e235851f672549
This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602
When handle_invite_replaces() was called, and either ast_bridge_impart()
failed or there was no bridge (because the channel we're picking up was
still ringing), chan_sip would leak a channel.
Thanks Matt and Corey for checking the bridge path.
ASTERISK-25226 #close
Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8
On reload, previously allowed codecs were not removed. Therefore, it was not
possible to remove codecs while Asterisk was running. Furthermore, newly added
codecs got appended behind the previous codecs. Therefore, it was not possible
to add a codec with a priority of #1. This change removes the old capabilities
before the current ones are added.
ASTERISK-25182 #close
Reported by: Alexander Traud
patches:
asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520)
Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802
Due to the use of stasis_unsubscribe_and_join in the peer destructor
it is possible for a deadlock to occur when an event callback is
occurring at the same time.
This happens because the peer may be destroyed while holding the
peers container lock. If this occurs the event callback will never
be able to acquire the container lock and the unsubscribe will
never complete.
This change makes it so the peers that have been removed from the
peers container are not destroyed with the container lock held.
ASTERISK-25163 #close
Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33
If a client sends and INVITE which is 401 rejected, then subsequently
sends a new INVITE with the auth info and uses a different fromtag
from the first INVITE, Asterisk will accept the new INVITE as part of
the original dialog - match_req_to_dialog() specifically ignores the
fromtag. However it does not update the stored dialog with the new
fromtag.
This results in Asterisk being unable to match future packets that are
part of this dialog (such as the ACK to the OK or the OK to the BYE),
and the call is dropped.
This problem was originally found when using an NEC-i SV8100-GE (NEC SIP
Card).
* After a successful match of a packet to the dialog, if the packet is
not a SIP_RESPONSE, authentication is present and the fromtags are
different, the stored fromtag is updated with the one from the recent
INVITE.
ASTERISK-25154 #close
Reported by: Damian Ivereigh
Tested by: Damian Ivereigh
Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e
Prior to this patch, chan_pjsip was failing to pass the endpoint's
context and the desired extension to the ast_channel_alloc_* routine.
This caused a new channel snapshot to be issued without a context and
extension, which can cause some reporting issues for users of AMI, CEL,
and other APIs. The channel driver would later set the context and
extension on the channel such that the channel would start in the
correct location in the dialplan, but the information reported in the
initial event would be incorrect.
This patch modifies the channel driver such that it now passes the
context and extension directly into the allocation routine. This
provides the information in the new channel snapshot published over
Stasis.
ASTERISK-25156 #close
Reported by: cloos
Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e
Although ast_context_find, ast_context_find_or_create and
ast_context_destroy perform locking of the contexts table,
any context pointer can become invalid at any time that the
contexts table is unlocked. This change adds locking around
all complete operations involving these functions.
Places where ast_context_find was followed by ast_context_destroy
have been replaced with calls ast_context_destroy_by_name.
ASTERISK-25094 #close
Reported by: Corey Farrell
Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa
Many uses of stasis_unsubscribe in modules can be reached through unload.
These have been switched to stasis_unsubscribe_and_join.
Some subscription callbacks do nothing, for these I've created a noop
callback function in stasis.c. This is used by some modules that monitor
MWI topics in order to enable cache, since the callback does not become
invalid after dlclose it is safe to use stasis_unsubscribe on these, even
during module unload.
ASTERISK-25121 #close
Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c
channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/
send_ping. This deadlock happens because the scheduled task send_lagrq(or
send_ping) starts execution after the call hangup procedure starts but before
it deletes the tasks in the scheduler.
The solution is to delete scheduled lagrq (and ping) task asynchronously
(i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will
be called in a new context (doesn't have callno locked).
This commit also cleans up the procedure of sending LAGRQ and PING.
main/sched.c: Do not assert when deleting non existant entry from scheduler.
This assert seems to be the reason for a lot of awkward code to avoid it.
ASTERISK-24983 #close
Reported by: Y Ateya
Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c
Add missing return -1 when no endpoint name is specified.
ASTERISK-25086 #close
Reported by: snuffy
Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e
If an ISDN call is hungup by both sides at the same time a crash could
happen.
* Added missing NULL checks for the owner channel after calling
pri_queue_pvt_cause_data() in two places. Code after those calls need to
check the owner channel pointer for NULL before use because
pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the
owner and the owner may get hung up.
ASTERISK-21893 #close
Reported by: Alexandr Gordeev
Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a
ast_module_info->self is often needed to register items with the core. Many
modules have ad-hoc code to make this pointer available to auxiliary sources.
This change updates the module build process to make the needed information
available to all sources in a module.
ASTERISK-25056 #close
Reported by: Corey Farrell
Change-Id: I18c8cd58fbcb1b708425f6757becaeca9fa91815
A few cases exist where headers of optional_api provders are included but
not needed. This causes unneeded calls to ast_optional_api_use.
* Don't include optional_api.h from sip_api.h.
* Move 'struct ast_channel_monitor' to channel.h.
* Don't include monitor.h from chan_sip.c, channel.c or features.c.
The move of struct ast_channel_monitor is needed since channel.c depends on
it. This has no effect on users of monitor.h since channel.h is included
from monitor.h.
ASTERISK-25051 #close
Reported by: Corey Farrell
Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478
This patch has two main purposes:
1) Improve warning messages when ACLs are configured improperly.
2) Prevent misconfigured ACLs from allowing potentially unwanted
traffic.
To acomplish point (2) in most cases, whatever configuration object that
the ACL belonged to was not allowed to load.
The one exception is res_pjsip_acl. In that case, ACLs are their own
configuration object. Furthermore, the module loading code has no
indication that a ACL configuration had a failure. So the tactic taken
here is to create an ACL that just blocks everything.
ASTERISK-24969
Reported by Corey Farrell
Change-Id: I2ebcb6959cefad03cea4d81401be946203fcacae
Some telco switches occasionally ignore ISDN RESTART requests. The fix
for ASTERISK-19608 added an escape clause for B channels in the restarting
state if the telco ignores a RESTART request. If the telco fails to
acknowledge the RESTART then Asterisk will assume the telco acknowledged
the RESTART on the second call attempt requesting the B channel by the
telco. The escape clause is good for dealing with RESTART requests in
general but it does cause the next call for the restarting B channel to be
rejected if the telco insists the call must go on that B channel.
chan_dahdi doesn't really need to issue a RESTART request in response to
receiving a cause 44 (Requested channel not available) code. Sending the
RESTART in such a situation is not required (nor prohibited) by the
standards. I think chan_dahdi does this for historical reasons to deal
with buggy peers to get channels unstuck in a similar fashion as the
chan_dahdi.conf resetinterval option.
* Add the chan_dahdi.conf force_restart_unavailable_chans compatability
option that when disabled will prevent chan_dahdi from trying to RESTART
the channel in response to a cause 44 code.
ASTERISK-25034 #close
Reported by: Richard Mudgett
Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65
Flags are 64 bits. Without LLU suffix the value of 1<<31 is negative.
Although it doesn't have an effect on the current implementation, it will
be problem if more flags are added.
Change-Id: Ic290c81cfbbbf062872392d99d3322932cc49487
* The REF_DEBUG compiler flag no longer has any effect on code that uses
Astobj2. It is used to determine if reference debugging is enabled by
default. Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
This was possible now that we no longer require a dual ABI.
ASTERISK-24974 #close
Reported by: Corey Farrell
Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1
A typo in commit f8e21a1adf resulted in a compilation error in
chan_skinny. This patch fixes the typo.
ASTERISK-24917
Change-Id: Id7f4ad1fe948eb2408622e80c27936ce4516c33c
clang can warn about a so called tautological-compare, when it finds
comparisons which are logically always true, and are therefor deemed
unnecessary.
Exanple:
unsigned int x = 4;
if (x > 0) // x is always going to be bigger than 0
Enum Case:
Each enumeration is its own type. Enums are an integer type but they
do not have to be *signed*. C leaves it up to the compiler as an
implementation option what to consider the integer type of a particu-
lar enumeration is. Gcc treats an enum without negative values as
an int while clang treats this enum as an unsigned int.
rmudgett & mmichelson: cast the enum to (unsigned int) in assert.
The cast does have an effect. For gcc, which seems to treat all enums
as int, the cast to unsigned int will eliminate the possibility of
negative values being allowed. For clang, which seems to treat enums
without any negative members as unsigned int, the cast will have no
effect. If for some reason in the future a negative value is ever
added to the enum the assert will still catch the negative value.
ASTERISK-24917
Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62
The chan_dahdi channel driver is a very old driver. The ability for it to
support ISDN was added well after the initial analog support. Setting the
softhangup flags is a carry over from the original analog code. The
driver was not updated to call ast_queue_hangup() which will post the AMI
HangupRequest event.
* Changed sig_pri.c to call ast_queue_hangup() instead of setting the
softhangup flag when the remote party initiates a hangup.
ASTERISK-24895 #close
Reported by: Andrew Zherdin
Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325
This new macro allows a single line to add all additional
sources to a module. This helps prevent modules from
missing steps, and makes future changes easier since
they can be made in a single place.
ASTERISK-24960 #close
Reported by: Corey Farrell
Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
With this patch, chan_pjsip/res_pjsip now sets the native formats to the
codecs negotiated by a call.
* The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native
formats to include all the negotiated audio codecs instead of only the
initial preferred audio codec and later the currently received audio
codec.
* The audio frame handling in channel.c:ast_read() is more streamlined and
will automatically adjust to changes in received frame formats. The new
policy is to remove translation and pass the new frame format to the
receiver except if the translation was to a signed linear format. A more
long winded version is commented in ast_read() along with some caveats.
* The audio frame handling in channel.c:ast_write() is more streamlined
and will automatically adjust any needed translation to changes in the
frame formats sent. Frame formats sent can change for many reasons such
as a recording is being played back or the bridged peer changed the format
it sends. Since it is a normal expectation that sent formats can change,
the codec mismatch warning message is demoted to a debug message.
* Removed the short circuit check in
channel.c:ast_channel_make_compatible_helper(). Two party bridges need to
make channels compatible with each other. However, transfers and moving
channels among bridges can result in otherwise compatible channels having
sub-optimal translation paths if the make compatible check is short
circuited. A result of forcing the reevaluation of channel compatibility
is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc
options take effect consistently now. It is unfortunate that these two
options are enabled by default and negate some of the benefits to the
changes in channel.c:ast_read() by forcing translation through signed
linear on a two party bridge.
* Improved the softmix bridge technology to better control the translation
of frames to the bridge. All of the incoming translation is now normally
handled by ast_read() instead of splitting any translation steps between
ast_read() and the slin factory. If any frame comes in with an unexpected
format then the translation path in ast_read() is updated for the next
frame and the slin factory handles the current frame translation.
This is the final patch in a series of patches aimed at improving
translation path choices. The other patches are on the following reviews:
https://reviewboard.asterisk.org/r/4600/https://reviewboard.asterisk.org/r/4605/
ASTERISK-24841 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4609/
........
Merged revisions 434671 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for automatically detecting the type of DTMF that a
PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
the channel created for an endpoint will attempt to determine if RFC 4733
DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
for the channel will be set to inband.
Review: https://reviewboard.asterisk.org/r/4438
ASTERISK-24706 #close
Reported by: yaron nahum
patches:
yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
........
Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
POKE is used to check for peer availability; however, in networks with packet
loss, the current calculations may result in POKE expiration times that are too
short. This patch alters the expiration/retry time logic to take into account
the last known qualify round trip time, as opposed to always using a static
value for each peer.
Review: https://reviewboard.asterisk.org/r/4536
ASTERISK-22352 #close
Reported by: Frederic Van Espen
ASTERISK-24894 #close
Reported by: Y Ateya
patches:
poke_noanswer_duration.diff submitted by Y Ateya (License 6693)
........
Merged revisions 434564 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434565 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new configuration parameter, 'calltokenexpiration', that
controls how long before an authentication call token is expired. The default
maintains the RFC specified 10 seconds. Setting it to a higher value may be
useful in lossy networks.
Review: https://reviewboard.asterisk.org/r/4588
ASTERISK-24939 #close
Reported by: Y Ateya
patches:
ctoken_configuration.diff submitted by Y Ateya (License 6693)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes autological comparison warnings in the following:
* chan_skinny: letohl may return a signed or unsigned value, depending on the
macro chosen
* func_curl: Provide a specific cast to CURLoption to prevent mismatch
* cel: Fix enum comparisons where the enum can never be negative
* enum: Fix comparison of return result of dn_expand, which returns a signed
int value
* event: Fix enum comparisons where the enum can never be negative
* indications: tone_data.freq1 and freq2 are unsigned, and hence can never be
negative
* presencestate: Use the actual enum value for INVALID state
* security_events: Fix enum comparisons where the enum can never be negative
* udptl: Don't bother to check if the return value from encode_length is less
than 0, as it returns an unsigned int
* translate: Since the parameters are unsigned int, don't bother checking
to see if they are negative. The cast to unsigned int would already blow
past the matrix bounds.
* res_pjsip_exten_state: Use a temporary value to cache the return of
ast_hint_presence_state
* res_stasis_playback: Fix enum comparisons where the enum can never be
negative
* res_stasis_recording: Add an enum value for the case where the recording
operation is in error; fix enum comparisons
* resource_bridges: Use enum value as opposed to -1
* resource_channels: Use enum value as opposed to -1
Review: https://reviewboard.asterisk.org/r/4533
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4533.patch submitted by dkdegroot (License 6600)
........
Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434470 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a channel enters the bridging system it is first made compatible with
the bridge and then the bridge technology makes the channel compatible
with the technology. For all but the DAHDI native and softmix bridge
technologies the make channel compatible with the bridge step is an
effective noop because the other technologies allow all audio formats.
For the DAHDI native bridge technology it doesn't matter because it is not
an initial bridge technology and chan_dahdi allows only one native format
per channel. For the softmix bridge technology, it is a noop at best and
harmful at worst because the wrong translation path could be setup if the
channel's native formats allow more than one audio format.
This is an intermediate patch for a series of patches aimed at improving
translation path choices.
* Removed code dealing with the unnecessary step of making the channel
compatible with the bridge.
ASTERISK-24841
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4600/
........
Merged revisions 434424 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes an access to the peer callnumber that is unprotected by a
corresponding mutex. The peer->callno value can be changed by multiple threads,
and all data inside the iaxs array must be procted by a corresponding lock
of iaxsl.
The patch moves the unprotected access to a location where the mutex is
safely obtained.
Review: https://reviewboard.asterisk.org/r/4599/
ASTERISK-21211 #close
Reported by: Jaco Kroon
patches:
asterisk-11.2.1-iax2_poke-segfault.diff submitted by Jaco Kroon (License 5671)
........
Merged revisions 434291 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434292 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will
attempt to handle both IPv4 and IPv6 addresses, although the information will
be stored in a struct with an AF_INET6 address type. However, the current
NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly.
This patch adds an additional check for the mapped address case, allowing
the NAT code to handle clients even when the address is IPv6.
Review: https://reviewboard.asterisk.org/r/4563/
ASTERISK-18032 #close
Reported by: Christoph Timm
patches:
nat_with_ipv6.diff submitted by Valentin Vidić (License 6697)
........
Merged revisions 434288 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434289 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes several warnings pointed out by the clang compiler.
* chan_pjsip: Removed check for data->text, as it will always be non-NULL.
* app_minivm: Fixed evaluation of etemplate->locale, which will always
evaluate to 'true'. This patch changes the evaluation to use
ast_strlen_zero.
* app_queue:
- Fixed evaluation of qe->parent->monfmt, which always evaluates to
true. Instead, we just check to see if the dereferenced pointer
evaluates to true.
- Fixed evaluation of mem->state_interface, wrapping it with a call to
ast_strlen_zero.
* res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero.
Review: https://reviewboard.asterisk.org/r/4541
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4541.patch submitted by dkdegroot (License 6600)
........
Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434286 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These are fixes for compilation under gcc 5.0...
chan_sip.c: In parse_request needed to make 'lim' unsigned.
inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99
inline semantics (same as clang).
ccss.c: In ast_cc_set_parm, needed to fix weird comparison.
dsp.c: Needed to work around a possible compiler bug. It was throwing
an array-bounds error but neither
sgriepentrog, rmudgett nor I could figure out why.
manager.c: In action_atxfer, needed to correct an array allocation.
This patch will go to 11, 13, trunk.
Review: https://reviewboard.asterisk.org/r/4581/
Reported-by: Jeffrey Ollie
Tested-by: George Joseph
ASTERISK-24932 #close
........
Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 434114 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes some invalid enum conversion warnings caught by clang. In
particular:
* chan_sip: Several functions mixed usage of the st_refresher_param
enum and st_refresher enum. This patch corrects the functions to use the
right enum.
* chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state.
* strings: Fixed incorrect usage of AO2 flags with strings container.
* res_stasis: Change a return enumeration to stasis_app_user_event_res.
Review: https://reviewboard.asterisk.org/r/4535
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4535.patch submitted by dkdegroot (License 6600)
........
Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 433747 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
........
Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The res_pjsip modules were manually checking both name and number
presentation values when there is a function that determines the combined
presentation for a party ID struct. The function takes into account if
the name or number components are valid while the manual code rarely
checked if the data was even valid.
* Made use ast_party_id_presentation() rather than manually checking party
ID presentation values.
* Ensure that set_id_from_pai() and set_id_from_rpid() will not return
presentation values other than what is pulled out of the SIP headers. It
is best if the code doesn't assume that AST_PRES_ALLOWED and
AST_PRES_USER_NUMBER_UNSCREENED are zero.
* Fixed copy paste error in add_privacy_params() dealing with RPID
privacy.
* Pulled the id->number.valid test from add_privacy_header() and
add_privacy_params() up into the parent function add_id_headers() to skip
adding PAI/RPID headers earlier.
* Made update_connected_line_information() not send out connected line
updates if the connected line number is invalid. Lower level code would
not add the party ID information and thus the sent message would be
unnecessary.
* Eliminated RAII_VAR usage in send_direct_media_request().
Review: https://reviewboard.asterisk.org/r/4472/
........
Merged revisions 432892 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Switch logger callid's from AO2 objects to simple integers.
This helps in two ways. Copying integers is faster than
referencing AO2 objects, so this will result in a small
reduction in logger overhead. This also erases the possibility
of an infinate loop caused by an invalid callid in
threadstorage.
ASTERISK-24833 #comment Committed callid conversion to trunk.
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4466/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The distinctive ring feature interferes with detecting Caller ID and
appears to have been broken for years. What happens is if you have a
ring-ring cadence as used in the UK you get too many DAHDI events for the
distinctive ring pattern array and Caller ID detection is aborted. I
think when Zapata/DAHDI added the ring begin event it broke distinctive
ring. More events happen than before and the code does no filtering of
which event times are recorded in the pattern array.
* Made distinctive ring only record the ringt count when the ring ends
instead of on just any DAHDI event. Distinctive ring can be ring,
ring-ring, ring-ring-ring, or different ring durations for the up to three
rings.
* Fixed the distinctive ring detection enable (chan_dahdi.conf option
usedistinctiveringdetection) to be per port instead of somewhat per port
and somewhat global. This has been broken since v1.8.
* Fixed using the default distinctive ring context when the detected
pattern does not match any configured dringX patterns. The default
context did not get set when the previous call was a matched distinctive
ring pattern and the current call is not matched. This has been broken
since v1.8.
* Made distinctive ring have no effect on Caller ID detection when it is
disabled. Caller ID detection just monitors for 10 seconds before giving
up.
* Fixed leak of struct callerid_state memory when a polarity reversal
during Caller ID detection causes the incoming call to be aborted.
DAHDI-1143
AST-1545
ASTERISK-24825 #close
Reported by: Richard Mudgett
ASTERISK-17588
Reported by: Daniel Flounders
Review: https://reviewboard.asterisk.org/r/4444/
........
Merged revisions 432530 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 432534 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk fails to start a PBX thread for a new channel - for example, when
the maxcalls setting in asterisk.conf is exceeded - we currently send a final
response, and then attempt to send a BYE request to the UA. Since that's all
sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt
such that we don't get stuck sending BYE requests to something that does not
want it.
Note that this patch is a slight modification of the one on ASTERISK-15434.
For clarity, it explicitly calls sipalreadygone with the calls to transmit a
final response.
ASTERISK-21845
ASTERISK-15434 #close
Reported by: Makoto Dei
Tested by: Matt Jordan
patches:
sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027)
........
Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 432321 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we receive an SDP as part of an offer/answer for a peer/friend has been
configured to require encryption, and that SDP offer/answer failed to provide
acceptable crypto attributes, we currently issue a WARNING that uses the phrase
"we" and "requested". In this case, both of those terms are ambiguous - the
user will probably think "we" is Asterisk (it most likely isn't) and it may
not be a "request", so much as an SDP that was received in some fashion.
This patch makes the WARNING messages slightly less bad and a bit more
accurate as well.
ASTERISK-23214 #close
Reported by: Rusty Newton
........
Merged revisions 432277 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 432278 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the monitor thread is stopped, its pthread ID is set to a specific value
(AST_PTHREADT_STOP) so that later portions of the code can determine whether
or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit
failed to check for that value, checking instead only for AST_PTHREAD_STOP.
Passing the invalid yet very specific value to pthread_kill causes a crash.
This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that
it doesn't attempt to poke the thread if the thread has already been stopped.
ASTERISK-24800 #close
Reported by: JoshE
........
Merged revisions 432198 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 432199 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses the following problems:
* ari/resource_channels: In ARI, we currently create a format capability
structure of SLIN and apply it to the new channel being created. This was
originally done when the PBX core was used to create the channel, as there
was a condition where a newly created channel could be created without any
formats. Unfortunately, now that the Dial API is being used, this has two
drawbacks:
(a) SLIN, while it will ensure audio will flows, can cause a lot of
needless transcodings to occur, particularly when a Local channel is
created to the dialplan. When no format capabilities are available, the
Dial API handles this better by handing all audio formats to the requsted
channels. As such, we defer to that API to provide the format
capabilities.
(b) If a channel (requester) is causing this channel to be created, we
currently don't use its format capabilities as we are passing in our own.
However, the Dial API will use the requester channel's formats if none
are passed into it, and the requester channel exists and has format
capabilities. This is the "best" scenario, as it is the most likely to
create a media path that minimizes transcoding.
Fixing this simply entails removing the providing of the format capabilities
structure to the Dial API.
* chan_pjsip: Rather than blindly picking the first format in the format
capability structure - which actually *can* be a video or text format - we
select an audio format, and only pick the first format if that fails. That
minimizes the weird scenario where we attempt to transcode between video/audio.
* res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure.
Since ast_request already limits us down to one format capability once the
format capabilities are passed along, there's no reason to squelch it here.
* channel: Fixed a comment. The reason we have to minimize our requested
format capabilities down to a single format is due to Asterisk's inability
to convey the format to be used back "up" a channel chain. Consider the
following:
PJSIP/A => L;1 <=> L;2 => PJSIP/B
g,u,a g,u,a g,u,a u
That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials
PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local
channel has inherited those format capabilities down the line; PJSIP/B
supports only ulaw. According to these format capabilities, ulaw is
acceptable and should be selected across all the channels, and no
transcoding should occur. However, there is no way to convey this: when L;2
and PJSIP/B are put into a bridge, we will select ulaw, but that is not
conveyed to PJSIP/A and L;1. Thus, we end up with:
PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
g g X u u
Which causes g722 to be written to PJSIP/B.
Even if we can convey the 'ulaw' choice back up the chain (which through
some severe hacking in Local channels was accomplished), such that the chain
looks like:
PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
u u u u
We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back
with only 'ulaw'. This results in all the channel structures being set up
correctly, but PJSIP/A *still* sending g722 and causing the chain to fall
apart.
There's a lot of difficulty just in setting this up, as there are numerous
race conditions in the act of bridging, and no clean mechanism to pass the
selected format backwards down an established channel chain. As such, the
best that can be done at this point in time is clarifying the comment.
Review: https://reviewboard.asterisk.org/r/4434/
ASTERISK-24812 #close
Reported by: Matt Jordan
........
Merged revisions 432195 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.
*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.
*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
only transfer channels to a SIP URI, i.e., you had to pass
'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
still supported, it is somewhat unintuitive - particularly in a world full
of endpoints. As such, we now also support specifying the PJSIP endpoint to
transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
updating its Contact header. Alas, that resulted in the forwarding
destination set by the dialplan application/ARI resource/whatever being
rewritten with very incorrect information. Hence, we now don't bother
updating an outgoing response if it is a 302. Since this took a looong time
to find, some additional debug statements have been added to those modules
that update the Contact headers.
Review: https://reviewboard.asterisk.org/r/4316/
ASTERISK-24015 #close
Reported by: Private Name
ASTERISK-24703 #close
Reported by: Matt Jordan
........
Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are three CLI commands to stop and restart Asterisk each.
1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
New channels are prevented while the shutdown request is pending.
2) core stop/restart gracefully - Stop or restart Asterisk when there are
no calls remaining in the system. New channels are prevented while the
shutdown request is pending.
3) core stop/restart when convenient - Stop or restart Asterisk when there
are no calls in the system. New calls are not prevented while the
shutdown request is pending.
ARI has made stopping/restarting Asterisk more problematic. While a
shutdown request is pending it is desirable to continue to process ARI
HTTP requests for current calls. To handle the current calls while a
shutdown request is pending, a new committed to shutdown phase is needed
so ARI applications can deal with the calls until the system is fully
committed to shutdown.
* Added a new shutdown committed phase so ARI applications can deal with
calls until the final committed to shutdown phase is reached.
* Made refuse new HTTP requests when the system has reached the final
system shutdown phase. Starting anything while the system is actively
releasing resources and unloading modules is not a good thing.
* Split the bridging framework shutdown to not cleanup the global bridging
containers when shutting down in a hurry. This is similar to how other
modules prevent crashes on rapid system shutdown.
* Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
ast_shutting_down(). You should not have to include channel.h just to
access these system functions.
ASTERISK-24752 #close
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/4399/
........
Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a SIP device that has its registration stored in RealTime unregisters,
the entry for that device is updated with blank values, i.e., "", indicating
that it is no longer registered. Unfortunately, one of those values that is
'blanked' is the device's port. If the column type for the port is not a
string datatype (the recommended type is integer), an ODBC or database error
will be thrown. MariaDB does not coerce empty strings to a valid integer value.
This patch updates the query run from chan_sip such that it replaces the port
value with a value of '0', as opposed to a blank value. This is the value that
other database backends coerce the empty string ("") to already, and the
handling of reading a RealTime registration value from a backend already
anticipates receiving a port of '0' from the backends.
ASTERISK-24772 #close
Reported by: Richard Miller
patches:
chan_sip.diff uploaded by Richard Miller (License 5685)
........
Merged revisions 431673 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 431674 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
scenarios when we are required to use SIPS URIs in Contact
headers. Asterisk's non-compliance with this could actually
cause calls to get dropped when communicating with clients
that are strict about checking the Contact header.
Both of the SIP stacks in Asterisk suffered from this issue.
This changeset corrects the behavior in chan_sip.
ASTERISK-24646 #close
Reported by Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/4346
........
Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 431424 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When refreshing (with a small expiration) a registration that was sent to
chan_sip the nonce would be considered stale and reject the registration.
What was happening was that the initial registration's "dialog" still existed
in the dialogs container and upon refresh the dialog match algorithm would
choose that as the "dialog" instead of the newly created one. This occurred
because the algorithm did not check to see if the from tag matched if
authentication info was available after the 401. So, it ended up assuming
the original "dialog" was a match and stopped the search. The old "dialog"
of course had an old nonce, thus the stale nonce message.
This fix attempts to leave the original functionality alone except in the case
of a REGISTER. If a REGISTER is received if searches for an existing "dialog"
matching only on the callid. If the expires value is low enough it will reuse
dialog that is there, otherwise it will create a new one.
ASTERISK-24715 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4367/
........
Merged revisions 431187 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 431194 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses compilation errors on OS X. It's been a while, so
there's quite a few things.
* Fixed __attribute__ decls in route.h to be portable.
* Fixed htonll and ntohll to work when they are defined as macros.
* Replaced sem_t usage with our ast_sem wrapper.
* Added ast_sem_timedwait to our ast_sem wrapper.
* Fixed some GCC 4.9 warnings using sig*set() functions.
* Fixed some format strings for portability.
* Fixed compilation issues with res_timing_kqueue (although tests still fail
on OS X).
* Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue
on OS X).
ASTERISK-24539 #close
Reported by: George Joseph
ASTERISK-24544 #close
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/4327/
........
Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the SIP registrations were migrated to using ao2 in what was then trunk,
the explicit destruction of the registrations on module reload was removed and
not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the
issue reporter, on ASTERISK-24673 confirmed that the reference in the
registry_list container was being leaked.
Since the purpose of cleanup_all_regs is to prep a registration for
destruction, this function now calls an ao2_callback function callback with the
OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the registrations.
This cleans up each registration, and also removes it from the registration
container registry_list.
Review: https://reviewboard.asterisk.org/r/4355/
ASTERISK-24640 #close
Reported by: Max Man
ASTERISK-24673 #close
Reported by: Stefan Engström
Tested by: Stefan Engström
........
Merged revisions 430864 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Calling ast_channel_bridge_peer() cannot be done while holding any channel
locks. The reported issue hit the deadlock in chan_iax2, but an audit of
the ast_channel_bridge_peer() calls found three more locations where the
same deadlock can occur.
* Made CHANNEL(peer), res_fax, and the SNMP agent not call
ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I
had to rework the logic to not hold the channel lock.
* Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done
for legacy reasons that no longer apply.
* Removed the iax.conf forcejitterbuffer option. It is now always enabled
when the jitterbuffer option is enabled. If you put a jitter buffer on a
channel it will be on the channel.
ASTERISK-24600 #close
Reported by: Jeff Collell
Review: https://reviewboard.asterisk.org/r/4342/
........
Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Different clients react differently to being told that a blind transfer
has failed. Some will simply send a BYE and be done with it. Others will
attempt to reinvite themselves back onto the call.
In the latter case, we were creating a new channel and then leaving it to
sit forever doing nothing. With this code change, that new channel will
not be created and the dialog with the transferring channel will be cleaned
up properly.
ASTERISK-24624 #close
Reported by Zane Conkle
Review: https://reviewboard.asterisk.org/r/4339
........
Merged revisions 430714 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The 'pjsip_get_dest_info' function is used to determine if the signaling transport
of the dialog is secure or not. This function was added in PJSIP 2.3 and does not
exist in earlier versions.
This configure check allows Asterisk to build and run with older versions at the
loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of
this argument will require upgrading to PJSIP 2.3.
ASTERISK-24665 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4329/
........
Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
........
Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Given the following scenario:
* Three SIP phones (A, B, C), all communicating via a proxy with Asterisk
* A call is established between A and B. B performs a SIP attended transfer of
A to C. B sets the call on hold (A is hearing MOH) and dials the extension of
C. While phone C is ringing, B transfers the call (that is, what we typically
call a 'blond transfer').
* When the transfer completes, A hears the ringing of phone C, while B is idle.
In the SIP messaging for the above scenario, a REFER request is sent to
transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an
UPDATE request to phone C to update party information. This update is sent
directly to phone C, not through the intervening proxy. This has the unfortunate
side effect of providing route information, which is then set on the sip_pvt
structure for C. If someone (e.g. B) is trying to get the call back (through a
directed pickup), Asterisk will send a CANCEL request to C. However, since we
have now updated the route set, the CANCEL request will be sent directly to C
and not through the proxy. The phone ignores this CANCEL according to RFC3261
(Section 9.1).
This patch updates reqprep such that the route is not updated if an UPDATE
request is being sent while the INVITE state is INV_PROCEEDING or
INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent
to the correct location.
Review: https://reviewboard.asterisk.org/r/4279
ASTERISK-24628 #close
Reported by: Karsten Wemheuer
patches:
issue.patch uploaded by Karsten Wemheuer (License 5930)
........
Merged revisions 429982 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429983 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the configuration section scheme of chan_dahdi.conf is used (keyword
dahdichan instead of channel) all setvar= options are completely ignored.
No variable defined this way appears in the created DAHDI channels.
* Move the clearing of setvar values to after the deferred processing of
dahdichan.
AST-1378 #close
Reported by: Guenther Kelleter
Patch by: Guenther Kelleter
........
Merged revisions 429825 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429829 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the featdmf signaling mode the incoming MF Caller-ID information is
formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}#
Rather than discarding the ani2 digits, populate the CALLERID(ani2) value
with what is received instead.
AST-1368 #close
Reported by: Denis Martinez
Patches:
extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett
........
Merged revisions 429783 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429784 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, we were using the PJSIP dialog's secure flag
to determine if a secure transport was being used. Unfortunately,
the dialog's secure flag was only set if a SIPS URI were in use,
as required by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested
in is not dialog security, but transport security. This code change
switches to a model where we use the dialog's target URI to determine
what transport would be used to communicate, and then check if that
transport is secure.
AST-1450 #close
Reported by John Bigelow
Review: https://reviewboard.asterisk.org/r/4277
........
Merged revisions 429739 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:
-out += sprintf(out, "%%%02X", (unsigned char) *ptr);
+out += sprintf(out, "%%%02X", (unsigned) *ptr);
That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.
This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)
Review: https://reviewboard.asterisk.org/r/4263/
ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
........
Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 429675 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously when SRTP was enabled on a channel it was not possible
to switch to T.38 as no crypto attributes would be present.
This change makes it so it is now possible. If a T.38 re-invite
comes in SRTP is terminated since in practice you can't encrypt
a UDPTL stream. Now... if we were doing T.38 over RTP (which
does exist) then we'd have a chance but almost nobody does that so
here we are.
ASTERISK-24449 #close
Reported by: Andreas Steinmetz
patches:
udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523)
........
Merged revisions 429632 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 429633 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When direct media is enabled and a pjsip channel is answered a race would occur
between the handling of the answer and bridge setup. Sometimes the media
negotiation would take place after the native bridge was setup. This resulted
in a NULL media address, which in turn resulted in Asterisk using its address
as the remote media address when sending a reinvite. This patch makes the
chan_pjsip answer handler synchronous thus alleviating the race condition (the
bridge won't start setting things up until after it returns).
ASTERISK-24563 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4257/
........
Merged revisions 429477 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Given the scenario where a PJSIP channel is in a native RTP bridge with direct
media and the channel is then hung up the code will currently re-INVITE the channel
back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
this greatly.
This change makes it so that if a re-INVITE transaction is in progress the BYE
is queued to occur after the completion of the transaction (be it through normal
means or a timeout).
Review: https://reviewboard.asterisk.org/r/4248/
........
Merged revisions 429409 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When endpoints with direct_media enabled, behind a firewall (Asterisk on a
separate network) and were bridged sometimes Asterisk would send the ip
address of the firewall in the sdp to one of the phones in the reinvite
resulting in one way audio. When sending the reinvite Asterisk will retrieve
the media address from the associated rtp instance, but if frames were being
read this can be overwritten with another address (in this case the
firewall's). This patch ensures that Asterisk uses the original device
address when using direct media.
ASTERISK-24563
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4216/
........
Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 429196 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
........
Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to the serialized architecture of chan_pjsip there exists a race condition where a CANCEL may
be received and processed before responses (such as 180 Ringing, 183 Session Progress, and 200 OK)
are sent. Since the session is in an unexpected state PJSIP will assert when this is attempted.
This change makes it so that these responses are not sent on disconnected sessions.
ASTERISK-24471 #close
Reported by: yaron nahum
........
Merged revisions 428301 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 428302 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For chan_motif the direct return value of the underlying config options framework
was passed back. This can relay various states which the module loader would not
interpet as success. It has been changed so only on errors will it report back
an error.
For chan_pjsip the code implemented a dummy reload function which always
returned an error. This has been removed as all configuration is held within
res_pjsip instead.
ASTERISK-23651 #close
Reported by: Rusty Newton
........
Merged revisions 427981 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 427982 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Configuration of DTLS in the general section will be applied to any users
or peers. If configuration exists at their level it overrides the general
section values.
ASTERISK-24128 #close
Reported by: Michael K.
patches:
dtls_default_settings.patch submitted by Michael K. (license 6621)
Review: https://reviewboard.asterisk.org/r/3867/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r227276, a while loop was turned into a for loop. Unfortunately, a portion
of the while loop was left in the code such that, when a static gateway is
encountered in the list of MGCP gateways, the next gateway would be skipped.
At best, we would simply flip past a gateway; at worst, this could lead to a
crash.
ASTERISK-24500 #close
Reported by: Xavier Hienne
patches:
chan_mgcp.patch uploaded by Xavier Hienne (License 6657)
........
Merged revisions 427613 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 427614 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 427615 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Review: https://reviewboard.asterisk.org/r/4103/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.
This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.
#SIPit31
ASTERISK-24382 #close
Reported by: Matt Jordan
........
Merged revisions 424621 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 424622 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.
A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade. With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer. Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.
* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.
* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.
* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.
ASTERISK-24356 #close
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/4034/
........
Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This gets rid of most old libc free/malloc/realloc and replaces them
with ast_free and friends. When compiling with MALLOC_DEBUG you'll
notice it when you're mistakenly using one of the libc variants. For
the legacy cases you can define WRAP_LIBC_MALLOC before including
asterisk.h.
Even better would be if the errors were also enabled when compiling
without MALLOC_DEBUG, but that's a slightly more invasive header
file change.
Those compiling addons/format_mp3 will need to rerun
./contrib/scripts/get_mp3_source.sh.
ASTERISK-24348 #related
Review: https://reviewboard.asterisk.org/r/4015/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Outgoing PJSIP calls can result in non-negotiated formats listed in the
channel's native formats if video formats are listed in the endpoint's
configuration. The resulting call could then use a non-negotiated format
resulting in one way audio.
* Simplified the update of session->req_caps in set_caps(). Why do
something in five steps when only one is needed?
AFS-162 #close
Review: https://reviewboard.asterisk.org/r/4000/
........
Merged revisions 423561 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This module supports sending both unicast and multicast RTP
to a specified target. Multicast functionality is the same as
chan_multicast_rtp was. In the case of unicast a specific
IP address and port can be specified, along with optional RTP
engine and format in the form of:
UnicastRTP/<ip address>:<port>/<engine>/<format>
This can be useful for sending a copy of a media stream to
another application for processing.
Review: https://reviewboard.asterisk.org/r/3981/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Testsuite tests will occasionally fail because on reception of a 200 OK SIP response,
an AST_CONTROL_ANSWER frame is queued prior to when media has finished being
negotiated. This is because session supplements are called into before PJSIP's
inv_session code has told us that media has been updated. Sometimes the queued answer
frame is handled by the PBX thread before the ensuing media negotiations occur, causing
a test failure.
As it turns out, there is another place that session supplements could be called into, which is
after media has finished getting negotiated. What this commit introduces is a means for session
supplements to indicate when they wish to be called into when handling an incoming SIP response.
By default, all session supplements will be run at the same point that they were prior to this
commit. However, session supplements may indicate that they wish to be handled earlier than
normal on redirects, or they may indicate they wish to be handled after media has been negotiated.
In this changeset, two session supplements have been updated to indicate a preference for when
they should be run: res_pjsip_diversion executes before handling redirection in order to get
information from the Diversion header, and chan_pjsip now handles responses to INVITEs after
media negotiation to fix the race condition mentioned previously.
ASTERISK-24212 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/3930
........
Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The reporter on the issue found some issues when upgrading from version 10 to 11
on 55 hosts.
Two situations that can occur with dynamic registrations.
1. With dnsmgr disabled, if the host is not resolvable we are not trying to
resolve the host again when it is time to attempt to register again. This
results in never registering to the host.
2. With dnsmgr enabled, when the host is temporarily not resolvable the
address is set to 0.0.0.0:0 and then when the host is resolvable the port
is not being restored and stays set to 0.
This patch resolves these two issues by:
* Storing the hostname so that it can be used for resolving with DNS.
* Resolve the hostname on the next scheduled attempt to register.
* Storing the port used to reach the host so that when the hostname is
resolvable again, we can set the port again if the port is still unset after
looking up the host.
ASTERISK-23767 #close
Reported by: David Herselman
Tested by: David Herselman, Michael L. Young
Patches:
asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3856/
........
Merged revisions 422274 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 422275 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 422276 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Users now have the ability to bind the rtpengine instance to a specific IP
address. For example, you want chan_sip (call control) on eth0 but rtp (media)
on eth1.
ASTERISK-24280 #close
Reported by: Paul Belanger
Tested by: Paul Belanger
Review: https://reviewboard.asterisk.org/r/3952/
Patches:
rtpengine.diff uploaded by Paul Belanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code originally worked around an issue within res_rtp_asterisk itself.
The wrong socket was being used for the STUN check for RTCP, causing the
port to be the same as RTP. This was subsequently fixed and the RTCP port
provided for the ICE candidate is correct and does not need to be incremented.
ASTERISK-23997 #close
Reported by: Badalian Vyacheslav
Patches:
plus1.diff submitted by Badalian Vyacheslav (license 5249)
........
Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 421911 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.
* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite. AFS-63 was effectively reintroduced because of the media
formats work. res_pjsip_sdp_rtp.c:set_caps()
* Improved the unexpected frame format WARNING message to include more
information.
* Added protective locking while altering formats on a channel. Reworked
set_format() to simplify and protect the formats under manipulation.
* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())
AFS-137 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3906/
........
Merged revisions 421645 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A calls B
B answers
B SIP attended transfers to C
C answers, B and C can see each other's connected line information
B completes the transfer
A has number but no name connected line information about C
while C has the full information about A
I examined the incoming and outgoing party id information handling of
chan_pjsip and found several issues:
* Fixed ast_sip_session_create_outgoing() not setting up the configured
endpoint id as the new channel's caller id. This is why party A got
default connected line information.
* Made update_initial_connected_line() use the channel's CALLERID(id)
information. The core, app_dial, or predial routine may have filled in or
changed the endpoint caller id information.
* Fixed chan_pjsip_new() not setting the full party id information
available on the caller id and ANI party id. This includes the configured
callerid_tag string and other party id fields.
* Fixed accessing channel party id information without the channel lock
held.
* Fixed using the effective connected line id without doing a deep copy
outside of holding the channel lock. Shallow copy string pointers can
become stale if the channel lock is not held.
* Made queue_connected_line_update() also update the channel's
CALLERID(id) information. Moving the channel to another bridge would need
the information there for the new bridge peer.
* Fixed off nominal memory leak in update_incoming_connected_line().
* Added pjsip.conf callerid_tag string to party id information from
enabled trust_inbound endpoint in caller_id_incoming_request().
AFS-98 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3913/
........
Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed the iax.conf bandwidth option. This is the root cause of
ASTERISK-24150.
* Added checks in iax2_request() to ensure that there are actual formats
requested for the new channel to prevent any more fracks from issues like
ASTERISK-24150. This is a consequence of the iax.conf bandwidth option
not working.
* Fixed struct iax2_codec_pref.order member size mismatch issue when
converting to and from the codec preference order list passed over the
wire. In addition the values sent over the wire are now compatible with
previous Asterisk versions.
* Fixed several issues dealing with the struct iax2_codec_pref members.
Off-by-one, array limit errors, and the order/framing members always need
to be updated together.
* Made iax2_request() setup the channel's native format preference order
according to the user's wishes. The new media format strategy needs the
order specified earler.
* Fixed usage of ast_format_compatibility_bitfield2format(). The function
can return NULL if the bitfield was not associated with a function.
* Deleted dead code iax2_codec_pref_getsize() and
iax2_codec_pref_setsize().
* Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call
iax2_codec_pref_to_cap() instead of inlining it.
* Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and
IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8.
* Renamed prefs to prefs_global so it won't get confused with the local
pref versions.
* Fixed too small buffer in handle_cli_iax2_show_peer().
* Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete
lines.
* Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an
optimization so iax2_request() and iax2_call() do less work.
* Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when
the pbx could not get started.
* Made set_config() setup a local prefs list along side the local
capability format bitfield. Once the config is loaded, then the local
copies are put into the global versions.
* Fix unininialized codec_buf in function_iaxpeer().
ASTERISK-24150 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/3890/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r420089 | mjordan | 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) | 72 lines
ARI: Add channel technology agnostic out of call text messaging
This patch adds the ability to send and receive text messages from various
technology stacks in Asterisk through ARI. This includes chan_sip (sip),
res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the
endpoints resource, and can be sent directly through that resource, or to a
particular endpoint.
For example, the following would send the message "Hello there" to PJSIP
endpoint alice with a display URI of sip:asterisk@mycooldomain.org:
ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There
This is equivalent to the following as well:
ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There
Both forms are available for message technologies that allow for arbitrary
destinations, such as chan_sip.
Inbound messages can now be received over ARI as well. An ARI application that
subscribes to endpoints will receive messages from those endpoints:
{
"type": "TextMessageReceived",
"timestamp": "2014-07-12T22:53:13.494-0500",
"endpoint": {
"technology": "PJSIP",
"resource": "alice",
"state": "online",
"channel_ids": []
},
"message": {
"from": "\"alice\" <sip:alice@127.0.0.1>",
"to": "pjsip:asterisk@127.0.0.1",
"body": "Watson, come here.",
"variables": []
},
"application": "testsuite"
}
The above was made possible due to some rather major changes in the message
core. This includes (but is not limited to):
- Users of the message API can now register message handlers. A handler has
two callbacks: one to determine if the handler has a destination for the
message, and another to handle it.
- All dialplan functionality of handling a message was moved into a message
handler provided by the message API.
- Messages can now have the technology/endpoint associated with them.
Various other properties are also now more easily accessible.
- A number of ao2 containers that weren't really needed were replaced with
vectors. Iteration over ao2_containers is expensive and pointless when
the lifetime of things is well defined and the number of things is very
small.
res_stasis now has a new file that makes up its structure, messaging. The
messaging functionality implements a message handler, and passes received
messages that match an interested endpoint over to the app for processing.
Note that inadvertently while testing this, I reproduced ASTERISK-23969.
res_pjsip_messaging was incorrectly parsing out the 'to' field, such that
arbitrary SIP URIs mangled the endpoint lookup. This patch includes the
fix for that as well.
Review: https://reviewboard.asterisk.org/r/3726
ASTERISK-23692 #close
Reported by: Matt Jordan
ASTERISK-23969 #close
Reported by: Andrew Nagy
........
r420090 | mjordan | 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) | 2 lines
Remove automerge properties :-(
........
r420097 | mjordan | 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) | 2 lines
test_message: Fix strict-aliasing compilation issue
........
Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes
to the RTCP topics in Stasis and receives RTCP information back from the
message bus. It encodes into HEPv3 packets and sends the information to the
res_hep module for transmission.
Using this, someone with a Homer server can get live call quality monitoring
for all RTP-based channels in their Asterisk 12+ systems.
In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and
chan_pjsip that were uncovered by the tests written for the Asterisk Test
Suite. This patch fixes the following:
1) chan_pjsip failed to set its channel unique ids on its RTP instance on
outbound calls. It now does this in the appropriate location, in the
serialized call callback.
2) The rtp_engine was overflowing some values when packed into JSON.
Specifically, some longs and unsigned ints can't be be packed into integer
values, for obvious reasons. Since libjansson only supports integers,
floats, strings, booleans, and objects, we print these values into strings.
3) res_rtp_asterisk had a few problems:
(a) it would emit a source IP address of 0.0.0.0 if bound to that IP
address. We now use ast_find_ourip to get a better IP address, and
properly marshal the result into an ast_strdupa'd string.
(b) Reports can be generated with no report bodies. In particular, this
occurs when a sender is transmitting information to a receiver (who
will send no RTP back to the sender). As such, the sender has no report
body for what it received. We now properly handle this case, and the
sender will emit SR reports with no body. Likewise, if we receive an
RTCP packet with no report body, we will still generate the appropriate
events.
ASTERISK-24119 #close
........
Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In Asterisk, it is possible for a device to have a status of ONHOLD. This is
not typically an easy thing to determine, as a channel being on hold is not
a direct channel state. Typically, this has to be calculated outside of the
core independently in channel drivers, notably, chan_sip and chan_pjsip. Both
of these channel drivers already have to calculate device state in a fashion
more complex than the core can handle, as they aggregate all state of all
channels associated with a peer/endpoint; they also independently track
whether or not one of those channels is currently on hold and mark the device
state appropriately.
In 12+, we now have the ability to report an AST_DEVICE_ONHOLD state for all
channels that defer their device state to the core. This is due to channel hold
state actually now being tracked on the channel itself. If a channel driver
defers its device state to the core (which many, such as DAHDI, IAX2, and
others do in most situations), the device state core already goes out to get a
channel associated with the device. As such, it can now also factor the channel
hold state in its calculation.
This patch adds this logic to the device state core. It also uses an existing
mapping between device state and channel state to handle more channel states.
chan_pjsip has been updated slightly as well to make use of this (as it was,
for some reason, reporting a channel state of BUSY as a device state of INUSE,
which feels slightly wrong).
Review: https://reviewboard.asterisk.org/r/3771/
ASTERISK-24038 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch serves two purposes:
(1) It fixes some bugs with endpoint subscriptions not reporting all of the
channel events
(2) It serves as the preliminary work needed for ASTERISK-23692, which allows
for sending/receiving arbitrary out of call text messages through ARI in a
technology agnostic fashion.
The messaging functionality described on ASTERISK-23692 requires two things:
(1) The ability to send/receive messages associated with an endpoint. This is
relatively straight forwards with the endpoint core in Asterisk now.
(2) The ability to send/receive messages associated with a technology and an
arbitrary technology defined URI. This is less straight forward, as
endpoints are formed from a tech + resource pair. We don't have a
mechanism to note that a technology that *may* have endpoints exists.
This patch provides such a mechanism, and fixes a few bugs along the way.
The first major bug this patch fixes is the forwarding of channel messages
to their respective endpoints. Prior to this patch, there were two problems:
(1) Channel caching messages weren't forwarded. Thus, the endpoints missed
most of the interesting bits (such as channel creation, destruction, state
changes, etc.)
(2) Channels weren't associated with their endpoint until after creation.
This resulted in endpoints missing the channel creation message, which
limited the usefulness of the subscription in the first place (a major use
case being 'tell me when this endpoint has a channel'). Unfortunately,
this meant another parameter to ast_channel_alloc. Since not all channel
technologies support an ast_endpoint, this patch makes such a call
optional and opts for a new function, ast_channel_alloc_with_endpoint.
When endpoints are created, they will implicitly create a technology endpoint
for their technology (if one does not already exist). A technology endpoint is
special in that it has no state, cannot have channels created for it, cannot
be created explicitly, and cannot be destroyed except on shutdown. It does,
however, have all messages from other endpoints in its technology forwarded to
it.
Combined with the bug fixes, we now have Stasis messages being properly
forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar),
where bar has a single channel associated with it and foo has two channels
associated with it. The messages would be forwarded as follows:
channel PJSIP/foo-1 --
\
--> endpoint PJSIP/foo --
/ \
channel PJSIP/foo-2 -- \
---- > endpoint PJSIP
/
channel PJSIP/bar-1 -----> endpoint PJSIP/bar --
ARI, through the applications resource, can:
- subscribe to endpoint:PJSIP/foo and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo
- subscribe to endpoint:PJSIP/bar and get notifications for channels
PJSIP/bar-1 and endpoint PJSIP/bar
- subscribe to endpoint:PJSIP and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar
Note that since endpoint PJSIP never changes, it never has events itself. It
merely provides an aggregation point for all other endpoints in its technology
(which in turn aggregate all channel messages associated with that endpoint).
This patch also adds endpoints to res_xmpp and chan_motif, because the actual
messaging work will need it (messaging without XMPP is just sad).
Review: https://reviewboard.asterisk.org/r/3760/
ASTERISK-23692
........
Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The iax2_best_codec function was changed to convert the formats
into a format compatibilities structure and grab the first
format from it. The resulting order differs from the previous
order of iax2_best_codec which causes unexpected formats to
get chosen (such as g723).
This commit brings back the old behavior of iax2_best_codec by
having a specified preference list.
Review: https://reviewboard.asterisk.org/r/3835/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After merging the media formats branch, chan_iax2 was discarding
codec preferences for the purpose of choosing which codec a
channel would use once a call started. This patch restores the
Asterisk 1.8-12 codec choice behaviors.
ASTERISK-23958 #close
Review: https://reviewboard.asterisk.org/r/3800/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
progressinband=never in sip.conf is easily defeated if an onward trunk sends a
progress indication of its own. This is almost certain to happen if the onward
trunk is ISDN or IAX as these technologies send a progress indication even if
early media is not required. This progress message is passed to the caller,
and causes the "never" option to be rather badly named.
This patch changes the behaviour of this setting in the following ways:
1) In sip_write(), do not pass the media unless we have either progressed
beyond INV_EARLY_MEDIA, or we are in INV_EARLY_MEDIA state, and early
media is both set-up and wanted. This helps resolve double-ringing on some
buggy handsets.
2) In sip_indicate(), if we see AST_CONTROL_PROGRESS, but
SIP_PROG_INBAND_NEVER is set, send a 180 Ringing instead to avoid implicitly
enabling early media. Avoid sending double ring indications.
NOTE: the meaning of the SIP_PROGRESS_SENT flag changes slightly in this patch
to also encapsulate the fact that a channel has *sent or received* a 183
Progress indication. This makes the updated code in sip_write() much more
simple.
Review: https://reviewboard.asterisk.org/r/3700
ASTERISK-23972 #close
Reported by: Steve Davies
patches:
inband_never_present_early_media2 uploaded by Steve Davies (License 5012)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Most channel drivers let you specify a default accountcode to be set on
channels associated with a particular peer/endpoint/object. Prior to this
patch, chan_pjsip/res_pjsip did not support such a setting.
This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'.
When a channel is created that is associated with an endpoint with this value
set, the channel will automatically have its accountcode property set to the
value configured for the endpoint.
Review: https://reviewboard.asterisk.org/r/3724/
ASTERISK-24000 #close
Reported by: Matt Jordan
........
Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This corrects two issues with the extra field information in Asterisk
12+ in channel event logs.
It is possible to inject custom values into the dialstatus provided by
ast_channel_dial_type() Stasis messages that fall outside the
enumeration allowed for the DIALSTATUS channel variable. CEL now
filters for the allowed values and ignores other values.
The "hangupsource" extra field key is always blank if the far end
channel is a chan_pjsip channel. This is because the hangupsource is
never set for the pjsip channel driver. This change sets the
hangupsource whenever a hangup is queued for chan_pjsip channels.
This corrects an issue with the pjsip channel driver where the
hangupcause information was not being set properly.
Review: https://reviewboard.asterisk.org/r/3690/
........
Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.
Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.
The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.
For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.
And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.
To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.
Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.
We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.
It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.
And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.
Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.
This patch removes:
* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge
It removes the following applications/functions:
* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO
It removes the colon delimiter from the SIPPEER function.
Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.
Review: https://reviewboard.asterisk.org/r/3698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new inband_on_setup_ack option causes Asterisk to assume inband audio
may be present when a SETUP_ACKNOWLEDGE message is received.
Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a
dialtone is sent from the network side, progress indicator 8 "Inband info
now available" MAY be sent to the CPE if no digits were received with the
SETUP. It is thus implied that the ie is mandatory if digits came with
the SETUP and dialtone is needed. This option should be enabled, when the
network sends dialtone and you want to hear it, but the network doesn't
send the progress indicator when needed.
NOTE: For Q.SIG setups this option should be enabled when outgoing overlap
dialing is also enabled because Q.SIG does not send the progress indicator
with the SETUP ACK.
The commit -r413714 (AST-1338) which causes this issue was dealing with a
SIP-to-ISDN interoperability issue.
This commit is a merge of the two patches indicated below.
ASTERISK-23897 #close
Reported by: Pavel Troller
Patches:
pri-4.diff (license #6302) patch uploaded by Pavel Troller
jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/3633/
........
Merged revisions 417956 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 417957 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 417958 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.
This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide
a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation
completes. Configuration options to chan_sip and chan_pjsip have also been added to
allow behavior to be tweaked (such as forcing the AVP type media transports in SDP).
ASTERISK-22961 #close
Reported by: Jay Jideliov
Review: https://reviewboard.asterisk.org/r/3679/
Review: https://reviewboard.asterisk.org/r/3686/
........
Merged revisions 417678 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is essentially a backport of a small portion of r397526 from
ASTERISK-21981. In that patch, pass through support and format attribute
negotiation was added for Opus. Part of that included being more tolerant to
whitespace in the fmtp line of an SDP; that part of the patch is being
applied here.
As the author of the backport pointed out, in SDP, the fmtp line is allowed to
include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001)
includes an example for this. This was not removed in the updated RFC 4867 in
2007.
Review: https://reviewboard.asterisk.org/r/3658
#ASTERISK-23916 #close
Reported by: Alexander Traud
patches:
sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520)
........
Merged revisions 417587 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 417588 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 417589 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a client takes a long time to process information received from Asterisk,
a write operation using fwrite may fail to write all information. This causes
the underlying file stream to be in an unknown state, such that the socket
must be disconnected. Unfortunately, there are two problems with this in
Asterisk's existing websocket code:
1. Periodically, during the read loop, Asterisk must write to the connected
websocket to respond to pings. As such, Asterisk maintains a reference to
the session during the loop. When ast_http_websocket_write fails, it may
cause the session to decrement its ref count, but this in and of itself
does not break the read loop. The read loop's write, on the other hand,
does not break the loop if it fails. This causes the socket to get in a
'stuck' state, preventing the client from reconnecting to the server.
2. More importantly, however, is that the fwrite in ast_http_websocket_write
fails with a large volume of data when the client takes awhile to process
the information. When it does fail, it fails writing only a portion of
the bytes. With some debugging, it was shown that this was failing in a
similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite
with a long enough timeout solved the problem.
Note that this version of the patch, unlike r417310 in Asterisk 11, exposes
configuration options beyond just chan_sip's sip.conf. Configuration options
to configure the write timeout have also been added to pjsip.conf and ari.conf.
#ASTERISK-23917 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3624/
........
Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a DAHDI span disappears, we wish for its representation in Asterisk
to be destroyed as well.
The information about the span's removal may come from several paths:
1. DAHDI sends DAHDI_EVENT_REMOVE on every channel.
2. An extra DAHDI_EVENT_REMOVED is sent on every subsequent call to
DAHDI_GET_EVENT.
3. Every read (including the internal one by libpri on the D-channel)
returns -ENODEV.
Asterisk responsds to DAHDI_EVENT_REMOVE on a channel by destroying it.
Destroying a channel requires holding the channel list lock (iflock).
Destroying a channel that is part of a span requires holding the span's
lock. Destroying a channel from a context that holds the span lock,
while at the same time another channel is destroyed directly, leads to a
deadlock. Solution: don't destroy span while holding the channels list
lock.
Thus changes in this patch:
* Deferring removal of PRI spans in response to events: doomed spans
are collected on a list.
* Doomed spans are removed periodically by the monitor thread.
* ENODEV reads from the D-channel will warant the same deferred removal.
Review: https://reviewboard.asterisk.org/r/3548/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* SS7 support now requires libss7 v2.0 or later. The new libss7 is not
backwards compatible.
* Added SS7 support for connected line and redirecting.
* Most SS7 CLI commands are reworked as well as new SS7 commands added.
See online CLI help.
* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.
* ISUP timer support reworked and now requires explicit configuration.
See ss7.timers.sample.
Special thanks to Kaloyan Kovachev for his support and persistence in
getting the original patch by adomjan updated and ready for release.
SS7-27 #close
Reported by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a framehook is removed - such as the fax gateway framehook - the bridge
framework will re-evaluate the bridge mixing technologies to see if it can
improve the bridging. When this occurs, get_rtp_info will be called to
determine if local or remote bridging can be used. Using remote bridging
will cause a fax to fail, as direct media negotiation will cause some small
number of packets to not arrive at the remote endpoint.
This patch forces local native bridging if T.38 negotiation is in progress or
has been established.
........
Merged revisions 416318 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Simply establishing a TCP connection and never sending anything to the
configured HTTP port in http.conf will tie up a HTTP connection. Since
there is a maximum number of open HTTP sessions allowed at a time you can
block legitimate connections.
A similar problem exists if a HTTP request is started but never finished.
* Added http.conf session_inactivity timer option to close HTTP
connections that aren't doing anything. Defaults to 30000 ms.
* Removed the undocumented manager.conf block-sockets option. It
interferes with TCP/TLS inactivity timeouts.
* AMI and SIP TLS connections now have better authentication timeout
protection. Though I didn't remove the bizzare TLS timeout polling code
from chan_sip.
* chan_sip can now handle SSL certificate renegotiations in the middle of
a session. It couldn't do that before because the socket was non-blocking
and the SSL calls were not restarted as documented by the OpenSSL
documentation.
* Fixed an off nominal leak of the ssl struct in
handle_tcptls_connection() if the FILE stream failed to open and the SSL
certificate negotiations failed.
The patch creates a custom FILE stream handler to give the created FILE
streams inactivity timeout and timeout after a specific moment in time
capability. This approach eliminates the need for code using the FILE
stream to be redesigned to deal with the timeouts.
This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of
the SSL_read/SSL_write operations.
ASTERISK-23673 #close
Reported by: Richard Mudgett
........
Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
* A number of chatty verbose messages were removed or demoted to DEBUG
messages. Verbose messages with a verbosity level of 5 or higher were -
if kept as verbose messages - demoted to level 4. Several messages
that were emitted at verbose level 3 were demoted to 4, as announcement
of dialplan applications being executed occur at level 3 (and so the
effects of those applications should generally be less).
* Some verbose messages that only appear when their respective 'debug'
options are enabled were bumped up to always be displayed.
* Prefix/timestamping of verbose messages were moved to the verboser
handlers. This was done to prevent duplication of prefixes when the
timestamp option (-T) is used with the CLI.
* Verbose magic is removed from messages before being emitted to
non-verboser handlers. This prevents the magic in multi-line verbose
messages (such as SIP debug traces or the output of DumpChan) from
being written to files.
* _Slightly_ better support for the "light background" option (-W) was
added. This includes using ast_term_quit in the output of XML
documentation help, as well as changing the "Asterisk Ready" prompt to
bright green on the default background (which stands a better chance of
being displayed properly than bright white).
Review: https://reviewboard.asterisk.org/r/3547/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk started counting the session timer at INVITE while the other
end correctly started at 200. This meant that for short session-expiries
(90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk
would wrongly assume that the timer was hit before the other end thought
it was time to send a session refresh. This resulted in prematurely
ended calls.
This changes the session timer to start counting first at 200 like RFC
says it should.
(Also removed a few excess NULL checks that would never hit, because if
they did, asterisk would have crashed already.)
ASTERISK-22551 #close
Reported by: i2045
Review: https://reviewboard.asterisk.org/r/3562/
........
Merged revisions 414620 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 414628 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 414636 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When overlap dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an interoperability
problem with SIP. sig_pri doesn't know if there is dialtone present when
a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an
AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183
Session Progress and blocks the desired 180 Ringing message when the
ALERTING message comes in.
* Made the configure script detect if the installed version of libpri
supports the SETUP_ACKNOWLEDGE enhancements.
* Using the new API, made generate an AST_CONTROL_PROGRESS frame on an
incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio
is present instead of assuming that dialtone is present.
* Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio
available indication only if dialtone is expected. The change also makes
the fallback behaviour of sending the PROGRESS message better by sending
it only if dialtone is expected.
* Changed receiving a PROCEEDING message to not generate an
AST_CONTROL_PROGRESS frame if the progress indication ie indicates
non-end-to-end-ISDN. This helps interoperability with SIP.
* Changed sending a PROCEEDING message in response to an
AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It
was silly to do so anyway because the channel driver doesn't know if
inband audio is even available. This helps interoperability with SIP.
This patch and a corresponding change in libpri work together to allow
Asterisk to control the inband audio available progress indication ie on
the SETUP_ACKNOWLEDGE message when dialtone is present.
AST-1338 #close
Reported by: Tyler Stewart
Review: https://reviewboard.asterisk.org/r/3521/
........
Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 413765 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 413771 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Per rfc3892, the Referred-By header in a REFER must be copied into the
referenced request (IE. The outgoing INVITE to the transfer target).
* Automatically put the Referred-By header in the outgoing INVITE message
if the SIPREFERREDBYHDR channel variable is defined with a value.
* Made chan_sip.c:get_refer_info() set SIPREFERREDBYHDR for inheritance so
chan_pjsip has a better chance to interoperate.
* Fixed refer_blind_callback() and refer_incoming_refer_request() to not
modify the data in the pointer returned by pjsip_msg_find_hdr_by_name().
It seems wrong to modify that data since the calling routine doesn't own
the buffer.
ASTERISK-23501 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/3514/
........
Merged revisions 413210 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed early exit in sip_msg_send() not destroying the message iterator.
* Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
tolerant of a NULL iter parameter in case ast_msg_var_iterator_init()
fails.
* Made ast_msg_var_iterator_destroy() clean up any current message data
ref.
* Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(),
ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and
ast_msg_var_iterator_destroy() use iter instead of i.
* Eliminated RAII_VAR usage in res_pjsip_messaging.c:vars_to_headers().
........
Merged revisions 413139 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 413142 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a task was in-flight which required the channel or bridge lock
it was possible for the synchronous task retrieving the call-id
to deadlock as it holds those locks.
After discussing with Mark Michelson the synchronous task was
removed and the call-id accessed directly. This should be safe as
each object involved is guaranteed to exist and the call-id will
never change.
........
Merged revisions 413140 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) Added the unistim.conf variable dtmf_duration which can select the DTMF playback duration from 0ms to 150ms (0 is off and is the new default)
2) Enabled the transmission of month names, which are sent with the date and changed the dateformat variable to accept the values 0-3 as per the UNISTIM standard (2 & 3 match the previous 1 & 2 formats).
3) Enabled the "Mute" packet so muting microphone works as expected and microphone muted for all calls while LED light on
4) Changed Duree to Timer on i2004 display
(closes issue ASTERISK-23592)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r411189, some behavior was changed which made sendrpid behavior
act in a more trusting manner by sending full user data for peers
set with private caller presence in P-Asserted-Identity headers.
Since this changed long time expected behaviors, we decided to pull
that patch when that was pointed out by the community. Instead, this
patch provides a trust_id_outbound setting which will expose the data
per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
at all if set to 'no'. By default trust_id_outbound will be set to
'legacy' which will preserve the behavior prior to these patches.
Extra special thanks to Walter Doekes for providing advice and
feedback.
(closes issue AST-1301)
(closes issue ASTERISK-19465)
Reported by: Krzysztof Chmielewski
Review: https://reviewboard.asterisk.org/r/3447/
........
Merged revisions 412744 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 412746 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 412747 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is a continuation of https://reviewboard.asterisk.org/r/3349/,
committed in r412303.
It resolves a finding oej had that the phone-context be available in a
channel variable separate from SIPDOMAIN. This patch adds that variable as
SIPURIPHONECONTEXT. It also allows a local number (or global number specified
in the TEL URI) to be used to look up as a peer.
(issue ASTERISK-17179)
Review: https://reviewboard.asterisk.org/r/3349/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Remove unused RAII_VAR() declarations. The compiler cannot catch these
because the cleanup function "references" the unused variable. Some
actually allocated and released resources that were never used.
* Fixed some whitespace issues in stasis_bridges.c.
........
Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The failing assertion ensures that the final snapshot gets generated so
CDR records can get finalized. The only place where a channel staging
snapshot flag could be left set is in chan_sip.c:handle_request_bye().
The function could return before clearing the flag because the channel
could dissappear while the function had to have the channel unlocked.
* Fixed handle_request_bye() channel snapshot staging coverage area to not
have a return in the middle of it and be unable to clear the staging flag.
* Pushed the channel snapshot staging coverage area into
ast_rtp_instance_set_stats_vars() to ensure that the staging is not
interrutped.
* Made callers of ast_rtp_instance_set_stats_vars() not call it with any
channels or channel driver private locks held to eliminate the deadlock
potential. The callers must hold references to the passed in channel and
rtp objects.
* Eliminated sip_hangup() trying to get the bridge peer. It is futile at
this point because the channel could never be in a bridge.
Review: https://reviewboard.asterisk.org/r/3431/
........
Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines
chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)
Prior to this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
---
........
Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 412329 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for handling TEL URIs in inbound INVITE requests.
This includes the Request URI and the From URI. The number specified in
the Request URI will be the destination of the inbound channel in the dialplan.
The phone-context specified in the Request URI will be stored in the
TELPHONECONTEXT channel variable.
Review: https://reviewboard.asterisk.org/r/3349
ASTERISK-17179 #close
Reported by: Geert Van Pamel
Tested by: Geert Van Pamel
patches:
asterisk-12.0.0-chan_sip-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)
asterisk-12.0.0-reqresp_parser-RFC3966_patch.txt uploaded by Geert Van Pamel (License 6140)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
Every run will now blow away the previous run (as large ref files
sometimes caused issues). We now also no longer open/close the file
on each write, instead relying on fflush to make sure data gets written
to the file (in case the ao2 call being performed is about to cause a
crash)
(3) It goes with a comma delineated format for the ref debug file. This
makes parsing much easier. This also now includes the thread ID of the
thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
contrib/scripts folder.
(5) The old refcounter implementation in utils/ has been removed.
Review: https://reviewboard.asterisk.org/r/3377/
........
Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 412115 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 412153 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The masquerade supertest frequently fails because either the local channel
chain doesn't completely optimize out or the DTMF handshake doesn't
completely get accross. Local channel optimization requires frames
flowing to trigger when optimization can happen. When optimization
happens the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for timing
purposes while sending nothing. If internal timing is not enabled when
MOH is playing, Asterisk switches to received timing when an audio frame
is received. With optimization dropping media frames and MOH not sending
frames unless it receives frames, occasionaly there are no more frames
being passed and the test fails.
* The asterisk command line -I option and the asterisk.conf
internal_timing option are removed. Asterisk now always uses internal
timing when needed if any timing module is loaded. The issue
ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken
if other internal timing modules besides DAHDI are used. The
ast_read_generator_actions() now only does received timing if it has no
choice for frame generators like MOH, silence, and playback streaming.
* Cleaned up some code dealing with frame generators in
ast_deactivate_generator(), generator_write_format_change(),
ast_activate_generator(), and ast_channel_stop_silence_generator().
* Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and
ast_opt_internal_timing.
ASTERISK-22846 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3414/
........
Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock, then
pvt->provisional_keepalive_sched_id will be changed to a new sched_id
value by update_provisional_keepalive(), but that new sched_id then may
be overwritten with -1 by send_provisional_keepalive_full(), killing
the pvt's reference to a schedule and "leaking" the reference.
(closes issue ASTERISK-22079)
Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches:
provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012)
........
Merged revisions 411088 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 411089 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 411091 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
........
Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When accidentally compiling against a wrong version of
pjsip headers with a different pjsip_inv_session size,
the invite_tsx structure could be null in the answer()
function. This led to a crash because it attempted to
send the session response with an uninitialized packet
pointer. This patch presets packet to null and adds a
diagnostic log message to explain why the call fails.
Review: https://reviewboard.asterisk.org/r/3267/
........
Merged revisions 408970 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling
ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the
NOTIFY request that informs the transferor if the transfer succeeded or failed.
This patch also clears said flag from the off nominal NOTIFY paths in the
local_attended_transfer code, as once we've sent the NOTIFY request it is safe
to send by the BYE request.
This was caught by the blind-transfer-accountcode test in the Asterisk Test
Suite.
(closes issue ASTERISK-23290)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3214/
........
Merged revisions 408069 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Move route code to sip/route.c + sip/include/route.h
* Rename functions to sip_route_*
* Replace ad-hoc list code with macro's from linkedlists.h
* Create sip_route_process_header() to processes Path and Record-Route headers
(previously done with different code in build_route and build_path)
* Add use of const where possible
* Move struct uriparams, struct contact and contactliststruct from sip.h to
reqresp_parser.h. sip/route.c uses reqresp_parser.h but not sip.h, this was
a problem. These moved declares are not used outside of reqresp_parser.
* While modifying reqprep() the lack of {} caused me trouble. I added them.
* Code outside route.c treats sip_route as an opaque structure, using macro's
or procedures for all access.
(closes issue ASTERISK-22582)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3173/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
(closes issue AST-1302)
Review: https://reviewboard.asterisk.org/r/3174/
........
Merged revisions 407678 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 407727 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 407729 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Locking issues in skinny when picking up a call that doesn't exist. Cleaned
up sub locking by fully removing and using the chan lock instead. Also
changed ast_call_pickup to check whether chan was masq'd.
(closes issue ASTERISK-23249)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-locking01.diff uploaded by wedhorn (license 5019)
........
Merged revisions 407197 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
........
Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.
Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.
While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.
(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
........
Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
dahdi show channels output slices the callerid (which is dnid copied over on
PRI channels). If the channel naming structures look like:
'DAHDI/i1/1408409XXXX-6'
then the output slices 1408409XXXX down to 1408409XXX. This patch just opens
it up to 15 chars so you can see the whole thing.
(closes issue ASTERISK-22918)
Reported by: outtolunc
Patches:
svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198)
........
Merged revisions 404784 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 404785 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 404786 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channel creation in Asterisk is broken up into two steps: requesting and calling.
In some cases a channel may be requested but never called. This happens in the
ChanIsAvail dialplan application for determining if something is reachable or
not. The PJSIP channel driver did not take this situation into account and
attempted to end a session that was never called out on.
The code now checks the session state to determine if the session has been
called out on and if not terminates it instead of ending it.
(closes issue ASTERISK-23074)
Reported by: Kilburn
........
Merged revisions 404652 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If ignore_failed_channels is set to "true" for a channel, the channel
will continue to be configured even if configuring it has failed.
This allows Asterisk to start before all the DAHDI initialization is
done and thus not force the starting order dahdi -> asterisk.
Review: https://reviewboard.asterisk.org/r/3063/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
........
Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Moved channel locking into setsubstate so that a process can complete
working on a sub before another starts changing it. The existing code
was causing some Fracks with schedule deletion.
Removed multiple rtp cleanup. Now only cleansup up once, fixing ao2
object cleanup issues.
........
Merged revisions 404306 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On session registration, if device is already reporting that it is
connected to a device, an innocuous packet (update time) is sent to
the already connected device. If the tcp connection is down, the
device will be unregistered and the new connection allowed.
Without this patch, network issues can see a situation where a device
can not reregister until after 3*timeout.
........
Merged revisions 404292 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
........
Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
........
Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch was intended to eliminate a deadlock that occurs when
masquerades occur in pjsip channels, but has some potential side
effects. Mark Michelson is currently working on addressing this
problem from another angle.
(issue ASTERISK-22936)
Reported by: Jonathan Rose
........
Merged revisions 403705 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
........
Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
........
Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.
(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
........
Merged revisions 402708 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 402709 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 402710 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal. That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...". This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.
(closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek
Review: https://reviewboard.asterisk.org/r/2988/
........
Merged revisions 402604 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 402605 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 402606 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This started off as a fix for the failing IAX2 acl_call test in the Asterisk
Test Suite. When inspecting why that test was failing, it became clear that all
attempts to bind to any local loopback address was failing:
[Nov 2 15:56:28] VERBOSE[15787] chan_iax2.c: == Binding IAX2 to address
127.0.0.1:4569
[Nov 2 15:56:28] DEBUG[15787] netsock2.c: Splitting '127.0.0.1' into...
[Nov 2 15:56:28] DEBUG[15787] netsock2.c: ...host '127.0.0.1' and port ''.
[Nov 2 15:56:28] ERROR[15787] netsock2.c: getaddrinfo("127.0.0.1", "(null)",
...): ai_family not supported
[Nov 2 15:56:28] WARNING[15787] acl.c: Unable to lookup '127.0.0.1'
While there's conceivably other ways for getaddrino to return EAI_FAMILY, the
most common way is if AF_INET, AF_INET6, or AF_UNSPEC is not provided as the
desired family. The culprit was the call to ast_get_ip, defined in acl.h. This
function uses the family from the passed in addr object (which it will also
populate when it returns!) when it eventually calls getaddrinfo.
This patch fixes the use of ast_get_ip that were not specifying the family in
chan_iax2. This prevents uninitialized use of the structure, so that the
addresses resolve correctly.
Review: https://reviewboard.asterisk.org/r/2991
........
Merged revisions 402505 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing. The display of
"N" used to mean NAT (i.e. yes). The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear. That was a great suggestion.
Therefore, this patch does the following:
* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.
* A column for the 'Comedia' setting has been added. It too will display the
setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.
* UPGRADE.txt has been updated to document this change.
(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
asterisk-forcerport-display-clarification_v3.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2941
........
Merged revisions 402111 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 402112 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401704 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 401705 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 401706 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When reworking chan_iax2 for IPv6, the ability to bind to multiple addresses
was removed by mistake. This patch restores this functionality and adds notes
about IPv6 addresses in the sample config.
(closes issue ASTERISK-22741)
Reported by: Joshua Colp
Tested by: Michael L. Young
Patches:
asterisk-22741-fix-binding-multiple-addr.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2945/
........
Merged revisions 401488 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
........
Merged revisions 401167 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 401168 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401169 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
........
Merged revisions 400970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 400971 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 400984 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
........
Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r400318 | mmichelson | 2013-10-02 17:08:49 -0500 (Wed, 02 Oct 2013) | 12 lines
Remove unnecessary waits from stasis.
Since caches are updated on publisher threads, there is no need
to wait for the cache updates to occur after a stasis message
is published.
In the case of chan_pjsip device state changes, this set of
changes caused an improvement to performance.
Review: https://reviewboard.asterisk.org/r/2890
........
r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, 02 Oct 2013) | 3 lines
Remove svn:mergeinfo property.
........
Merged revisions 400318-400319 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
Minor performance bump by not allocate manager variable struct if we don't need it
........
r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
Stasis performance improvements
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
........
r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
Taskprocessor optimization; switch Stasis to use taskprocessors
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
........
r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
Optimize how Stasis forwards are dispatched
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
........
r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
Remove dispatch object allocation from Stasis publishing
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
........
Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk receives a 200 OK in response to an invite, that peer should have
sent an SDP at some point by then. If the channel has never received an SDP,
media won't have been set and the remote address won't be known. Endpoints in
general should not be doing this. This patch makes it so that Asterisk will
simply hang up a call if it sends a 200 OK at this point. So far this odd
behavior for endpoints has only been observed in tests which involved manually
created SIP transactions in SIPp.
(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/
........
Merged revisions 399939 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 399962 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 399976 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The PRI and SS7 link control threads are not stopped correctly when the
chan_dahdi.so module is unloaded. The link control threads pri_dchannel()
and ss7_linkset() are not awakened from a poll() to cancel the thread.
* Added a SIGURG signal after requesting the thread cancel to break the
link control thread poll() immediately.
For SS7 it was slightly worse, the link poll() timeout would always be
whatever was the last libss7 scheduled event time used. If no libss7
scheduled event was pending, the thread could run more often than
necessary.
* Set nextms to 60 seconds for the ss7_linkset() poll() if there is no
other libss7 scheduled event.
........
Merged revisions 399818 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 399834 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 399842 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1st Issue
When a realtime peer sends an un-REGISTER request, Asterisk
un-registers the peer but the database table record still has regseconds and
fullcontact for the peer. This results in calls attempting to be routed to the
peer which is no longer registered. The expected behavior is to get
busy/congested when attempting to call an un-registered peer through the
dialplan.
What was discovered is that we are clearing out the peer's registration in the
database in parse_register_contact() when calling expire_register() but then
upon returning from parse_register_contact(), update_peer() is run which stores
back in the database table regseconds and fullcontact.
2nd Issue
The reporter pointed out that the 200 ok being returned by Asterisk
after un-registering a peer contains a Contact header with ;expires= and the
Expires header is not set to 0. This is actually a regression.
Tests were created for this second issue (ASTERISK-22548). The tests have been
reviewed and a Ship It! was received on those tests.
This patch does the following:
* Do not ignore the Expires header value even when it is set to 0. The patch
sets the pvt->expiry earlier on in the function so that it is set properly and
used.
* If pvt->expiry is 0, do not call update_peer since that means the peer has
already been un-registered and there is no need to update the database record
again since nothing has changed.
(closes issue ASTERISK-22428)
Reported by: Ben Smithurst
Tested by: Ben Smithurst, Michael L. Young
Patches:
asterisk-22428-rt-peer-update-and-expires-header.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2869/
........
Merged revisions 399794 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 399795 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 399796 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a new IAX2 client registers, the astdb database is updated with the
value of minregexpire defined in iax.conf instead of using the expiry time
that is provided by the client. The provided expiry time of the client is
updated after inserting the astdb entry. As a consequence, restarting or
reloading asterisk creates clients whose registration may expire before
they reregister. The clients are therefore unavailable after minregexpire
seconds until they reregister.
* Move updating of the expiry time to before inserting into the astdb.
(closes issue ASTERISK-22504)
Reported by: Stefan Wachtler
Patches:
chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler
........
Merged revisions 399158 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 399159 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 399160 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
........
Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:
1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.
Also added an SDP when an update is sent out.
(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
........
Merged revisions 398806 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix bridgecallno deadlock avoidance. When doing deadlock avoidance, you
need to retest the status of values for each loop to see if you still need
the lock for bridgecallno.
* As a safety check, after acquiring the bridgecallno lock you should
check if iaxs[bridgecallno] is NULL just like the current callno checks.
* Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any
deferred frames to ensure that the iostate is IDLE when it is placed back
into the idle list. defer_full_frame() tries to ensure
iax2_process_thread() wakes up to process the frame.
........
Merged revisions 398379 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 398380 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 398381 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix the misdn debug output to remote consoles. chan_misdn uses
ast_console_puts() which doesn't know about verbose levels. Better to use
ast_verbose() instead. Without this patch the misdn debug messages are
appended to the verbose level which ever was set by the message sent to
the console before, i.e. any undefined level.
(closes issue AST-1218)
Reported by: Guenther Kelleter
Patches:
misdnlog.patch (license #6372) patch uploaded by Guenther Kelleter
........
Merged revisions 398235 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 398236 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 398237 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].
This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.
For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.
Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)
The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.
Other changes made as a part of this patch:
* The stubs for http_websocket that wrap system calls set errno to
ENOSYS.
* res_http_websocket now properly increments module use count.
* In loader.c, the while() wrappers around dlclose() were removed. The
while(!dlclose()) is actually an anti-pattern, which can lead to
infinite loops if the module you're attempting to unload exports a
symbol that was directly linked to.
* The special handling of nonoptreq on systems without weak symbol
support was removed, since we no longer rely on weak symbols for
optional_api.
[1]: https://wiki.asterisk.org/wiki/x/wACUAQ
(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
........
Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Refactored cases where a combination of ast_verbose/options_verbose were
present. Also in general tried to eliminate, in as many places as possible,
where the options_verbose global variable was being used. Refactored the way
local and remote consoles handle verbose message logging in an attempt to
solve the various discrepancies that sometimes would show between the two.
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2798/
........
Merged revisions 397948 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 397958 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the SIP channel driver processes an invalid SDP that defines media
descriptions before connection information, it may attempt to reference
the socket address information even though that information has not yet
been set. This will cause a crash.
This patch adds checks when handling the various media descriptions that
ensures the media descriptions are handled only if we have connection
information suitable for that media.
Thanks to Walter Doekes, OSSO B.V., for reporting, testing, and providing
the solution to this problem.
(closes issue ASTERISK-22007)
Reported by: wdoekes
Tested by: wdoekes
patches:
issueA22007_sdp_without_c_death.patch uploaded by wdoekes (License 5674)
........
Merged revisions 397756 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 397757 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 397758 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 397759 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A remote exploitable crash vulnerability exists in the SIP channel driver if an
ACK with SDP is received after the channel has been terminated. The handling
code incorrectly assumed that the channel would always be present.
This patch adds a check such that the SDP will only be parsed and applied if
Asterisk has a channel present that is associated with the dialog.
Note that the patch being applied was modified only slightly from the patch
provided by Walter Doekes of OSSO B.V.
(closes issue ASTERISK-21064)
Reported by: Colin Cuthbertson
Tested by: wdoekes, Colin Cutherbertson
patches:
issueA21064_fix.patch uploaded by wdoekes (License 5674)
........
Merged revisions 397710 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 397711 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 397712 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 397713 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397753 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch
in memory pools but had a math error determining the buffer size and
didn't address other similar memory pool mismatches.
* Effectively reverted the previous patch to go in the same direction as
trunk for the returned memory pool of ast_bt_get_symbols().
* Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is
defined.
* Fixed some formatting in ast_bt_get_symbols().
* Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is
enabled.
* Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled.
* Moved __dump_backtrace() because of compile issues with the utils
directory.
(closes issue ASTERISK-22221)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2778/
........
Merged revisions 397525 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 397528 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a response to an initial incoming INVITE results in a transport error
the INVITE transaction is removed from the INVITE session. Any attempts
to answer the INVITE session after this results in a crash as it requires
the INVITE transaction to exist. This change explicitly locks the dialog
and checks to ensure that the INVITE transaction exists before answering.
(closes issue AST-1203)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a From header on an outbound out-of-call SIP MESSAGE were
malformed, the result could crash Asterisk.
In addition, if a From header on an incoming out-of-call SIP
MESSAGE request were malformed, the message was happily accepted
rather than being rejected up front. The incoming message path
would not result in a crash, but the behavior was bad nonetheless.
(closes issue ASTERISK-22185)
reported by Zhang Lei
........
Merged revisions 397254 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The SIP_CODEC family of variables let you set the preferred codec to be
offered on an outbound INVITE request. However, for video calls, you need to
be able to set both the audio and video codecs to be offered. This patch lets
the SIP_CODEC variables accept a comma delineated list of codecs. The first
codec in the list is set as the preferred codec; additional codecs are still
offered however.
This lets a dialplan writer set both audio and video codecs, e.g.,
Set(SIP_CODEC=ulaw,h264)
Note that this feature was written by both Dennis Guse and Frank Haase
Review: https://reviewboard.asterisk.org/r/2728
(closes issue ASTERISK-21976)
Reported by: Denis Guse
Tested by: mjordan, sysreq
patches:
patch-channels-chan__sip.c-393919 uploaded by dennis.guse (license 6513)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In 1.8, r384779 introduced a regression by retrieving an old dialog and keeping
the old recv address since recv was already set. This has caused a problem when
a proxy is involved since responses to incoming requests from the proxy server,
after an outbound call is established, are never sent to the correct recv
address.
In 11, r382322 introduced this regression.
The fix is to revert that change and always store the recv address on incoming
requests.
Thank you Walter Doekes for helping to point out this error and Mark Michelson
for your input/review of the fix.
(closes issue ASTERISK-22071)
Reported by: Alex Zarubin
Tested by: Alex Zarubin, Karsten Wemheuer
Patches:
asterisk-22071-store-recvd-address.diff by Michael L. Young (license 5026)
........
Merged revisions 397204 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 397205 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.
This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.
Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This makes it so that we can detect failures to originate as with
earlier versions of Asterisk, which restores the Asterisk 11 behavior
for the originate manager action. This was causing the ACL tests for
SIP and IAX2 to fail since those tests expected originate failures
when ACLs would cause rejections. Also, this patch fixes crashes in
chan_sip when ACLs rejected peers during registration verification.
(closes issue ASTERISK-22212)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2753/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code adds chan_dahdi the command 'dahdi create channels <range>'
(where <range> is a single <n>-<m> or 'new') and updates 'dahdi destroy
channel' with a similar 'dahdi destroy channels'. It allows DAHDI
channels and spans to be added after the initial channel load
(without destroying all other channels as in 'dahdi restart').
It also includes some fixes to the D-Channel / span destruction code
(r394552).
This change is intended to provide a hook for a script running from
udev once a span has been assigned ("registered") / unassigned
("unregistered") for its channels. The udev hook configures the span's
channels with dahdi_cfg -S, and can then ask Asterisk to create ethe
channels. See the scripts added to DAHDI-tools in 2.7.0.
Review: https://reviewboard.asterisk.org/r/1598/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does three things:
1. It provides a Surrogate channel technology with a consolidated
"implementation detail flag" on the channel technology. This tells
consumers of Stasis that the creation of this channel is an implementation
detail in Asterisk and can be ignored (if they so choose). This
consolidates the conference recorder/announcer flags as well - these flags
had no additional meaning beyond "ignore this channel please".
2. It modifies allocation of a channel in two ways:
(a) If a channel technology can be determined from the name, we set it
directly in the allocation routine. This prevents the initial
publication of the message from going out with a NULL channel technology
where possible. This lets Stasis consumers get the right channel
technology on the first publication.
(b) It reorganizes allocation to make use of the 'finalized' property on the
channel. This was already used to know that a channel had completely
finished its construction in the masquerade routine; now we also use it
to know whether or not the setting of certain channel properties is
occurring during or post construction. The various set routines were
modified accordingly as well.
3. The masquerade event is now dead, Jim. It no longer served any purpose
whatsoever - if you perform a call pickup you'll get a Pickup event;
if you perform an attended transfer you will still get those events; if you
steal a channel to put it elsewhere you'll get the corresponding NewExten or
BridgeEnter events.
Review: https://reviewboard.asterisk.org/r/2740
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a peer is used in a register line and TLS is defined as the transport, the
registration fails since the transport on the dialog is never set properly
resulting in UDP being used instead of TLS.
This patch sets the dialog's transport based on the transport that was defined
in the register line. If the register line does not specify a transport, the
parsing function for the register line always defaults back to UDP.
(closes issue ASTERISK-21964)
Reported by: Doug Bailey
Tested by: Doug Bailey
Patches:
asterisk-21964-set-reg-dialog-transport.diff
by Michael L. Young (license 5026)
........
Merged revisions 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 396248 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.
(closes issue ASTERISK-22039)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2717
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It moves the pickup code out of features.c and into pickup.c
* It removes the vast majority of dead code out of features.c. In particular,
this includes the parking code.
(issue ASTERISK-22134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds support for externally initiated parking requests. In particular,
chan_skinny has a protocol level message that initiates a call park.
This patch now supports that option, as well as the protocol specific
mechanisms in chan_dahdi/sig_analog and chan_mgcp.
* A parking bridge features virtual table has been added that provides
access to the parking functionality that the Bridging API needs. This
includes requests to park an entire 'call' (with little or no additional
information, thank you chan_skinny), perform a blind transfer to a parking
extension, determine if an extension is a parking extension, as well as the
actual "do the parking" request from the Bridging API.
* Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
functions
* The removal of some - but not all - dead parking code from features.c
This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)
Review: https://reviewboard.asterisk.org/r/2710
(closes issue ASTERISK-22134)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.
Review: https://reviewboard.asterisk.org/r/2708/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch makes it so that all registration attempts that fail that
also permanently modify the registration state will raise an appropriate
AMI event.
Note that this patch was forward ported to trunk and the Stasis Core
message bus by mjordan.
(closes issue ASTERISK-21368)
Reported by: Dmitriy Serov
patches:
chan_sip.c.diff uploaded by Demon (license 6479)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created a native_dahdi bridging technology for use with the new bridging
API.
The new bridging technology is part of the chan_dahdi channel driver
because it is very specific to that driver. Rather than include the new
code directly into chan_dahdi.c the new bridge technology is in its own
file and linked into chan_dahdi.so. A large part of this change is the
mechanical process of moving declarations around so chan_dahdi.c can be
split up into more files later.
* Changed the bridging core to pass NULL frames into the channel
technologies instead of discarding them. The channel technologies may
need the proding to determine if their configuration is still valid.
(closes issue ASTERISK-21886)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2681/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch introduces DTLS-SRTP support to chan_pjsip and the options
necessary to configure it including an option to allow choosing between
32 and 80 byte SRTP tag lengths.
During the implementation and testing of this patch, three other bugs
were found and their fixes are included with this patch. The two in
chan_sip were a segfault relating to DTLS setup and mistaken call
rejection. The third bug fix prevents chan_pjsip from attempting to
perform bridge optimization between two endpoints if either of them is
running any form of SRTP.
Review: https://reviewboard.asterisk.org/r/2683/
(closes issue ASTERISK-21419)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds some of the more recent control frame types to the IAX2
parser. When IAX2 debugging is enabled, it will now show more of the control
frame types.
(closes issue ASTERISK-22120)
Reported by: Birger "WIMPy" Harzenetter
patches:
iaxcmds.diff uploaded by wimpy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.
This adds tests for blind transfers, several types of attended
transfers, and call pickup.
The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.
Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a new channel driver for creating channels for specific purposes
in bridges, primarily to act as either recorders or announcers. Adds
ARI commands for playing announcements to ever participant in a bridge
as well as for recording a bridge. This patch also includes some
documentation/reponse fixes to related ARI models such as playback
controls.
(closes issue ASTERISK-21592)
Reported by: Matt Jordan
(closes issue ASTERISK-21593)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2670/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Temporary peers do not have an associated Stasis endpoint and quite a
bit of code in chan_sip assumes that all peers have a Stasis endpoint.
All endpoint accesses in chan_sip are now wrapped in an endpoint
NULL-check.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.
This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.
This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).
It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.
Review: https://reviewboard.asterisk.org/r/726/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2657/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Instead of formatting a NULL pointer into a "%s" format string (which is
usually not a good thing to do), we instead print "Unknown".
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
from mixing different variants or general MFC-R2 settings within the same E1 line.
Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or
the whole E1 has the same country variant and R2 settings.
In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1.
This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every
time a new channel => section is found and the configuration was changed.
(closes issue ASTERISK-21117)
Reported by: Rafael Angulo
Related Elastix issue: http://bugs.elastix.org/view.php?id=1612
........
Merged revisions 394106 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 394173 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Major changes are to pull all packet reading functions into skinny_session
and move timeout handling to scheduling arrangements. Thread cancelling is
now undertaken directly rather than waiting for the read to timeout
(cleanup is popped on thread cancel). Also added some keepalive timings in
debugging messages.
Keepalive timeout has been increased from 1.1 by keepalive to 3 times
keepalive. This seems to align (after keepalives stabilise) with when
devices reset after not receiving keepalives. Probably needs more work,
especially around the first and/or second keepalives that vary
significantly by device and firmware version.
Review: https://reviewboard.asterisk.org/r/2611/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
information in the RTCP events. Because Stasis provides a cache, Jaco's
patch was modified to pass the channel uniqueid to the RTP layer as
opposed to a pointer to the channel. This has the following benefits:
(1) It keeps the RTP engine 'clean' of references back to channels
(2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
Potentially, other implementations (such as res_rtp_multicast) could also
raise RTCP information. The engine provides structs to represent RTCP headers
and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
raise an event when we sent a RR report.
Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.
Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.
Review: https://reviewboard.asterisk.org/r/2603/
(closes issue ASTERISK-20574)
Reported by: Jaco Kroon
patches:
asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)
(closes issue ASTERISK-21471)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was a misunderstanding about ast_bridge_impart()'s handling of the
imparted channel's reference. The channel reference is passed by the
caller unless ast_bridge_impart() returns an error.
* Fixed a memory leak in conf_announce_channel_push() if the impart
failed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes the following memory leaks:
* http.c: The structure containing the addresses to bind to was not being
deallocated when no longer used
* named_acl.c: The global configuration information was not disposed of
* config_options.c: An invalid read was occurring for certain option types.
* res_calendar.c: The loaded calendars on module unload were not being
properly disposed of.
* chan_motif.c: The format capabilities needed to be disposed of on module
unload. In addition, this now specifies the default options for the
maxpayloads and maxicecandidates in such a way that it doesn't cause the
invalid read in config_options.c to occur.
(issue ASTERISK-21906)
Reported by: John Hardin
patches:
http.patch uploaded by jhardin (license 6512)
named_acl.patch uploaded by jhardin (license 6512)
config_options.patch uploaded by jhardin (license 6512)
res_calendar.patch uploaded by jhardin (license 6512)
chan_motif.patch uploaded by jhardin (license 6512)
........
Merged revisions 392810 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1. Security events
2. Websocket support
3. Diversion header + redirecting support
4. An anonymous endpoint identifier
5. Inbound extension state subscription support
6. PIDF notify generation
7. One touch recording support (special thanks Sean Bright!)
8. Blind and attended transfer support
9. Automatic inbound registration expiration
10. SRTP support
11. Media offer control dialplan function
12. Connected line support
13. SendText() support
14. Qualify support
15. Inband DTMF detection
16. Call and pickup groups
17. Messaging support
Thanks everyone!
Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
This means CDRs track well with what an actual channel is doing - which
is useful in transfer scenarios (which were previously difficult to pin
down). It does, however, mean that CDRs cannot be 'fooled'. Previous
behavior in Asterisk allowed for CDR applications, channels, and other
properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
be what everyone wants, but it is a defined behavior and as such, it is
predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
changes have been made to ResetCDR and ForkCDR in particular. Many of the
options for these two applications no longer made any sense with the new
framework and the (slightly) more immutable nature of CDRs.
There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.
(closes issue ASTERISK-21196)
Review: https://reviewboard.asterisk.org/r/2486/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1). When touching the bridgecallno, we need to lock it.
2). stop_stuff() which calls iax2_destroy_helper()
Assumes the lock on the pvt is already held, when iax2_destroy_helper() is called.
Thus we need to lock the bridgecallno pvt before we call stop_stuff(iaxs[fr->callno]->bridgecallno);
3). When evaluating the state of 'callno->transferring' of the current leg,
we can't change it to READY unless the bridgecallno is locked.
Why, if we are interrupted by the other call leg before 'transferring = TRANSFER_RELEASED',
the interrupt will find that it is READY and that the bridgecallno is also READY so Releases the legs.
(closes issue ASTERISK-21409)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2594/
........
Merged revisions 391062 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 391063 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Features configuration is handled in its own API in
features_config.h and features_config.c. This way, features
configuration is accessible to anything that needs it.
In addition, features configuration has been altered to
be more channel-oriented. Most callers of features API
code will be supplying a channel so that the individual
channel's settings will be acquired rather than the global
setting.
Missing from this commit is XML documentation for the
features configuration. That will be handled in a separate
commit.
Review: https://reviewboard.asterisk.org/r/2578/
(issue ASTERISK-21542)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The attended transfer API call can complete the attended transfer in a number of ways
depending on the current bridged states of the channels involved.
The hiding of masquerades is done in some bridging-related functions, such as the manager
Bridge action and the Bridge dialplan application. In addition, call pickup was edited
to "move" a channel rather than masquerade it.
Review: https://reviewboard.asterisk.org/r/2511
(closes issue ASTERISK-21334)
Reported by Matt Jordan
(closes issue Asterisk-21336)
Reported by Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
* ChanSpyStart/Stop
* MonitorStart/Stop
* MusicOnHoldStart/Stop
* FullyBooted/Reload
* All Voicemail/MWI related events
In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.
Review: https://reviewboard.asterisk.org/r/2532
(closes issue ASTERISK-21462)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Breaks many things until they can be reworked. A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CallforwardNoAnswer uses a sched to determine when to forward the call.
Defaults to 20secs but configurable in skinny.conf.
Adds dialType to each subchannel structure to be used to differentiate
between normal dials that result in a call being placed (default) and
other uses for the skinny_dialer (such as cfwd digit collection).
Restructured all cfwd handling to use this new arrangement.
(closes issue ASTERISK-21292)
Reported by: wedhorn
Tested by: myself
Patches:
skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r388005, macros were introduced to consistently define message
types. This added an assert if a message type was used either before
it was initialized or after it had been cleaned up. It turns out that
this assertion fires during shutdown.
This actually exposed a hidden shutdown ordering problem. Since
unsubscribing is asynchronous, it's possible that the message types
used by the subscription could be freed before the final message of
the subscription was processed.
This patch adds stasis_subscription_join(), which blocks until the
last message has been processed by the subscription. Since joining was
most commonly done right after an unsubscribe, a
stasis_unsubscribe_and_join() convenience function was also added.
Similar functions were also added to the stasis_caching_topic and
stasis_message_router, since they wrap subscriptions and have similar
problems.
Other code in trunk was refactored to join() where appropriate, or at
least verify that the subscription was complete before being
destroyed.
Review: https://reviewboard.asterisk.org/r/2540
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Currently, the buffer for processing "inkeys" is limited to 256 characters. If
the user has many keys and the names of those key files are long, the 256
character limit is not enough.
* Change inkeys buffer to be dynamic
(closes issue ASTERISK-21398)
Reported by: Pavel Kopchyk
Tested by: Pavel Kopchyk, Michael L. Young
Patches:
asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2501/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The prior code committed, r385473, failed to take into consideration that not
all outgoing calls will be to a peer. My fault.
This patch does the following:
* Check if there is a related peer involved. If there is, check and set NAT
settings according to the peer's settings.
* Fix a problem with realtime peers. If the global setting has auto_force_rport
set and we issued a "sip reload" while a peer is still registered, the peer's
flags for NAT are reset to off. When this happens, we were always setting the
contact address of the peer to that of the full contact info that we had.
(closes issue ASTERISK-21374)
Reported by: jmls
Tested by: Michael L. Young
Patches:
asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2524/
........
Merged revisions 388601 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An endpoint is an external device/system that may offer/accept
channels to/from Asterisk. While this is a very useful concept for end
users, it is surprisingly not a core concept within Asterisk itself.
This patch defines ast_endpoint as a separate object, which channel
drivers may use to expose their concept of an endpoint. As the channel
driver creates channels, it can use ast_endpoint_add_channel() to
associate channels to the endpoint. This updated the endpoint
appropriately, and forwards all of the channel's events to the
endpoint's topic.
In order to avoid excessive locking on the endpoint object itself, the
mutable state is not accessible via getters. Instead, you can create a
snapshot using ast_endpoint_snapshot_create() to get a consistent
snapshot of the internal state.
This patch also includes a set of topics and messages associated with
endpoints, and implementations of the endpoint-related RESTful
API. chan_sip was updated to create endpoints with SIP peers, but the
state of the endpoints is not updated with the state of the peer.
Along for the ride in this patch is a Stasis test API. This is a
stasis_message_sink object, which can be subscribed to a Stasis
topic. It has functions for blocking while waiting for conditions in
the message sink to be fulfilled.
(closes issue ASTERISK-21421)
Review: https://reviewboard.asterisk.org/r/2492/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 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: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2475/
........
Merged revisions 387875 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 387880 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387885 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
........
Merged revisions 387345 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387369 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
........
Merged revisions 387319 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Remove t and ama local variables. There is no way they could be
anything other than default because p->owner can only be NULL at this
point.
* Rename tmp and tmp2 to owner and chan respectively.
* Remove redundant initialization of channel context, exten, priority.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Pretending that chan_local locals container can have more than one bucket
is silly. The container has no key to help search.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387211 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)
........
Merged revisions 387134 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.
SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.
API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 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
........
Merged revisions 385634 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385635 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/
........
Merged revisions 385473 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385474 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
........
Merged revisions 385403 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385406 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
........
Merged revisions 385173 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385174 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/
........
Merged revisions 384827 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384828 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
........
Merged revisions 384689 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 384003 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The softbutton endcall should not turn a transfer into a blind transfer but
hangup the exten being called and leave the original call on hold. This does
that.
(closes issue ASTERISK-21321)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-xferendcall01.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383948 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
........
Merged revisions 383863 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 383878 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code caused a compiler warning when --enable-dev-mode was not used.
The warning was that this variable was set but not used. That was indeed
the case as the only place this is used is as an argument to SKINNY_DEBUG
which is compiled out when not in dev mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.
Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 383165 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 383166 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 383125 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 382847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 382848 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added an option "discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event. This essentially stops telling the peer
to start music on hold.
(issue ABE-2899)
Reported by: Denis Alberto Martinez
Review: https://reviewboard.asterisk.org/r/2336/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 382739 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Originally, way back in r201583, we added the alternate RTP address so
that the RTP engine would expect to receive audio from a new source
when a glare re-INVITE occurred. In r382589, we remove the alternate
RTP source, as the 'secret' probation mode allows for switching to a new
RTP source when a previous source stops sending RTP. At the time, it
seemed appropriate to set the RTP source based on the information in the
glared re-INVITE.
Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs
with no SDP - such as in a connected line update that glances - we'll set
the RTP source to an invalid address. In subsequent re-INVITE requests from
this Asterisk instance, we'll then send an invalid media address, which will
result in the remote side sending a 488. Whoops.
There isn't any need to reset the RTP source - if we're using strictrtp, we'll
simply synchronize to a new source when we stop getting packets from the old
one. If we aren't using strictrtp, then again there shouldn't be a problem.
Note that the Asterisk Test Suite's connectedline test caught this error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Often, Asterisk may realize that a change in the source of an RTP stream is
about to occur and ask that the RTP engine reset it's lock on the current RTP
source. In certain scenarios, it may take awhile for the new remote system to
send RTP packets, while the old remote system may continue providing RTP during
that time period. This causes Asterisk to re-lock onto the old source, thereby
rejecting the new source when the old source stops sending RTP and the new
source begins.
This patch prevents that by having a constant secondary, 'secret' probation
mode enabled when an RTP source has been chosen. RTP packets from other sources
are always considered, but never chosen unless the current RTP source stops
sending RTP.
Review: https://reviewboard.asterisk.org/r/2364
(closes issue AST-1124)
Reported by: John Bigelow
Tested by: John Bigelow
(closes issue AST-1125)
Reported by: John Bigelow
Tested by: John Bigelow
........
Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.
A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.
Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/
(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
oolong-path-support-trunk in team branch by oej (License 5267)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 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)
........
Merged revisions 382322 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382323 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)
........
Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 382234 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Basically sets the callerid and callername to the first device talked to for the
purposes of putting the the calls made log on the device. Does not affect the device
displaying who the device is currently talking to.
Also some minor changes to use sub->exten in lieu of l->lastnumberdialed.
(closes issue ASTERISK-21095)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-calllogsoutbound03.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds both fixed and variable prinotify messages and clearprinotify messages to skinny.
Also adds cli function for pushing messages to devices. i
Initial code by snuffy, expanded by myself to include fixed messages.
(closes issue ASTERISK-21091)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-prinotify02.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 381975 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 381976 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a module's configuration is not loadable, we still load the module but it
is not in a running state. When trying to troubleshoot, let's say, why
chan_motif is ignoring inbound XMPP traffic, there is no way to indicate that a
loaded module is not currently running.
(closes issue ASTERISK-21108)
Reported by: Rusty Newton
Tested by: Michael L. Young
Patches:
asterisk-21108_add_status-v2.diff Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2331/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Patch adds all the packet and structure stuff to skinny to enable setting
service URLs in skinny, such as corporate directories.
This stuff is only relevant during load/unload as when activated. Also
some minor changes removing duplicated counting of addons and speedials in
handle_skinny_show_devices.
Review: https://reviewboard.asterisk.org/r/2321/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Auto complete for skinny debug allows multiple options and negation, also add
debug all option. Usage example: 'skinny debug all -packets' (each can be
autocompleted including -packet).
Change show device to use device name. Remove the duplicate ast_strdup's from
place calling device complete return immediately from complete devicename and
complete linename so that multiple options are displayed on the CLI if more
than one option available.
Review: https://reviewboard.asterisk.org/r/2333/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 381613 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381615 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)
........
Merged revisions 381566 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381568 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)
........
Merged revisions 381553 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:
1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
options that use the configuration framework
This patch include configuration documentation for the following modules:
* chan_motif
* res_xmpp
* app_confbridge
* app_skel
* udptl
Two new CLI commands have been added:
* config show help - show configuration help by module, category, and item
* xmldoc dump - dump the in-memory representation of the XML documentation to
a new XML file.
Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058
patches:
on review 2058 uploaded by twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Don't know why it seemed to work during testing, but it really is needed
for protocol v17 (and probably above).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix chan_skinny so that it respects callerID presentation of inbound calls to
device and a couple of other minor fixes: 145 packet (add OCTAL_FROM amd callerid),
and dont send dialednumber message if protocol >= 17.
(closes issue ASTERISK-21066)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-respect-clid-restrictions-v2.diff uploaded by snuffy (license 5024)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While adding red-black tree containers to astobj2 in r376575, Richard pointed
out the way chan_iax2 finds unused call numbers will prevent ao2_container
integrity checks at runtime.
This patch removes the ao2_container and instead uses fixed sized arrays and a
modified Fisher-Yates-Durstenfeld shuffle to maintain the call number list.
While the locking semantics are similar to the ao2_container implementation,
this implementation should be faster and more memory efficient.
Review: https://reviewboard.asterisk.org/r/2288/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381427 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.
........
Merged revisions 381282 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add extra string to transmit_callinfo_var, Only set string2 to tonum for outgoing calls
and changes to send_callinfo and push_callinfo to not set callid name to last number.
(closes issue ASTERISK-21063)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-callinfoupdate03.diff uploaded by wedhorn (license 5019)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 380942 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Make skinny reset vmexten and immeddial to '\0' on reload to ensure that
it is set to '\0' if the appropriate item is removed/commented in
skinny.conf. Also small fix re immeddial char in skinny.conf and add
immedial setting to skinny show settings.
(closes issue ASTERISK-21037)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
immed_dial_fix.diff uploaded by snuffy (license 5024)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add new variable length displayprompt packet (0x0145) to skinny. Uses the new
packet if the device is reporting protocol versions >= 17.
Add the use of octal codes for sending prompts to both the new and old
displayprompt messages (also cleaned up soft_key_template_default to use the
defined octal codes).
Review: https://reviewboard.asterisk.org/r/2294/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add packet 0x014A (variable length call info messages) to skinny for newer
firmware. Plenty of unknown information but includes the equivalent functionality
as the fixed size callinfo packet already included.
Only send this packet if protocol reported is >= 17.
Review: https://reviewboard.asterisk.org/r/2290/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380755 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/
........
Merged revisions 380696 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 380698 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is mostly a reorganization of existing code with a few exceptions:
* Added doxygen comments to all of the extracted functions.
* Split reload_firmware(int unload) into iax_firmware_reload() and
iax_firmware_unload() for readability.
* Create iax_firmware_traverse() to support the 'iax2 show firmware' CLI
command.
* Renamed iax_check_version() to iax_firmware_get_version() and change its
arguments and return value so that it returns a success/failure value and sets
the selected version into an out parameter to avoid confusion with failure and
version 0.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
........
Merged revisions 380480 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380500 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
........
Merged revisions 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 380465 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch just moves the IAX2 source and header files into a separate iax2
sub-directory in the channels directory, similar to how the sip source files are
structured.
The only thing that was added was an #ifndef to protect provision.h from multiple
inclusion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380433 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/
........
Merged revisions 380364 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 380384 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380386 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/
........
Merged revisions 380347 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 380350 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380383 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
........
Merged revisions 380331 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a dial softkey when the device is in DAFD. The softkey is greyed (unusable)
until a possible dialplan match is entered. Code includes updating
transmit_selectsoftkeys to allow the use of a button mask. Also add option
to use # or * as a dial now button. Original patch by snuffy cleaned up by myself.
Review: https://reviewboard.asterisk.org/r/2277/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380057 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)
........
Merged revisions 380043 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During the conversion to the newer CLI command structure the old definitions were
commented out. I think it's safe to remove them completely now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379936 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().
........
Merged revisions 379808 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379809 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)
........
Merged revisions 379677 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379678 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)
........
Merged revisions 379392 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379393 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The chan_misdn channel driver will send a channel with an invalid destination
to the 'i' extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it bounces the channel
to this extension. Dialplan writers everywhere moaned at yet another
inconsistency.
This is yet another example of why duplicating logic in multiple places results
in bugs that stick around in Jira for just under three years.
Yes: ASTERISK-15456 was created on January 18th, 2010. Patch committed on
January 15th, 2013. Ouch.
(closes issue ASTERISK-15456)
Reported by: Thomas Omerzu
patches:
chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license 5927)
........
Merged revisions 379145 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 379146 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.
This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.
Several things to note:
* The Right Thing(TM) to do would probably be to replace the
ast_build_string stuff with building an ast_xml_doc. That's a much
bigger change, and out of scope for the original ticket, so I
refrained myself.
* It is with great sadness that I wrote my own ast_xml_escape
function. There's one in libxml2, but it's knee-deep in
libxml2-ness, and not easily used to one-off escape a
string.
* I only escaped the string we know is causing problems
(local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
(closes issue ABE-2902)
Reported by: Guenther Kelleter
Tested by: Guenther Kelleter
Review: http://reviewboard.digium.internal/r/365/
........
Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
........
Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378934 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On a multihomed server when sending a NOTIFY message, we were not figuring out
which network should be used to contact the peer.
This patch fixes the problem by calling ast_sip_ouraddrfor() and then
build_via() so that our NOTIFY message contains the correct IP address.
Also, a debug message is being added to help follow the call-id changes that
occur. This was helpful for confirming that the IP address was set properly
since the call-id contains the IP address. It also will be helpful for
troubleshooting purposes when following a call in the debug logs.
(closes issue ASTERISK-20805)
Reported by: Bryan Hunt
Tested by: Bryan Hunt, Michael L. Young
Patches:
asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2255/
........
Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
........
Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378321 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a stack
allocation
* For XMPP (in res_jabber), the allocation has been eliminated since it was
unnecesary.
Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.
(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
........
Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 378287 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix local_alloc() unexpected limitation of exten and context length from
a combined length of 80 characters to a normal 80 characters each.
* Made local_alloc() and local_devicestate() parse the same way.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* awesome_locking() does not need to thrash the pvt lock as much.
* local_setoption() does not need to check for NULL pvt on cleanup since
it will never be NULL.
* Made ref the pvt before locking for consistency.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixup the vmexten so if globally set in general section will be honored by
chan_skinny. Also get rid of the 'global_' part of variable name to match
regexten.
(closes issue ASTERISK-20790)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-vm.diff uploaded by snuffy (license 5024)
........
Merged revisions 378010 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Review the syntax of the 'skinny debug' command to show more than
just 'show' for options to 'skinny debug' command.
(closes issue ASTERISK-20789)
Reported by: snuffy
Tested by: snuffy, myself
Patches:
skinny-debug.diff uploaded by snuffy (license 5024)
........
Merged revisions 377985 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
configured codecs to take precedence on an outgoing call.
This change introduces a new peer configuration property named
'ignore_requested_pref' that causes the requested codec to be ignored when
determining the preferred codec for an outgoing call leg. The consequence is
that Asterisk's usual efforts to prefer avoiding transcoding can be overridden
on a peer-by-peer basis where appropriate.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The issue comes from the fact that transfers may perform
a redirecting update on a channel. The issue is that lock
inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process.
The fix is to move when the redirecting update occurs to a
place where neither the tech_pvt or the channel is locked so
that the two can be locked in the proper order.
(closes issue ASTERISK-20708)
reported by Mark Michelson
patches:
ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)
Tested by:
Tim Ringenbach at Asteria Solutions Group
........
Merged revisions 377910 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During the TLS re-work in chan_sip some TLS specific code was moved
into a separate function. This function operates on a copy of the
incoming SIP request. This copy was never deinitialized causing a
memory leak for each request processed.
This function is now given a SIP request structure which it can use
to copy the incoming request into. This reduces the amount of memory
allocations done since the internal allocated components are reused
between packets and also ensures the SIP request structure is
deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763)
Reported by: deti
........
Merged revisions 377257 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377258 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377259 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When setting up an RTP instance the RTCP portion of the instance
keeps a reference to the instance itself. In order to release this
reference and stop RTCP the stop API call must be called before
destroying the instance.
(closes issue ASTERISK-20751)
Reported by: joshoa
........
Merged revisions 377021 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
