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
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
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
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
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
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
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
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
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
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After the "progressinband" value setting of "never" was updated to never send a
183 this separated its use from the "no" value. Since "never" was the default,
but most users probably expect "no" this patch updates the default for the
"progressinband" setting to "no."
ASTERISK-24835 #close
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4606/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434654 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434565 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434470 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434424 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434292 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434289 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)
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Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
