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This reverts commit 6b5a7ced136b7178ae0b2ba39221eba1cd2e37c9.
Change-Id: I61d1dbb2e69e1977f684b7dfc8e98211024e1cd1
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Libedit 3.1 is not build with unicode on as a default and so the
prototype for the el_gets callback changed from expecting a char buffer
to accepting a wchar buffer. If ast_el_read_char isn't changed,
the cli reads garbage from teh terminal.
Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and
updated ast_el_read_char to use the HAVE_ define to detemrine whether
to use char or wchar.
ASTERISK-26592 #close
Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a
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Fix ASTERISK-26565 by adding ast_rtp_instance_stop before
rtp instance destroy for chan_unistim. Also several fixes
for displayed text translation.
Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc
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into 13
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* Don't hold the req_wrapper lock too long in endpt_send_request(). We
could block the PJSIP monitor thread if the timeout timer expires.
sip_get_tpselector_from_endpoint() does a sorcery access that could take
awhile accessing a database. pjsip_endpt_send_request() might take awhile
if selecting a transport.
* Shorten the time that the req_wrapper lock is held in the callback
functions.
* Simplify endpt_send_request() req_wrapper->timeout code.
* Removed some redundant req_wrapper->timeout_timer->id assignments.
Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9
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Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b
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Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94
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When optimistic SRTP was on it was possible for us to still
set up a call without an audio stream if an offer was received
with required SRTP.
This change makes it so this scenario will now fail with a 488
response.
ASTERISK-26575
Change-Id: I7d14187037681f48879bd20319ac79d0877318f3
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ASTERISK-26558
Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e
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This reverts commit 9231a56cf3d6f5eca1bf2d37d827453400690773.
Multiple testsuite failures were detected after the fact.
Change-Id: I3bac8d7c3ddb69a4ddf6c5d6de0ffa5ff7ff3af7
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This reverts commit 5c10091f3d1430c6fc04015226f8c3e3aa9d8282.
Multiple testsuite failures were detected after the fact.
Change-Id: I397a841acc17ae230c512449cd6bed89d2ef3b73
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This reverts commit 2e3a3545754749de21873bfdc6d1a40ec7d8893f.
Multiple testsuite failures were detected after the fact.
Change-Id: Ia45fa4633fae74dca345b24bb6722737c63035de
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This reverts commit 44f7e252397fd87420b3374df26941d7436401b3.
Multiple testsuite failures were detected after the fact.
Change-Id: I56299087da22128a95f0c8f3955f740890d7ca65
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2 of the sanitizers didn't have default values so in systems that
don't support sanitizers menuselect would spit out warnings. They
were harmless but confusing. They've now been set to "0".
Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58
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late." into 13
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sets the variable ABANDONED to TRUE if the call was not answered.
ASTERISK-26558
Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3
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res_pjsip_sesssion was hooking into transaction and invite state
changes. One of the reasons for doing so was due to the
PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the
message sending process, and so we should call session supplements to
alter the outgoing message.
In reality, this event was meant to indicate that the message either
a) had already been sent, or
b) required a DNS lookup and would be sent when the DNS query
completed.
In case (a), this meant we were altering an already-sent
request/response for no reason. In case (b), this potentially meant we
could be trying to alter a request/response at the same time that the
DNS resolution completed. In this case, it meant we might be stomping on
memory being used by the thread actually sending the message. This
caused potential crashes and memory corruption.
This patch removes the calls to session supplements from the case where
the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to
alter the message at this point is too late, and it can cause nothing
but harm to try to do it. Because there were no longer any calls to the
handle_outgoing() function, it has been removed.
Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92
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This is another case where manual frame deferral can be replaced with
centralized routines instead.
Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e
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Rather than use manual frame deferral, just let the channel API do it
for us.
ASTERISK-26343
Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49
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AGI recently was modified to defer important frames. This was because
when AGI was used in a connected line interception routine, the
resulting connected line frame would end up getting discarded by the
AGI.
However, this caused bad behavior in other cases. Specifically, during a
transfer, if someone attempted to manually set the Caller ID on a
channel in an AGI, the deferred connected line frame would end up
overwriting what had been manually set in the AGI.
Since the initial issue was specific to interception routines, this
change removes the manual frame deferral from AGI and instead uses the
new frame deferral API in interception routines.
ASTERISK-26343 #close
Reported by Morton Tryfoss
Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208
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into 13
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function." into 13
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systems" into 13
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video source" into 13
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There are several places in Asterisk that have duplicated logic
for deferring important frames until later.
This commit adds a couple of API calls to facilitate this automatically.
ast_channel_start_defer_frames(): Future reads of deferrable frames on
this channel will be deferred until later.
ast_channel_stop_defer_frames(): Any frames that have been deferred get
requeued onto the channel.
ASTERISK-26343
Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641
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This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc.
Unfortunately, the aforementioned commit caused a regression (incoming calls
would eventually disconnect). Thus it is being removed.
ASTERISK-26523 #close
ASTERISK-25270
Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d
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into 13
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Changed output packets queue processing algo to one read-one write
instead of all read-all send
Remove h.245 tunneling parameter from ReleaseComplete packet
ASTERISK-24400 #close
Reported by: Dmitry Melekhov
Tested by: Dmitry Melekhov
Change-Id: I0b31933b062a21011dbac9a82b8bcfe345f406f6
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reset registration attempts count on success registration on gatekeeper
Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336
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into 13
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This is a regression over Asterisk 11, introduced by
2dc8a060064f359a17f5ebcd515d85fe5203c019. Previously, recordings started via
the automon DTMF code would automatically be mixed together using sox because
app_monitor would be called with the m option. This commit restores this
behavior.
Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759
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Not surprisingly, using Respoke (and possibly other systems) it is
possible to blow past the 16k limit for a WebSocket packet size. This
patch bumps it up to 32k, which, at least for Respoke, is sufficient.
For now.
Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that
matter), this patch adds a LOW_MEMORY directive that sets the buffer to
8k for systems who have asked for their reduced memory availability to
be considered.
Change-Id: Id235902537091b58608196844dc4b045e383cd2e
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