Matt Jordan [Fri, 12 Aug 2016 18:53:41 +0000 (13:53 -0500)]
manager: Add <see-also> links between related events
This patch adds some see-also references between related AMI events. It
focuses primarily on those events that are guaranteed to come in pairs,
such as DTMFBegin/DTMFEnd, as well as those that occur during the life
cycle of an Asterisk channel, such as Newchannel/Hangup.
Matt Jordan [Fri, 12 Aug 2016 16:15:38 +0000 (11:15 -0500)]
func_channel: Reorganize documentation
* Following the example of the PJSIP channel driver, the channel
technology specific documentation has been moved to the respective
channel drivers that provide that functionality. This has the benefit
of locating the documentation of items with those modules that provide
it.
* Examples of using the CHANNEL function for both standard items as well
as for PJSIP have been added.
* The 'max_forwards' standard item has been documented.
pjsip: Fix deadlock with suspend taskprocessor on masquerade
If both channels which should be masqueraded
are in the same serializer:
1st channel will be locked waiting condition 'complete'
2nd channel will be locked waiting condition 'suspended'
On heavy load system a chance that both channels will be in
the same serializer 'pjsip/distibutor' is very high.
To reproduce compile res_pjsip/pjsip_distributor.c with
DISTRIBUTOR_POOL_SIZE=1
Steps to reproduce:
1. Party A calls Party B (bridged call 'AB')
2. Party B places Party A on hold
3. Party B calls Voicemail app (non-bridged call 'BV')
4. Party B attended transfers Party A to voicemail using REFER.
5. When asterisk masquerades calls 'AB' and 'BV',
a deadlock is happened.
This patch adds a suspension indicator to the taskprocessor.
When a session suspends/unsuspends the serializer
it sets the indicator to the appropriate state.
The session checks the suspension indicator before
suspend the serializer.
Kevin Harwell [Tue, 9 Aug 2016 17:07:20 +0000 (12:07 -0500)]
alembic/sqlalchemy: auto increment only allowed on a single column
The extensions table defined two columns (id and priority) as primary key
autoincrement columns. However only one is allowed when defining the primary
key.
This patch removes the autoincrement attribute from the priority column since
it does not need to be as such and really should not have been on there in the
first place.
This patch also removes 'context', 'exten', and 'priority' from the primary key
index and creates a new combined unique contraint index on them.
This patch adds a new PJSIP specific dialplan function,
PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media
session will be refreshed via either an UPDATE or re-INVITE request.
When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function,
the formats in use on a PJSIP channel can be re-negotiated and changed
dynamically after call setup.
Mark Michelson [Tue, 9 Aug 2016 21:19:34 +0000 (16:19 -0500)]
res_rtp_asterisk: Cache local RTCP address.
When an RTCP packet is sent or received, res_rtp_asterisk generates a
Stasis event that contains the RTCP report as well as the local and
remote addresses that the report pertains to.
The addresses are determined using ast_find_ourip(). For the local
address, this will typically result in a lookup of the hostname of the
server, and then a DNS lookup of that hostname. If you do not have the
host in /etc/hosts, then this results in a full DNS lookup, which can
potentially block for some time.
This is especially problematic when performing RTCP reads, since those
are done on the same thread responsible for reading and writing media.
This patch addresses the issue by performing a lookup of the local
address when RTCP is allocated. We then use this cached local address
for the Stasis events when necessary.
The PJSIP taskprocessors could be overflowed on startup
if there are many (thousands) realtime endpoints
configured with unsolicited mwi.
The PJSIP stack could be totally unresponsive for a few minutes
after boot completed.
This patch creates a separate PJSIP serializers pool for mwi
and makes unsolicited mwi use serializers from this pool.
This patch also adds 2 new global options to tune taskprocessor
alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'.
This patch also adds new global option 'mwi_disable_initial_unsolicited'
to disable sending unsolicited mwi to all endpoints on startup.
If disabled then unsolicited mwi will start processing
on next endpoint's contact update.
Corey Farrell [Wed, 3 Aug 2016 20:39:46 +0000 (16:39 -0400)]
Add missing checks during startup.
This ensures startup is canceled due to allocation failures from the
following initializations.
* channel.c: ast_channels_init
* config_options.c: aco_init
On heavy loaded system with IMAP or DB storage,
'app_voicemail' taskprocessor queue could reach 500 scheduled tasks.
It could happen when the IMAP or DB server dies or is unreachable.
It could happen on startup when there are many (thousands)
realtime endpoints configured with unsolicited mwi.
If the taskprocessor queue reaches the high water level
then the alert is triggered and pjsip stops processing new requests
until the queue reaches the low water level to clear the alert.
This patch adds 2 new 'general' configuration options
to tune taskprocessor alert levels:
'tps_queue_high' - Taskprocessor high water alert trigger level.
'tps_queue_low' - Taskprocessor low water clear alert level
Mark Michelson [Fri, 29 Jul 2016 18:13:55 +0000 (13:13 -0500)]
Remove SILK payload mappings from Asterisk core.
SILK is a bit of a hog when it comes to using up our limited number of
dynamic payload types in the RTP engine. By freeing up four slots, it
allows for other codecs to potentially take the place.
Now, codec_silk.so will dynamically use the payload slots in the RTP
engine when it loads.
A better fix would be make RTP dynamic payload types actually
dynamic. However, at this stage of Asterisk 14 development, this is a
risky move that would be imprudent.
Joshua Colp [Mon, 1 Aug 2016 16:08:15 +0000 (16:08 +0000)]
sorcery: Use more compatible regex for local expressions.
This changes the use of an empty regex for both res_sorcery_config
and res_sorcery_memory to "." instead. This is a more compatible
regular expression which also works on FreeBSD.
David M. Lee [Wed, 27 Jul 2016 14:56:29 +0000 (09:56 -0500)]
Replace strdupa with more portable ast_strdupa
The strdupa function is a GNU extension, and not widely portable. We
have an ast_strdupa function used within Asterisk which is preferred.
I pulled the definition up from menuselect.c into the menuselect.h
header file so it can be shared across menuselect.
George Joseph [Sun, 24 Jul 2016 23:27:26 +0000 (17:27 -0600)]
menuselect: Various menuselect enhancements
* Add 'external' as a support level.
* Add ability for module directories to add entries to the menu
by adding members to the <module_prefix>/<module_prefix>.xml file.
* Expand the description field to 3 lines in the ncurses implementation.
* Allow the description field to wrap in the newt implementation.
* Add description field to the gtk implementation.
pbx.c: Fix handling of '-' in extension name and callerid
This adds a two strings to ast_exten. name to go with exten and
cidmatch_display to go with cidmatch. The new fields contain input used
to add the extension in the first place. The existing fields now
contain stripped input that excludes insignificant spaces and dashes.
These stripped fields should always be used for comparisons. The
unstripped fields should normally be used for display, but displaying
stripped values will not cause runtime errors.
Note the actual string is only stored twice if it contains dashes. If
no dashes are found then both 'char *' fields point to the same memory.
So this change has a minimum effect on memory usage.
The existing functions ast_get_extension_name and
ast_get_extension_cidmatch return unstripped values as they did before
this change. Other similar bugs likely still exist where unstripped
extensions are saved outside pbx.c then passed back in.
Richard Mudgett [Wed, 27 Jul 2016 22:17:53 +0000 (17:17 -0500)]
pbx.c: Allow dangerous functions when adding a hint to dialplan.
We can allow dangerous functions when adding a hint since altering
dialplan is itself a privileged activity. Otherwise, we could never
execute dangerous functions.
This patch fixes the issue in pjsip_tx_data_dec_ref()
when tx_data_destroy can be called more than once,
and checks if invalid value (e.g. NULL) is passed to.
This patch updates array limit checks and docs
in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability().
George Joseph [Sun, 17 Jul 2016 23:28:36 +0000 (17:28 -0600)]
pjproject_bundled: Update for pjproject 2.5.5
Add more --disable-* switches to Makefile.rules including
--disable-opus which was causing bundled pjproject to fail with
"undefined reference" errors in libasteriskpj.
Changed PJ_ENABLE_EXTRA_CHECK to 1.
Removed 2 obsolete patches and added a new one.
The new one was merged by Teluu on 6/27/2016.
David M. Lee [Wed, 27 Jul 2016 15:33:23 +0000 (10:33 -0500)]
Portably sscanf tv_usec
In a timeval, tv_usec is defined as a suseconds_t, which could be
different underlying types on different platforms. Instead of trying to
scanf directly into the timeval, scanf into a long int, then copy that
into the timeval.
Kevin Harwell [Wed, 27 Jul 2016 17:36:22 +0000 (12:36 -0500)]
rtp_engine: Failed assertion and wrong name given for codec
Fixed an assert check that would trigger when the passed in value was negative.
The negative value was being cast to an unsigned value. This resulted in the
check failing.
Also fixed another problem when loading formats in the engine. When setting the
mime type the format's name was being passed in instead of the codec's name.
When configuring SIP URIs in the pjsip.conf file it is
necessary to escape the semicolon so the parser does not
treat it as a comment. This change allows this to work in
the astconfigparser implementation.
A secondary bug where some data was lost if a configuration
option included a "=" in its value was also fixed.
A bug where sections would be considered equal despite
being different has also been fixed.
Richard Mudgett [Fri, 22 Jul 2016 03:28:25 +0000 (22:28 -0500)]
dsp.c: Fix erroneous fax tone detection.
The Goertzel calculations get less accurate the lower the signal level
being worked with becomes because there is less resolution remaining.
If it is too low we can erroneously detect a tone where none really
exists. The searched for fax frequencies not only need to be so much
stronger than the background noise they must also be a minimum strength.
* Add needed minimum threshold test to tone_detect().
* Set TONE_THRESHOLD to allow low volume frequency spread detection.
ASTERISK-26237 #close
Reported by: Richard Mudgett
Alexander Traud [Fri, 22 Jul 2016 10:46:02 +0000 (12:46 +0200)]
chan_sip: Enable Session-Timers for SIP over TCP (and TLS).
Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that
scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables
Session-Timers for SIP over TCP (and for SIP over TLS).
However with longer international calls via TCP, the SIP channel might break,
because all hops on the Internet route must stay online (have not a single power
outage, for example). Therefore with Session-Timers enabled (which are enabled
at default), you might see dropped calls. Consequently even with this change,
you might be better-off going for session-timers=refuse in your sip.conf.
George Joseph [Thu, 21 Jul 2016 14:05:03 +0000 (08:05 -0600)]
chan_sip: Prevent deadlock when issuing "sip show channels"
sip_show_channels locks the dialogs container first then locks each
sip_pvt so it can spit out the details. The rest of sip dialog
processing locks the sip_pvt first then locks the dialogs container
if it needs to. Both lock in the order they need but deadlocks can
result. To fix, sip_show_channels and sip_show_channelstats have
been converted to use an iterator rather than ao2_callback. This way
the container is locked only while getting the next entry and is
unlocked when the callback is called.
res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice.
This patch removed call of pjsip_tx_data_dec_ref in send_notify
if send_request failed.
The pjsip_dlg_send_request deletes the message on error by itself.
It seems this patch fixes next issues:
ASTERISK-26199
ASTERISK-26166
ASTERISK-26174
This adds support for tagging functions with the noreturn attribute.
If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE
and DO_CRASH are enabled then failed assertions never return. This can
resolve a large number of false positives with static analyzers.
Richard Mudgett [Tue, 12 Jul 2016 22:24:54 +0000 (17:24 -0500)]
res_fax: Fix FAXOPT(faxdetect) timeout option.
The fax detection timeout option did not work because basically the wrong
variable was checked in fax_detect_framehook(). As a result, the timer
would timeout immediately and disable fax detection.
* Fixed ignoring negative timeout values. We'd complain and then go right
on using the negative value.
* Fixed destroy_faxdetect() in the off-nominal case of an incomplete
object creation.
* Added more range checking to FAXOPT(gateway) timeout parameter.
ASTERISK-26214 #close
Reported by: Richard Mudgett
Richard Mudgett [Mon, 18 Jul 2016 21:16:56 +0000 (16:16 -0500)]
chan_dahdi: Add faxdetect_timeout option.
The new option allows the channel driver's faxdetect option to timeout on
a call after the specified number of seconds into a call. The new feature
is disabled if the timeout is set to zero. The option is disabled by
default.
* Don't clear dsp_features after passing them to the dsp code in
my_pri_ss7_open_media(). We should still remember them especially for the
new faxdetect_timeout option.
The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call. The new feature is disabled if the timeout is set
to zero. The option is disabled by default.
Unit tests: Use AST_TEST_DEFINE in conditional code only.
If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead
code. This places all existing unit tests into a conditional block if
they weren't already.
Alexander Traud [Mon, 18 Jul 2016 10:13:25 +0000 (12:13 +0200)]
res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets.
With this change, the initial RTP sequence number is randomly chosen not between
0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over
counter (ROC) synchronization is not lost for sRTP, when the very first RTP
packets get lost; see http://srtp.sourceforge.net/faq.html#Q6
projects / thirdparty / asterisk.git / log
