Corey Farrell [Mon, 15 Aug 2016 20:29:53 +0000 (16:29 -0400)]
Refactor usage pattern of xmldoc info tag.
This updates func_channel.c and main/message.c to use a generic xpointer
include instead of including info from each channel driver. Now the
name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
documentation for func_channel. Setting the name attribute of info to
MessageToInfo or MessageFromInfo causes it to be included in the
MessageSend application and AMI action.
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.
George Joseph [Thu, 11 Aug 2016 16:24:34 +0000 (10:24 -0600)]
res_pjsip_caller_id: Copy header name to short header name
When compact_headers was set, we were sending a zero-length header name
for PAI and RPID because we always forced the short header name length
to 0. We did this because we cloned the header from "From" and wanted
to clear "f" from the sname. By cloning however, we bypass pjproject's
automatic logic that sets sname to name if there's no compact form of
the header, which there isn't for PAI and RPID. So now we force sname
to be the same as name right after we set name.
res_pjsip_diversion needed the same treatment for the Diversion header.
George Joseph [Thu, 11 Aug 2016 17:18:15 +0000 (11:18 -0600)]
res_pjsip: Fail global load if debug or default_from_user are empty
If debug was specified in the global configuration but left blank,
the logger would treat it as a wildcard and log all hosts. If
default_from_user was empty, a crash would result.
The global apply handler now checks for empty strings.
Matt Jordan [Thu, 11 Aug 2016 16:13:07 +0000 (11:13 -0500)]
app_queue: Prevent crash when a call is forwarded to an invalid location
When a call forward attempt is made from a Queue member, the current
code will hang up the forwarding channel in an off-nominal condition
prior to raising the Stasis events informing the rest of Asterisk that
the call was forwarded. This will result in a slew of dreaded FRACKs,
most likely leading to a crash.
This patch modifies the code such that we don't hang up the forwarding
channel even in an off-nominal condition until we've safely raised the
Stasis messages.
Kevin Harwell [Thu, 11 Aug 2016 15:50:09 +0000 (10:50 -0500)]
alembic: add auth_username to endpoint's identify_by enum
A new identify_by option was added recently, auth_username. However, this
setting was not added as an allowable choice in the database enumeration
value.
This patch updates the current enumeration, adding in the new setting.
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.
projects / thirdparty / asterisk.git / log
