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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.
ASTERISK-26241 #close
Change-Id: I633ec139630cd83809aae00336cee4a10077e467
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A patch made to the master branch (Now the 14 branch) inadvertently made
libsrtp a required dependency in order to compile Asterisk. Rather than
create dummy defines to substitute for the defines supplied by libsrtp
when libsrtp is not available, most of the code in sdp_srtp.c is moved
into res_srtp.c. This gets more code out of Asterisk's core that isn't
used when SRTP is not available. This also makes another inadvertent
required dependency on libsrtp by Asterisk's core unlikely.
ASTERISK-26253 #close
Reported by: Ben Merrills
Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7
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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.
ASTERISK-26145 #close
Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b
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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.
ASTERISK-26183 #close
Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b
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libunbound at version 1.4.20 (which CentOS still uses) declared all
of their string function parameters as as 'char *'. 1.4.21 changed
them all to 'const char *'. Thankfully 1.4.21 also introduced the
UNBOUND_VERSION_MAJOR define so configure now checks for that and
sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks
that and casts away the 'const' if it's not set.
Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and
Fedora24 (1.5.4). There are a few failing tests to be addressed though.
ASTERISK-26283 #close
Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148
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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.
ASTERISK-26277 #close
Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b
(cherry picked from commit eec60dd77394f0519895fc6abce3a6f90f6470f1)
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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.
ASTERISK-26280 #close
Reported by Mark Michelson
Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556
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Modules must define AST_MODULE_SELF_SYM to be used as the name of a
generated function. This produces a friendly error when it's not
defined.
ASTERISK-26278 #close
Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b
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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.
ASTERISK-26230 #close
Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a
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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
ASTERISK-26229 #close
Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8
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This change replaces the custom unload process for the outbound
publish module with the common serializer shutdown group.
ASTERISK-25217 #close
Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6
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This file was out of sync with the current ARI definitions.
Change-Id: Ie7cb7d6d3c2eeb9cc9d683ca87b43b117e713d0a
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This ensures startup is canceled due to allocation failures from the
following initializations.
* channel.c: ast_channels_init
* config_options.c: aco_init
ASTERISK-26265 #close
Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611
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The regular expression would match causing the code that handled
the line if it was merely a comment to never get executed.
Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819
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transports."
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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.
ASTERISK-26206 #close
Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388
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ASTERISK-26256 #close
Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058
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Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe
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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.
Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612
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Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5
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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.
ASTERISK-26233 #close
Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f
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We can allow dangerous functions when adding a hint since altering
dialplan is itself a privileged activity. Otherwise, we could never
execute dangerous functions.
ASTERISK-25996 #close
Reported by: Andrew Nagy
Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba
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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().
Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40
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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.
ASTERISK-26148 #close
Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063
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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.
Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95
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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.
Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c
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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.
Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e
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