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Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK
(Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the
dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges.
Consequently, when the dynamic range is exhausted, this change utilizes payload
types in the range between 35 and 63 giving room for another 29 payload types.
ASTERISK-26311 #close
Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964
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PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS
patch.
The patch below fixes a write to freed memory under cartain DNS lookup
conditions.
0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch
ASTERISK-26516
Reported by: Richard Mudgett
Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5
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The res_pjsip_sdp_rtp module did not restrict the number of
formats added to a media stream in the SDP to the defined
limit. If allow=all was used with additional loaded codecs this
could result in the next media stream being overwritten some.
This change restricts the module to limit it to the defined
maximum and also increases the maximum in our bundled pjproject.
ASTERISK-26541 #close
Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8
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codecs.conf.sample was missing codec opus's configuration options, descriptions,
and examples. This patch adds the configuration options and examples to
codecs.conf.sample that can be used with codec_opus.
ASTERISK-26538 #close
Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b
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This patch adds three new CLI commands:
- ari show apps: list the registered ARI applications
- ari show app: show detailed information about an ARI application
- ari set debug: dump events being sent to an ARI application
Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.
ASTERISK-26488 #close
Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5
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If in sip.conf (general section) set option register_retry_403=no,
the command "sip show settings" return value:
Outbound reg. retry 403:0
If in sip.conf (general section) set option register_retry_403=yes,
the command "sip show settings" return value:
Outbound reg. retry 403:-1
* In static char "sip show settings" for "Outbound.reg. retry 403"
option use AST_CLI_YESNO
ASTERISK-26476 #close
Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9
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ASTERISK-25070
Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814
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PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD
define it to a constant. It is indeed not safe to assume there won't be
longer paths and Asterisk generally does err safely on such cases.
So even for HURD we'll just pretend PATH_MAX is 4096.
ASTERISK-25070 #close
Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3
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In order for pjsua and its python binding to actually negotiate
audio for the testsuite tests, it needs g711 and resample. The
pj* libraries themselves do not. Unfortunately, pjproject relies
on a brand new libresample that most distros don't ship so we need
to use the libresample already bundled with pjproject. Only the pjsua
executable and the _pjsua.so python library are linked with it so it
shouldn't interfere with asterisk itself.
Also it was pointed out that apply_patches couldn't handle multiple
patches that depended on each other during the dry-run, so the
dry-run was removed.
Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098
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The NewConnectedLine event has been added by commit fe7671f, but the
documentation was missing.
ASTERISK-26537 #close
Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6
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Headers declare that memcpy does not accept NULL argument for the first
two parameters. Add a conditional block to prevent memcpy and ast_free
from running on vectors with NULL element array.
ASTERISK-26526 #close
Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71
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Every ao2 object contains storage for a private variable data_size,
though the value is never read if AO2_DEBUG is disabled. This change
makes the variable conditional, reducing memory usage.
ASTERISK-26524 #close
Change-Id: If859929e507676ebc58b0f84247a4231e11da07f
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PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS
patch.
The patches below fix the DNS lookup race condition crash caused by
attempting to send the same message twice for the single DNS lookup.
0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch
0006-r5473-svn-backport-Fix-pending-query.patch
The patch below removes a cached DNS response from the hash table when
another thread is referencing the old entry. The table still contained
the entry when it was destroyed which can result in inexplicable crashes.
0006-r5475-svn-backport-Remove-DNS-cache-entry.patch
ASTERISK-26344 #close
Reported by: Ian Gilmour
ASTERISK-26387 #close
Reported by: Harley Peters
Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4
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main/Makefile includes third-party/pjproject/build.mak but
doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak"
evaluates to "/version.mak". Fix is to set PJDIR in main/Makefile
before the include.
Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604
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While publishing device state between multiple instances of Asterisk,
a crash will sporadically occur under high CPS which looks to be a
race condition operating on the publisher queue.
ASTERISK-26506
Change-Id: I28da25d346deb358eff1d563485cabc433ce1ed6
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It is only safe to run ast_register_cleanup callbacks when all modules
have been unloaded. Previously these callbacks were run during graceful
shutdown, making it possible to crash during shutdown.
ASTERISK-26513 #close
Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21
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Removing explicit transport definition for endpoints and registrations. It
isn't necessary and isn't generally advised.
ASTERISK-26514 #close
Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb
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Support for referring to DAHDI channels by logical names was added in
(FIXME: when? Asterisk 11? 1.8?) and was intended to be part of support
of refering to channels by name.
While technically usable, it has never been properly supported in
dahdi-tools, as using it would require many changes at the Asterisk
level. Instead logical mapping was added at the kernel level.
Thus it seems that refering to DAHDI channels by name is not really used
by anyone, and therefore should probably be removed.
Change-Id: I7d50bbfd9d957586f5cd06570244ef87bd54b485
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calls."
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Older versions of tar don't support the --strip-components option so
instead of doing 'tar --strip-components=1 -C source', we now just
untar to the tarball's root directory (pjproject-<version>) and
rename that directory to 'source'.
Also fixed an issue where the pjproject source directory is a hard
coded absolute pathname.
ASTERISK-26510 #close
ASTERISK-22480 #close
Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0
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ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
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The res_pjsip_caller_id module wrongly assumed that a
saved From header would always exist on sessions. This
is true until an inbound call is received and a session
timer causes an UPDATE to be sent. In this case there will
be no saved From header and a crash will occur. This change
makes it fall back to the From header of the outgoing request
if no saved From header is present.
ASTERISK-26307 #close
Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa
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When executing the MailboxExists dialplan application and
MAILBOX_EXISTS dialplan function the passed in temporary voice
mailbox was not cleared, causing it to try to free garbage.
ASTERISK-26503 #close
Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3
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When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
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When an explicit media_address is provided the address family
in the SDP needs to be set to reflect it.
ASTERISK-26309
Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79
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The test uses 4 threads to grow, count, lookup and shrink 15K objects
in a container. If there's only 1 execution engine available, the test
will complete in <50ms. If each threads gets its own execution engine,
the test may timeout after 60 seconds because the count thread does a
locked ao2_callback on the whole container in a tight loop with only
a sched_yield to give up time. The lock contention makes the test
execution times wildly variable and mostly timeout. 2 execution
engines are OK, 3 results in about 33% failure rate and >=4 causes
a 80% failure rate.
To fix, the sched_yield was changed to a usleep(500).
Also, the number of buckets specified for the container was an even
number so that was changed to the next prime number greater than
(MAX_HASH_ENTRIES / 100). That's 151 currently.
Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77
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On heavy loaded system the TCP/TLS incoming calls could be
disconnected by pjproject while these calls are being
processed by asterisk.
This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref
to inform pjproject that an INVITE session is in use.
ASTERISK-26482 #close
Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33
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- Call "rc_openlog" optional. If you do not call,
you will simply NULL instead of a name.
- On the one PID can be only one syslog channel.
And it can already be run in logger.c
- Calling rc_openlog we assigns a new name for
the channel syslog. This unexpected behavior for logger.c.
Most lesser evil, is to agree on a NULL name syslog
if the channel was not launched in logger.c.
It also solves the problem of memory leaks.
ASTERISK-26455 #close
Change-Id: Ic17c38de67583e971d78fe18807d1a9faf8f0afd
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