res_hep_pjsip.so and res_hep_rtcp.so will still load and do a lot of
unnecessary work even if 'enabled' is set to 'no' in hep.conf.
Change-Id: I3eddfeea09c6b5bc7c641952ee0ae487fd09b64b
When the RTCP code was transitioned over to Stasis a code change
was made to keep track of how many reports are present. This count
controlled where report blocks were placed in the RTCP report.
If a compound RTCP packet was received this logic would incorrectly
place a report block in the wrong location resulting in a write
to an invalid location.
This change removes this counting logic and always places the report
block at the first position. If in the future multiple reports are
supported the logic can be extended but for now keeping a count
serves no purpose.
ASTERISK-27382
ASTERISK-27429
Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116
This patch does three things associated with the initial incoming INVITE
request URI.
1) Add access to the full initial incoming INVITE request URI.
2) We were not setting DNID on incoming PJSIP channels. The DNID is the
user portion of the initial incoming INVITE Request-URI. The value is
accessed by reading CALLERID(dnid).
3) Fix CHANNEL(pjsip,target_uri) documentation.
* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).
* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.
* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.
* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.
ASTERISK-27478
Change-Id: I512e60d1f162395c946451becb37af3333337b33
Support for these protocols was added in the same commit as the 'proto'
field, so we can safely use the same ./configure check.
For reference: https://trac.pjsip.org/repos/changeset/4968
Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac
Use the new ast_cli_completion_add() function to improve completion
performance for commands like 'pjsip show endpoint.'
Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348
Using the LIKE operator requires a full table scan of 'astdb', whereas a
comparison operation is able to use the primary key index.
This patch adds a new function to the AstDB API for quick prefix matches
and updates res_sorcery_astdb to utilize it. This showed substantial
performance improvement in my test environment.
Related to ASTERISK~26806, but does not completely resolve it.
Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
res_stasis was missing AST_MODFLAG_LOAD_ORDER. Set res_stasis and
res_speech to start at (AST_MODPRI_APP_DEPEND - 1) so they are ready for
dependent modules.
Change-Id: I27f4f3810a95b6be8a5bfbf62be2ace6bfab6ff3
For both dynamic and static contacts it was possible that potential AOR
changes were not being applied to all contacts. This was because the qualify
and schedule code was only retrieving AOR's, and contacts with frequencies
greater than zero.
For instance the following could happen: and AOR/contact has a frequency of 5,
it then gets set to 0, and then a reload occurs. All scheduled OPTIONS are
stopped, a list of AOR's is retrieved with frequency > 0, but none are
selected since in this scenario all are 0. The contact for the one previously
set to 5 though does not get updated, so it's status remains "AVAILABLE".
This patch makes it so all contacts (static and dynamic) are selected, and
appropriately updated if need be.
ASTERISK-27467 #close
Change-Id: I7a920170f89c683af9505d4723a44fc6841decdb
Dynamic contacts were not being properly updated on reload. As a matter of
fact any changes to the AOR that a dynamic contact was associated with were
not being applied.
On reload, this patch makes it so for each dynamic contact, the associated
AOR is now retrieved and the AOR's fields are applied to the contact.
ASTERISK-27467
Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
More complicated direct media reinvite negotiations can result in longer
delays before direct media flows. The strictrtp learning timeout time
was too short. One log showed that the first RTP packet came in just
after three seconds.
* Increase the strictrtp learning timeout time from 1.5 to 5 seconds.
ASTERISK-27453
Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c
This change makes the presence of the GMIME_MAJOR_VERSION
definition optional, as not all versions of gmime actually
define it.
ASTERISK-27454
Change-Id: I01d99590045971ed6787899147170a5954077238
The patch for ASTERISK_24560 inverted a test checking if the bridge name
is being updated to a different name.
* Fix the test to return "Changing bridge name is not implemented" when
someone attempts to change the bridge name.
ASTERISK-27445
Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9
Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now,
srflx is gathered and sent via SDP, even when Asterisk is enabled for
Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used.
ASTERISK-27437
Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1
res_parking has an implicit load_pri of 0 meaining it's one of the very
first modules loaded after modules with global symbols. Set it
explicitly in the AST_MODULE_INFO block.
Change-Id: I297b6fb3ff6993ec004e667b22a74f5925906259
Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri
AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT.
Change-Id: I0123258eafce324249433a69df15a85cc16e509f
res_mwi_external_ami specified AST_MODFLAG_LOAD_ORDER but didn't set
load_pri, resulting in an actual load priority of 0. This module only
provides AMI actions so it has no reason to load early.
Change-Id: I82987fcf10d3ea42716b2f9df915b16687fd5839
Mac doesn't like the comparison of -1 to an enum, so store the result of
ast_sip_str_to_dtmf to an int so we can check for the negative return
value. ast_sip_str_to_dtmf returns an int so this is only delaying the
implicit type cast.
Change-Id: I0c262c1719ee951aae1f437d733a301cf5f8ad29
Some net-snmp builds do not provide the RONLY declare, only
NETSNMP_OLDAPI_RONLY. Map RONLY to NETSNMP_OLDAPI_RONLY to get around
this error.
Change-Id: Ida5c7ad9406515825485c4d3b4a34fd6ad0da577
It's impossible for gwtimeout or fdtimeout to be less than 0 because
they are unsigned int's. Remove checks and unreachable branches.
Change-Id: Ib2286960621e6ee245e40013c84986143302bc78
Some clients do not send rtp packets every ptime ms. This can lead to
situations in which the rtp source learning algorithm will never learn
the address of the client. This has been discovered on a Mac mini with
a pjsip based softphone after updating to Sierra: as soon as USB
headsets are involved, the softphone will send the second packet 30ms
after the first, the third 30ms after the second and the fourth 1ms
after the third. So in the old implmentation the rtp source learning
algorithm was repeatedly reset on the fourth packet.
The patch changes the algorithm in a way that doesn't take the arrival
time between two consecutive packets into account but the time between
the first and the last packet of a learning sequence.
The patch also fixes a second problem: when a user was using a wrong
value for the probation setting there was a LOG_WARNING output stating
that the value had been set to the default value instead. However
the code for setting the value back to defaults was missing.
ASTERISK-27421 #close
Change-Id: If778fe07678a6fd2041eaca7cd78267d0ef4fc6c
Domains themselves can be up to 255 characters long (per RFC 1035), so
our current buffer sizes are wholly inadequate for many use cases.
Change-Id: If3f30a68307f1365a1fe06bc4b854c62842c9292
We were not \0 terminating this string, so any attempt to print it would
in the best case show an empty string and in the worst case potentially
crash.
Change-Id: I63d96ef8f7516ac02a0f91e22dfa8acdc615042c
Previously for PJSIP the local address of WebSocket connections
was set to the remote address. For logging purposes this is
not particularly useful.
The WebSocket API has been extended to allow the local
address to be queried and this is used in PJSIP to set the
local address to the correct value.
The PJSIP HEP support has also been tweaked so that reliable
transports always use the local address on the transport
and do not try to (wrongly) guess. As they are connection
based it is impossible for the source to be anything else.
ASTERISK-26758
ASTERISK-27363
Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca
Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex
only so that they can anchor the potential match as a prefix and not
because they truly need regular expressions.
Rather than using regular expressions for simple prefix lookups, add
a new operation - ast_sorcery_retrieve_by_prefix - that does them.
Patches against 13 and 15 have a compatibility layer needed to
maintain ABI that is not needed in master.
Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79
