Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.
ASTERISK-24483
Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d
When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'. This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.
ASTERISK-23462 #close
Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318
* Mark the module deprecated.
* Disable the module by default.
* Produce a warning the first time a macro is used.
* Note deprecation related options in app_dial and app_queue.
ASTERISK-27350
Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc
As channels join and leave an SFU the bridge_softmix module
needs to renegotiate to add and remove their streams from
the other participants. Previously this was done by constructing
the ideal stream topology every time but in the case of leave
this was incomplete.
This change makes it so bridge_softmix keeps an ideal stream
topology for each channel and uses it when making changes. This
ensures that when we request a renegotiation we are always
certain that we are aiming for the best stream topology
possible. In the case of a channel leaving this ensures that
we try to have an existing participant fill their place if
a participant has a fixed limit on the maximum number of video
streams they allow.
ASTERISK-27354
Change-Id: I58070f421ddeadd2844a33b869b052630cf2e514
* Rename the Party A CDR container from active_cdrs_by_channel to
active_cdrs_master.
* Renamed the support functions associated with active_cdrs_master
appropriately.
ASTERISK-27335
Change-Id: I6104bb3edc3a0b7243ce502e45e8832b0cff14f7
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is because of a n*m loop used when
processing Party B information.
* Added a new CDR container that is keyed to Party B so we don't need such
a large loop when processing Party B information.
NOTE: To reduce the size of the patch I deferred to another patch the
renaming of the Party A active_cdrs_by_channel container to
active_cdrs_master and renaming the container's hash and cmp functions
appropriately.
ASTERISK-27335
Change-Id: I0bf66e8868f8adaa4b5dcf9e682e34951c350249
when 'all' is specified in an allow or disallow section, it should erase
all values from the inverse section in the default config. E.G.
allow=all should erase any deny values from default config &
vice-versa
ASTERISK-27333 #close
Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6
The sys/sysmacros.h include file does not exist in BSD systems and
is not required to build this module there.
Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section
already exist I moved that include line inside it's #else branch.
ASTERISK-27343 #close
Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1
PJSIP allows a domain name as external_media_address. This allows chan_pjsip to
be used behind a NAT with changing IP addresses. The IP address of that domain
is resolved to the c= line already. This change sets also the o= line to that
domain.
ASTERISK-27341 #close
Change-Id: I690163b6e762042ec38b3995aa5c9bea909d8ec4
When making channels compatible the bridge_simple module
will renegotiate one to better match the other. Some
endpoints incorrectly terminate the call if this process
fails.
To better handle this scenario the audio streams present
on the new requested topology will include any existing
negotiated formats that happen to exist on the first
valid audio stream. This ensures formats are persent that
are known to be acceptable to the remote endpoint.
ASTERISK-27259
Change-Id: I8fc0cc03e8bcfd0be8302f13b9f32d8268977f43
It's possible for bfdobj to be created but syms not created. If syms
was not allocated in the current loop iteration but was allocated in the
previous iteration it would crash.
ASTERISK-27340
Change-Id: I5b110c609f6dfe91339f782a99a431bca5837363
This avoids a crash on stopping a chan_sip which failed to start its TLS server.
ASTERISK-27339 #close
Change-Id: I327fc70db68eaaca5b50a15c7fd687fde79263d5
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is we were getting the global config
to determine if we needed to log a debugging message.
* Many calls to ao2_global_obj_ref() were just so we could determine if
debug mode is enabled. Made a global flag to check instead.
* Eliminated many RAII_VAR() usages associated with the remaining
ao2_global_obj_ref() calls.
* Added missing NULL checks for the returned ao2_global_obj_ref() value.
ASTERISK-27335
Change-Id: Iceaad93172862f610cad0188956634187bfcc7cd
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is we were getting the global config
even if we didn't need it.
* Most uses of the global config were only needed on off nominal code
paths so it makes sense to not get it until absolutely needed.
ASTERISK-27335
Change-Id: I00c63b7ec233e5bfffd5d976f05568613d3c2365
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is we were repeatedly setting string
fields to potentially the same string in base_process_party_a(). Setting
a string field involves allocating room for the new string out of a memory
pool which may have to allocate even more memory.
* Check to see if the string field is already set to the desired string.
ASTERISK-27335
Change-Id: I3ccb7e23f1488417e08cafe477755033eed65a7c
The string comparisons for setting these CDR variables was inverted. We
were repeatedly setting these CDR variables only if the channel snapshots
had the same value.
ASTERISK-27335
Change-Id: I9482073524411e7ea6c03805b16de200cb1669ea
Move ast_sip_add_usereqphone to be called after anonymization of URIs,
to prevent the user_eq_phone adding "user=phone" to URIs containing a
username that is not a phonenumber (RFC3261 19.1.1). An extra call to
ast_sip_add_usereqphone on the saved version before anonymization is
added to add user=phone" to the PAI.
ASTERISK-27047 #close
Change-Id: Ie5644bc66341b86dc08b1f7442210de2e6acdec6
ast_sip_add_usereqphone adds "user=phone" to the header every time is is
called without checking whether the param already exists. Preventing
this by searching to string representation of header for "user=phone".
ASTERISK-26988 #close
Change-Id: Ib84383b07254de357dc6a98d91fc1d2c2c3719e6
Add bridge_features structure to bridge creation. Specifically, this
implements mute and DTMF suppression, but others should be able to be
easily added to the same structure.
ASTERISK-27322 #close
Reported by: Darren Sessions
Sponsored by: AVOXI
Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61
