Age | Commit message (Collapse) | Author |
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This matches the behavior of the other SIP channel driver, chan_pjsip.
ASTERISK-27365
Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6
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Prevent unload of the module as certain pjsip initialization functions
cannot be reversed. This required a reorder of the module_load so that
the non-reversable pjsip functions are not called until all potential
errors have been ruled out.
ASTERISK-24483
Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7
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Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.
ASTERISK-24483
Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d
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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
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* 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
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Beside allowing AES-GCM again, this adds AES-192 again.
ASTERISK-27356
Change-Id: Ia97a435faf26300335d9552fa676b5d17e5f7233
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#ifdef"
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* 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
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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
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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
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it."
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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
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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
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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
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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
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This avoids a crash on stopping a chan_sip which failed to start its TLS server.
ASTERISK-27339 #close
Change-Id: I327fc70db68eaaca5b50a15c7fd687fde79263d5
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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
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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
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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
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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
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Add action to cancel feature attended transfer with AMI interface
ASTERISK-27215 #close
Change-Id: Iab8a81362b5a1757e2608f70b014ef863200cb42
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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
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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
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Try to defer some checks until needed in case there is an early exit.
Change-Id: Ibc6b34c38a4f60ad4f9b67984b7d070a07257064
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