MODULEINFO is checked while buidling/linking the module.
AST_MODULE_INFO is checked while loading/running the module.
ASTERISK-28838
Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042
This removes references that are no longer needed due to automatic
references created by module dependencies.
In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.
Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
Some SBCs require responses to empty SIP INFO packets
after establishing call via INVITE, if not responded to
they may drop your call after unspecified timeout of X minutes.
They are identified by having no Content-Type, check for this
and respond with 200 - OK message.
ASTERISK-24986 #close
Reported-by: Ilya Trikoz, Federico Santulli
Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0
PJSIP does not ensure that when printing the message body the
buffer will be NULL terminated. This is problematic when searching
for the signal and duration values of the DTMF.
This change ensures the buffer is always NULL terminated.
Change-Id: I52653a1a60c93092d06af31a27408d569cc98968
It is possible to receive incoming requests or responses after the channel
on an ast_sip_session has been destroyed and NULLed out. Handlers of these
sorts of requests or responses need to be prepared for the possibility
that the channel is NULL or else they could cause a crash.
While several places have been amended to deal with NULL channels, there
were still a couple of places that needed updating.
res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to
return early if there is no channel on the session.
res_pjsip_session.c: When handling a 302 response, we need to stop the
redirecting attempt if there is no channel on the session.
ASTERISK-25148 #close
reported by Mark Michelson
Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sean Bright
Naveen Albert
Alexander Traud
Corey Farrell
snuffy
Joshua Colp
Mark Michelson
Rodrigo Ramírez Norambuena
Kinsey Moore
Kevin Harwell