Commit graph

6 commits

Author SHA1 Message Date
Kinsey Moore
86a4ce4957 PJSIP: Enforce module load dependencies
This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub
have loaded properly before attempting to load any modules that depend
on them since the module loader system is not currently capable of
resolving module dependencies on its own.

ASTERISK-24312 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/4062/
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Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 425691 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-16 16:32:25 +00:00
Mark Michelson
dcf1ad14da Add module support level to ast_module_info structure. Print it in CLI "module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/3802



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25 16:47:17 +00:00
Kinsey Moore
e5542ab1e7 Prevent a crash in res_pjsip_dtmf_info.c
This change makes sure that a content type header exists before
checking the contents of the header against known SIP INFO DTMF content
types.
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Merged revisions 398206 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-03 18:09:02 +00:00
Kevin Harwell
aefebebd37 res_sip_dtmf_info: Support sending of 'raw' DTMF
Added the ability to handle 'raw' DTMF within the body of an INFO message.
Also made it so values 10-16 are mapped to valid DTMF values.

(closes issue ASTERISK-22144)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2776/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 22:09:16 +00:00
Joshua Colp
17f332169c Remove assumption in res_pjsip_dtmf_info that all INFO messages will contain a body.
(closes issue ASTERISK-22320)
Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 11:33:43 +00:00
Mark Michelson
735b30ad71 The large GULP->PJSIP renaming effort.
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
2013-07-30 18:14:50 +00:00
Renamed from res/res_sip_dtmf_info.c (Browse further)