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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
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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
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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
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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
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