asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	ed19b8ee76	Add debug message to res_pjsip_endpoint_identifier_ip to indicate when an endpoint is successfully retrieved. (closes issue ASTERISK-22101) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	a7be7ea9c1	Missed a conversion to pjsip.conf in documentation and sorcery. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
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	12 years ago

Author

SHA1

Message

Date

Mark Michelson

ed19b8ee76

Add debug message to res_pjsip_endpoint_identifier_ip to indicate when an endpoint is successfully retrieved.

(closes issue ASTERISK-22101)
reported by Rusty Newton



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397109 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Mark Michelson

a7be7ea9c1

Missed a conversion to pjsip.conf in documentation and sorcery.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395824 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

3 Commits (ed19b8ee76f4f5d51511ba6a99496166a8017cdd)