Commit Graph

3 Commits

Author SHA1 Message Date
John Bigelow d195728541 Add test suite events for when contacts are added or removed from an AOR
These are needed by the pjsip inbound registration test suite tests.

(issue ASTERISK-21833)
(issue ASTERISK-21834)
(issue ASTERISK-21835)
(issue ASTERISK-21837)

Review: https://reviewboard.asterisk.org/r/2700/
Review: https://reviewboard.asterisk.org/r/2739/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-12 22:05:18 +00:00
Joshua Colp 63a229e369 Fix a crash due to performing full URI validation on a contact which only contains '*'.
(closes issue AST-1198)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 23:38:00 +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