asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	4d35941891	pjsip: reinvite for connected line updates occurs when it should not Connected line updates are now only sent out if an actual update needs to occur. This happens under the following conditions: 1. The endpoint we are sending to is trusted. 2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent. 3. The connected id's number and name are valid. Also added an SDP when an update is sent out. (closes issue AST-1212) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2831/ ........ Merged revisions 398806 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398808 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

Kevin Harwell

4d35941891

pjsip: reinvite for connected line updates occurs when it should not

Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:

1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.

Also added an SDP when an update is sent out.

(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
........

Merged revisions 398806 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398808 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

2 Commits (2966ca288c2d0b97199c09f37e8d3f8cb6b94638)